| /* |
| * Copyright (c) 2012 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 "webrtc/modules/audio_processing/voice_detection_impl.h" |
| |
| #include "webrtc/common_audio/vad/include/webrtc_vad.h" |
| #include "webrtc/modules/audio_processing/audio_buffer.h" |
| #include "webrtc/rtc_base/constructormagic.h" |
| |
| namespace webrtc { |
| class VoiceDetectionImpl::Vad { |
| public: |
| Vad() { |
| state_ = WebRtcVad_Create(); |
| RTC_CHECK(state_); |
| int error = WebRtcVad_Init(state_); |
| RTC_DCHECK_EQ(0, error); |
| } |
| ~Vad() { |
| WebRtcVad_Free(state_); |
| } |
| VadInst* state() { return state_; } |
| private: |
| VadInst* state_ = nullptr; |
| RTC_DISALLOW_COPY_AND_ASSIGN(Vad); |
| }; |
| |
| VoiceDetectionImpl::VoiceDetectionImpl(rtc::CriticalSection* crit) |
| : crit_(crit) { |
| RTC_DCHECK(crit); |
| } |
| |
| VoiceDetectionImpl::~VoiceDetectionImpl() {} |
| |
| void VoiceDetectionImpl::Initialize(int sample_rate_hz) { |
| rtc::CritScope cs(crit_); |
| sample_rate_hz_ = sample_rate_hz; |
| std::unique_ptr<Vad> new_vad; |
| if (enabled_) { |
| new_vad.reset(new Vad()); |
| } |
| vad_.swap(new_vad); |
| using_external_vad_ = false; |
| frame_size_samples_ = |
| static_cast<size_t>(frame_size_ms_ * sample_rate_hz_) / 1000; |
| set_likelihood(likelihood_); |
| } |
| |
| void VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) { |
| rtc::CritScope cs(crit_); |
| if (!enabled_) { |
| return; |
| } |
| if (using_external_vad_) { |
| using_external_vad_ = false; |
| return; |
| } |
| |
| RTC_DCHECK_GE(160, audio->num_frames_per_band()); |
| // TODO(ajm): concatenate data in frame buffer here. |
| int vad_ret = WebRtcVad_Process(vad_->state(), sample_rate_hz_, |
| audio->mixed_low_pass_data(), |
| frame_size_samples_); |
| if (vad_ret == 0) { |
| stream_has_voice_ = false; |
| audio->set_activity(AudioFrame::kVadPassive); |
| } else if (vad_ret == 1) { |
| stream_has_voice_ = true; |
| audio->set_activity(AudioFrame::kVadActive); |
| } else { |
| RTC_NOTREACHED(); |
| } |
| } |
| |
| int VoiceDetectionImpl::Enable(bool enable) { |
| rtc::CritScope cs(crit_); |
| if (enabled_ != enable) { |
| enabled_ = enable; |
| Initialize(sample_rate_hz_); |
| } |
| return AudioProcessing::kNoError; |
| } |
| |
| bool VoiceDetectionImpl::is_enabled() const { |
| rtc::CritScope cs(crit_); |
| return enabled_; |
| } |
| |
| int VoiceDetectionImpl::set_stream_has_voice(bool has_voice) { |
| rtc::CritScope cs(crit_); |
| using_external_vad_ = true; |
| stream_has_voice_ = has_voice; |
| return AudioProcessing::kNoError; |
| } |
| |
| bool VoiceDetectionImpl::stream_has_voice() const { |
| rtc::CritScope cs(crit_); |
| // TODO(ajm): enable this assertion? |
| //RTC_DCHECK(using_external_vad_ || is_component_enabled()); |
| return stream_has_voice_; |
| } |
| |
| int VoiceDetectionImpl::set_likelihood(VoiceDetection::Likelihood likelihood) { |
| rtc::CritScope cs(crit_); |
| likelihood_ = likelihood; |
| if (enabled_) { |
| int mode = 2; |
| switch (likelihood) { |
| case VoiceDetection::kVeryLowLikelihood: |
| mode = 3; |
| break; |
| case VoiceDetection::kLowLikelihood: |
| mode = 2; |
| break; |
| case VoiceDetection::kModerateLikelihood: |
| mode = 1; |
| break; |
| case VoiceDetection::kHighLikelihood: |
| mode = 0; |
| break; |
| default: |
| RTC_NOTREACHED(); |
| break; |
| } |
| int error = WebRtcVad_set_mode(vad_->state(), mode); |
| RTC_DCHECK_EQ(0, error); |
| } |
| return AudioProcessing::kNoError; |
| } |
| |
| VoiceDetection::Likelihood VoiceDetectionImpl::likelihood() const { |
| rtc::CritScope cs(crit_); |
| return likelihood_; |
| } |
| |
| int VoiceDetectionImpl::set_frame_size_ms(int size) { |
| rtc::CritScope cs(crit_); |
| RTC_DCHECK_EQ(10, size); // TODO(ajm): remove when supported. |
| frame_size_ms_ = size; |
| Initialize(sample_rate_hz_); |
| return AudioProcessing::kNoError; |
| } |
| |
| int VoiceDetectionImpl::frame_size_ms() const { |
| rtc::CritScope cs(crit_); |
| return frame_size_ms_; |
| } |
| } // namespace webrtc |