4458d09ee4
It was hardly used, making the code more complex than needed and caused problems on iOS because it uses system. BUG=webrtc:5549 R=kjellander@webrtc.org Review URL: https://codereview.webrtc.org/1708353002 . Cr-Commit-Position: refs/heads/master@{#11677}
129 lines
4.5 KiB
C++
129 lines
4.5 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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//
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// Command line tool for speech intelligibility enhancement. Provides for
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// running and testing intelligibility_enhancer as an independent process.
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// Use --help for options.
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//
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#include <sys/stat.h>
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#include "gflags/gflags.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/common_audio/include/audio_util.h"
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#include "webrtc/common_audio/wav_file.h"
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
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#include "webrtc/modules/audio_processing/noise_suppression_impl.h"
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using std::complex;
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namespace webrtc {
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namespace {
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DEFINE_double(clear_alpha, 0.9, "Power decay factor for clear data.");
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DEFINE_int32(sample_rate,
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16000,
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"Audio sample rate used in the input and output files.");
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DEFINE_int32(ana_rate,
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60,
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"Analysis rate; gains recalculated every N blocks.");
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DEFINE_double(gain_limit, 1000.0, "Maximum gain change in one block.");
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DEFINE_string(clear_file, "speech.wav", "Input file with clear speech.");
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DEFINE_string(noise_file, "noise.wav", "Input file with noise data.");
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DEFINE_string(out_file, "proc_enhanced.wav", "Enhanced output file.");
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const size_t kNumChannels = 1;
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// void function for gtest
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void void_main(int argc, char* argv[]) {
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google::SetUsageMessage(
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"\n\nInput files must be little-endian 16-bit signed raw PCM.\n");
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google::ParseCommandLineFlags(&argc, &argv, true);
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size_t samples; // Number of samples in input PCM file
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size_t fragment_size; // Number of samples to process at a time
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// to simulate APM stream processing
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// Load settings and wav input.
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fragment_size = FLAGS_sample_rate / 100; // Mirror real time APM chunk size.
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// Duplicates chunk_length_ in
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// IntelligibilityEnhancer.
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struct stat in_stat, noise_stat;
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ASSERT_EQ(stat(FLAGS_clear_file.c_str(), &in_stat), 0)
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<< "Empty speech file.";
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ASSERT_EQ(stat(FLAGS_noise_file.c_str(), &noise_stat), 0)
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<< "Empty noise file.";
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samples = std::min(in_stat.st_size, noise_stat.st_size) / 2;
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WavReader in_file(FLAGS_clear_file);
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std::vector<float> in_fpcm(samples);
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in_file.ReadSamples(samples, &in_fpcm[0]);
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FloatS16ToFloat(&in_fpcm[0], samples, &in_fpcm[0]);
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WavReader noise_file(FLAGS_noise_file);
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std::vector<float> noise_fpcm(samples);
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noise_file.ReadSamples(samples, &noise_fpcm[0]);
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FloatS16ToFloat(&noise_fpcm[0], samples, &noise_fpcm[0]);
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// Run intelligibility enhancement.
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IntelligibilityEnhancer::Config config;
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config.sample_rate_hz = FLAGS_sample_rate;
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config.decay_rate = static_cast<float>(FLAGS_clear_alpha);
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config.analysis_rate = FLAGS_ana_rate;
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config.gain_change_limit = FLAGS_gain_limit;
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IntelligibilityEnhancer enh(config);
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rtc::CriticalSection crit;
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NoiseSuppressionImpl ns(&crit);
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ns.Initialize(kNumChannels, FLAGS_sample_rate);
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ns.Enable(true);
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AudioBuffer capture_audio(fragment_size,
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kNumChannels,
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fragment_size,
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kNumChannels,
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fragment_size);
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StreamConfig stream_config(FLAGS_sample_rate, kNumChannels);
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// Slice the input into smaller chunks, as the APM would do, and feed them
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// through the enhancer.
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float* clear_cursor = &in_fpcm[0];
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float* noise_cursor = &noise_fpcm[0];
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for (size_t i = 0; i < samples; i += fragment_size) {
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capture_audio.CopyFrom(&noise_cursor, stream_config);
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ns.AnalyzeCaptureAudio(&capture_audio);
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ns.ProcessCaptureAudio(&capture_audio);
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enh.SetCaptureNoiseEstimate(ns.NoiseEstimate());
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enh.ProcessRenderAudio(&clear_cursor, FLAGS_sample_rate, kNumChannels);
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clear_cursor += fragment_size;
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noise_cursor += fragment_size;
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}
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FloatToFloatS16(&in_fpcm[0], samples, &in_fpcm[0]);
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WavWriter out_file(FLAGS_out_file, FLAGS_sample_rate, kNumChannels);
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out_file.WriteSamples(&in_fpcm[0], samples);
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}
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} // namespace
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} // namespace webrtc
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int main(int argc, char* argv[]) {
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webrtc::void_main(argc, argv);
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return 0;
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}
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