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https://github.com/mollyim/webrtc.git
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bab128555a
- Switch from ptr+size to rtc::ArrayView - Remove `AgcManagerDirect::sample_rate_hz_` since it's always 16 kHz - Stop passing nullptr in agc_manager_direct_unittest.cc when `AgcManagerDirect::Process()` is called - Allow to correctly run the tests added in the child CL (see [1]) [1] https://webrtc-review.googlesource.com/c/src/+/250141 Bug: webrtc:7494 Change-Id: I0292d7038d6510ca7c58af32b6003a1e4b121b00 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250541 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35910}
85 lines
3.1 KiB
C++
85 lines
3.1 KiB
C++
/*
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* Copyright (c) 2015 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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#include "modules/audio_processing/vad/voice_activity_detector.h"
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#include <algorithm>
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace {
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const size_t kNumChannels = 1;
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const double kDefaultVoiceValue = 1.0;
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const double kNeutralProbability = 0.5;
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const double kLowProbability = 0.01;
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} // namespace
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VoiceActivityDetector::VoiceActivityDetector()
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: last_voice_probability_(kDefaultVoiceValue),
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standalone_vad_(StandaloneVad::Create()) {}
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VoiceActivityDetector::~VoiceActivityDetector() = default;
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// Because ISAC has a different chunk length, it updates
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// `chunkwise_voice_probabilities_` and `chunkwise_rms_` when there is new data.
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// Otherwise it clears them.
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void VoiceActivityDetector::ProcessChunk(const int16_t* audio,
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size_t length,
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int sample_rate_hz) {
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RTC_DCHECK_EQ(length, sample_rate_hz / 100);
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// TODO(bugs.webrtc.org/7494): Remove resampling and force 16 kHz audio.
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// Resample to the required rate.
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const int16_t* resampled_ptr = audio;
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if (sample_rate_hz != kSampleRateHz) {
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RTC_CHECK_EQ(
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resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels),
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0);
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resampler_.Push(audio, length, resampled_, kLength10Ms, length);
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resampled_ptr = resampled_;
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}
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RTC_DCHECK_EQ(length, kLength10Ms);
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// Each chunk needs to be passed into `standalone_vad_`, because internally it
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// buffers the audio and processes it all at once when GetActivity() is
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// called.
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RTC_CHECK_EQ(standalone_vad_->AddAudio(resampled_ptr, length), 0);
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audio_processing_.ExtractFeatures(resampled_ptr, length, &features_);
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chunkwise_voice_probabilities_.resize(features_.num_frames);
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chunkwise_rms_.resize(features_.num_frames);
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std::copy(features_.rms, features_.rms + chunkwise_rms_.size(),
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chunkwise_rms_.begin());
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if (features_.num_frames > 0) {
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if (features_.silence) {
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// The other features are invalid, so set the voice probabilities to an
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// arbitrary low value.
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std::fill(chunkwise_voice_probabilities_.begin(),
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chunkwise_voice_probabilities_.end(), kLowProbability);
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} else {
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std::fill(chunkwise_voice_probabilities_.begin(),
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chunkwise_voice_probabilities_.end(), kNeutralProbability);
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RTC_CHECK_GE(
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standalone_vad_->GetActivity(&chunkwise_voice_probabilities_[0],
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chunkwise_voice_probabilities_.size()),
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0);
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RTC_CHECK_GE(pitch_based_vad_.VoicingProbability(
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features_, &chunkwise_voice_probabilities_[0]),
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0);
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}
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last_voice_probability_ = chunkwise_voice_probabilities_.back();
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}
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}
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} // namespace webrtc
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