
This reverts commit 81c0cf287c8514cb1cd6f3baca484d668c6eb128. Reason for revert: internal test failures Original change's description: > Simplification and refactoring of the AudioBuffer code > > This CL performs a major refactoring and simplification > of the AudioBuffer code that. > -Removes 7 of the 9 internal buffers of the AudioBuffer. > -Avoids the implicit copying required to keep the > internal buffers in sync. > -Removes all code relating to handling of fixed-point > sample data in the AudioBuffer. > -Changes the naming of the class methods to reflect > that only floating point is handled. > -Corrects some bugs in the code. > -Extends the handling of internal downmixing to be > more generic. > > Bug: webrtc:10882 > Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828 > Commit-Queue: Per Åhgren <peah@webrtc.org> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28928} TBR=gustaf@webrtc.org,peah@webrtc.org Change-Id: I2729e3ad24b3a9b40b368b84cb565c859e79b51e No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10882 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150084 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28931}
170 lines
4.8 KiB
C++
170 lines
4.8 KiB
C++
/*
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* Copyright (c) 2012 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/voice_detection_impl.h"
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#include "api/audio/audio_frame.h"
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#include "common_audio/vad/include/webrtc_vad.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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class VoiceDetectionImpl::Vad {
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public:
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Vad() {
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state_ = WebRtcVad_Create();
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RTC_CHECK(state_);
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int error = WebRtcVad_Init(state_);
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RTC_DCHECK_EQ(0, error);
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}
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~Vad() { WebRtcVad_Free(state_); }
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VadInst* state() { return state_; }
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private:
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VadInst* state_ = nullptr;
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RTC_DISALLOW_COPY_AND_ASSIGN(Vad);
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};
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VoiceDetectionImpl::VoiceDetectionImpl(rtc::CriticalSection* crit)
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: crit_(crit) {
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RTC_DCHECK(crit);
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}
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VoiceDetectionImpl::~VoiceDetectionImpl() {}
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void VoiceDetectionImpl::Initialize(int sample_rate_hz) {
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rtc::CritScope cs(crit_);
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sample_rate_hz_ = sample_rate_hz;
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std::unique_ptr<Vad> new_vad;
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if (enabled_) {
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new_vad.reset(new Vad());
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}
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vad_.swap(new_vad);
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using_external_vad_ = false;
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frame_size_samples_ =
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static_cast<size_t>(frame_size_ms_ * sample_rate_hz_) / 1000;
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set_likelihood(likelihood_);
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}
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bool VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
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rtc::CritScope cs(crit_);
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RTC_DCHECK(enabled_);
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RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
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audio->num_frames_per_band());
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std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> mixed_low_pass_data;
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rtc::ArrayView<const int16_t> mixed_low_pass(mixed_low_pass_data.data(),
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audio->num_frames_per_band());
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if (audio->num_proc_channels() == 1) {
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FloatS16ToS16(audio->split_bands_const_f(0)[kBand0To8kHz],
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audio->num_frames_per_band(), mixed_low_pass_data.data());
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} else {
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const int num_channels = static_cast<int>(audio->num_channels());
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for (size_t i = 0; i < audio->num_frames_per_band(); ++i) {
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int32_t value =
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FloatS16ToS16(audio->split_channels_const_f(kBand0To8kHz)[0][i]);
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for (int j = 1; j < num_channels; ++j) {
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value +=
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FloatS16ToS16(audio->split_channels_const_f(kBand0To8kHz)[j][i]);
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}
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mixed_low_pass_data[i] = value / num_channels;
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}
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}
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int vad_ret = WebRtcVad_Process(vad_->state(), sample_rate_hz_,
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mixed_low_pass.data(), frame_size_samples_);
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if (vad_ret == 0) {
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stream_has_voice_ = false;
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return false;
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} else if (vad_ret == 1) {
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stream_has_voice_ = true;
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} else {
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RTC_NOTREACHED();
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}
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return stream_has_voice_;
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}
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int VoiceDetectionImpl::Enable(bool enable) {
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rtc::CritScope cs(crit_);
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if (enabled_ != enable) {
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enabled_ = enable;
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Initialize(sample_rate_hz_);
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}
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return AudioProcessing::kNoError;
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}
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bool VoiceDetectionImpl::is_enabled() const {
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rtc::CritScope cs(crit_);
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return enabled_;
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}
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int VoiceDetectionImpl::set_stream_has_voice(bool has_voice) {
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rtc::CritScope cs(crit_);
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using_external_vad_ = true;
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stream_has_voice_ = has_voice;
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return AudioProcessing::kNoError;
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}
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bool VoiceDetectionImpl::stream_has_voice() const {
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rtc::CritScope cs(crit_);
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// TODO(ajm): enable this assertion?
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// RTC_DCHECK(using_external_vad_ || is_component_enabled());
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return stream_has_voice_;
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}
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int VoiceDetectionImpl::set_likelihood(VoiceDetection::Likelihood likelihood) {
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rtc::CritScope cs(crit_);
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likelihood_ = likelihood;
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if (enabled_) {
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int mode = 2;
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switch (likelihood) {
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case VoiceDetection::kVeryLowLikelihood:
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mode = 3;
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break;
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case VoiceDetection::kLowLikelihood:
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mode = 2;
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break;
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case VoiceDetection::kModerateLikelihood:
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mode = 1;
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break;
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case VoiceDetection::kHighLikelihood:
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mode = 0;
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break;
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default:
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RTC_NOTREACHED();
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break;
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}
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int error = WebRtcVad_set_mode(vad_->state(), mode);
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RTC_DCHECK_EQ(0, error);
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}
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return AudioProcessing::kNoError;
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}
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VoiceDetection::Likelihood VoiceDetectionImpl::likelihood() const {
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rtc::CritScope cs(crit_);
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return likelihood_;
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}
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int VoiceDetectionImpl::set_frame_size_ms(int size) {
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rtc::CritScope cs(crit_);
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RTC_DCHECK_EQ(10, size); // TODO(ajm): remove when supported.
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frame_size_ms_ = size;
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Initialize(sample_rate_hz_);
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return AudioProcessing::kNoError;
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}
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int VoiceDetectionImpl::frame_size_ms() const {
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rtc::CritScope cs(crit_);
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return frame_size_ms_;
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}
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} // namespace webrtc
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