[pulseaudio-commits] [Git][pulseaudio/webrtc-audio-processing][master] Remove rnn_vad_tool.cc that contains main().

Tanu Kaskinen (@tanuk) gitlab at gitlab.freedesktop.org
Wed Sep 8 12:21:52 UTC 2021



Tanu Kaskinen pushed to branch master at PulseAudio / webrtc-audio-processing


Commits:
57ec282d by Ryo Kawaguchi at 2021-09-08T12:21:50+00:00
Remove rnn_vad_tool.cc that contains main().

- - - - -


3 changed files:

- webrtc/modules/audio_processing/agc2/rnn_vad/BUILD.gn
- − webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc
- webrtc/modules/audio_processing/meson.build


Changes:

=====================================
webrtc/modules/audio_processing/agc2/rnn_vad/BUILD.gn
=====================================
@@ -215,19 +215,4 @@ if (rtc_include_tests) {
       deps += [ ":unittests_bundle_data" ]
     }
   }
-
-  rtc_executable("rnn_vad_tool") {
-    testonly = true
-    sources = [ "rnn_vad_tool.cc" ]
-    deps = [
-      ":rnn_vad",
-      ":rnn_vad_common",
-      "../../../../api:array_view",
-      "../../../../common_audio",
-      "../../../../rtc_base:rtc_base_approved",
-      "../../../../test:test_support",
-      "//third_party/abseil-cpp/absl/flags:flag",
-      "//third_party/abseil-cpp/absl/flags:parse",
-    ]
-  }
 }


=====================================
webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc deleted
=====================================
@@ -1,120 +0,0 @@
-/*
- *  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 "absl/flags/flag.h"
-#include "absl/flags/parse.h"
-#include "common_audio/resampler/push_sinc_resampler.h"
-#include "common_audio/wav_file.h"
-#include "modules/audio_processing/agc2/rnn_vad/common.h"
-#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
-#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
-#include "rtc_base/logging.h"
-
-ABSL_FLAG(std::string, i, "", "Path to the input wav file");
-ABSL_FLAG(std::string, f, "", "Path to the output features file");
-ABSL_FLAG(std::string, o, "", "Path to the output VAD probabilities file");
-
-namespace webrtc {
-namespace rnn_vad {
-namespace test {
-
-int main(int argc, char* argv[]) {
-  absl::ParseCommandLine(argc, argv);
-  rtc::LogMessage::LogToDebug(rtc::LS_INFO);
-
-  // Open wav input file and check properties.
-  const std::string input_wav_file = absl::GetFlag(FLAGS_i);
-  WavReader wav_reader(input_wav_file);
-  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.
-  const std::string output_vad_probs_file = absl::GetFlag(FLAGS_o);
-  FILE* vad_probs_file = fopen(output_vad_probs_file.c_str(), "wb");
-  FILE* features_file = nullptr;
-  const std::string output_feature_file = absl::GetFlag(FLAGS_f);
-  if (!output_feature_file.empty()) {
-    features_file = fopen(output_feature_file.c_str(), "wb");
-  }
-
-  // Initialize.
-  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);
-  FeaturesExtractor features_extractor;
-  std::array<float, kFeatureVectorSize> feature_vector;
-  RnnBasedVad rnn_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());
-
-    // Extract features and feed the RNN.
-    bool is_silence = features_extractor.CheckSilenceComputeFeatures(
-        samples_10ms_24kHz, feature_vector);
-    float vad_probability =
-        rnn_vad.ComputeVadProbability(feature_vector, is_silence);
-    // Write voice probability.
-    RTC_DCHECK_GE(vad_probability, 0.f);
-    RTC_DCHECK_GE(1.f, vad_probability);
-    fwrite(&vad_probability, sizeof(float), 1, vad_probs_file);
-    // 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);
-      if (is_silence) {
-        // Do not write uninitialized values.
-        feature_vector.fill(0.f);
-      }
-      fwrite(feature_vector.data(), sizeof(float), kFeatureVectorSize,
-             features_file);
-    }
-  }
-
-  // Close output file(s).
-  fclose(vad_probs_file);
-  RTC_LOG(LS_INFO) << "VAD probabilities written to " << output_vad_probs_file;
-  if (features_file) {
-    fclose(features_file);
-    RTC_LOG(LS_INFO) << "features written to " << output_feature_file;
-  }
-
-  return 0;
-}
-
-}  // namespace test
-}  // namespace rnn_vad
-}  // namespace webrtc
-
-int main(int argc, char* argv[]) {
-  return webrtc::rnn_vad::test::main(argc, argv);
-}


=====================================
webrtc/modules/audio_processing/meson.build
=====================================
@@ -88,7 +88,6 @@ webrtc_audio_processing_sources = [
   'agc2/rnn_vad/pitch_search.cc',
   'agc2/rnn_vad/pitch_search_internal.cc',
   'agc2/rnn_vad/rnn.cc',
-  'agc2/rnn_vad/rnn_vad_tool.cc',
   'agc2/rnn_vad/spectral_features.cc',
   'agc2/rnn_vad/spectral_features_internal.cc',
   'agc2/saturation_protector.cc',



View it on GitLab: https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/commit/57ec282d4ff225baef20562e4212392925295ed0

-- 
View it on GitLab: https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/commit/57ec282d4ff225baef20562e4212392925295ed0
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