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

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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}
This commit is contained in:
Alessio Bazzica
2018-04-17 15:58:40 +02:00
committed by Commit Bot
parent 4478d9b773
commit a44b91de3e
4 changed files with 199 additions and 0 deletions
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1
modules/audio_processing/agc2/BUILD.gn
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@ -43,6 +43,7 @@ rtc_source_set("adaptive_digital") {
"../../../rtc_base:safe_minmax",
"../vad",
"../vad:vad_with_level",
"rnn_vad",
]
}

41
modules/audio_processing/agc2/rnn_vad/BUILD.gn Normal file
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@ -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",
]
}
}

23
modules/audio_processing/agc2/rnn_vad/common.h Normal file
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@ -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_

134
modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc Normal file
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@ -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);
}