Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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Mirko Bonadei
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modules/audio_processing/noise_suppression_impl.cc
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modules/audio_processing/noise_suppression_impl.cc
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/*
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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 "webrtc/modules/audio_processing/noise_suppression_impl.h"
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#if defined(WEBRTC_NS_FLOAT)
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#include "webrtc/modules/audio_processing/ns/noise_suppression.h"
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#define NS_CREATE WebRtcNs_Create
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#define NS_FREE WebRtcNs_Free
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#define NS_INIT WebRtcNs_Init
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#define NS_SET_POLICY WebRtcNs_set_policy
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typedef NsHandle NsState;
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#elif defined(WEBRTC_NS_FIXED)
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#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
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#define NS_CREATE WebRtcNsx_Create
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#define NS_FREE WebRtcNsx_Free
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#define NS_INIT WebRtcNsx_Init
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#define NS_SET_POLICY WebRtcNsx_set_policy
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typedef NsxHandle NsState;
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#endif
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namespace webrtc {
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class NoiseSuppressionImpl::Suppressor {
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public:
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explicit Suppressor(int sample_rate_hz) {
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state_ = NS_CREATE();
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RTC_CHECK(state_);
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int error = NS_INIT(state_, sample_rate_hz);
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RTC_DCHECK_EQ(0, error);
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}
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~Suppressor() {
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NS_FREE(state_);
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}
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NsState* state() { return state_; }
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private:
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NsState* state_ = nullptr;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Suppressor);
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};
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NoiseSuppressionImpl::NoiseSuppressionImpl(rtc::CriticalSection* crit)
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: crit_(crit) {
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RTC_DCHECK(crit);
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}
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NoiseSuppressionImpl::~NoiseSuppressionImpl() {}
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void NoiseSuppressionImpl::Initialize(size_t channels, int sample_rate_hz) {
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rtc::CritScope cs(crit_);
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channels_ = channels;
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sample_rate_hz_ = sample_rate_hz;
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std::vector<std::unique_ptr<Suppressor>> new_suppressors;
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if (enabled_) {
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new_suppressors.resize(channels);
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for (size_t i = 0; i < channels; i++) {
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new_suppressors[i].reset(new Suppressor(sample_rate_hz));
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}
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}
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suppressors_.swap(new_suppressors);
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set_level(level_);
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}
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void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
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RTC_DCHECK(audio);
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#if defined(WEBRTC_NS_FLOAT)
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rtc::CritScope cs(crit_);
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if (!enabled_) {
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return;
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}
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RTC_DCHECK_GE(160, audio->num_frames_per_band());
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RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
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for (size_t i = 0; i < suppressors_.size(); i++) {
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WebRtcNs_Analyze(suppressors_[i]->state(),
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audio->split_bands_const_f(i)[kBand0To8kHz]);
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}
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#endif
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}
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void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
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RTC_DCHECK(audio);
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rtc::CritScope cs(crit_);
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if (!enabled_) {
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return;
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}
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RTC_DCHECK_GE(160, audio->num_frames_per_band());
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RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
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for (size_t i = 0; i < suppressors_.size(); i++) {
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#if defined(WEBRTC_NS_FLOAT)
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WebRtcNs_Process(suppressors_[i]->state(),
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audio->split_bands_const_f(i),
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audio->num_bands(),
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audio->split_bands_f(i));
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#elif defined(WEBRTC_NS_FIXED)
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WebRtcNsx_Process(suppressors_[i]->state(),
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audio->split_bands_const(i),
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audio->num_bands(),
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audio->split_bands(i));
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#endif
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}
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}
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int NoiseSuppressionImpl::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(channels_, sample_rate_hz_);
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}
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return AudioProcessing::kNoError;
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}
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bool NoiseSuppressionImpl::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 NoiseSuppressionImpl::set_level(Level level) {
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int policy = 1;
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switch (level) {
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case NoiseSuppression::kLow:
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policy = 0;
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break;
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case NoiseSuppression::kModerate:
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policy = 1;
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break;
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case NoiseSuppression::kHigh:
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policy = 2;
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break;
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case NoiseSuppression::kVeryHigh:
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policy = 3;
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break;
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default:
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RTC_NOTREACHED();
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}
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rtc::CritScope cs(crit_);
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level_ = level;
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for (auto& suppressor : suppressors_) {
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int error = NS_SET_POLICY(suppressor->state(), policy);
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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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NoiseSuppression::Level NoiseSuppressionImpl::level() const {
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rtc::CritScope cs(crit_);
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return level_;
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}
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float NoiseSuppressionImpl::speech_probability() const {
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rtc::CritScope cs(crit_);
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#if defined(WEBRTC_NS_FLOAT)
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float probability_average = 0.0f;
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for (auto& suppressor : suppressors_) {
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probability_average +=
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WebRtcNs_prior_speech_probability(suppressor->state());
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}
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if (!suppressors_.empty()) {
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probability_average /= suppressors_.size();
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}
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return probability_average;
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#elif defined(WEBRTC_NS_FIXED)
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// TODO(peah): Returning error code as a float! Remove this.
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// Currently not available for the fixed point implementation.
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return AudioProcessing::kUnsupportedFunctionError;
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#endif
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}
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std::vector<float> NoiseSuppressionImpl::NoiseEstimate() {
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rtc::CritScope cs(crit_);
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std::vector<float> noise_estimate;
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#if defined(WEBRTC_NS_FLOAT)
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const float kNumChannelsFraction = 1.f / suppressors_.size();
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noise_estimate.assign(WebRtcNs_num_freq(), 0.f);
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for (auto& suppressor : suppressors_) {
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const float* noise = WebRtcNs_noise_estimate(suppressor->state());
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for (size_t i = 0; i < noise_estimate.size(); ++i) {
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noise_estimate[i] += kNumChannelsFraction * noise[i];
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}
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}
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#elif defined(WEBRTC_NS_FIXED)
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noise_estimate.assign(WebRtcNsx_num_freq(), 0.f);
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for (auto& suppressor : suppressors_) {
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int q_noise;
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const uint32_t* noise = WebRtcNsx_noise_estimate(suppressor->state(),
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&q_noise);
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const float kNormalizationFactor =
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1.f / ((1 << q_noise) * suppressors_.size());
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for (size_t i = 0; i < noise_estimate.size(); ++i) {
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noise_estimate[i] += kNormalizationFactor * noise[i];
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}
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}
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#endif
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return noise_estimate;
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}
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size_t NoiseSuppressionImpl::num_noise_bins() {
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#if defined(WEBRTC_NS_FLOAT)
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return WebRtcNs_num_freq();
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#elif defined(WEBRTC_NS_FIXED)
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return WebRtcNsx_num_freq();
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#endif
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}
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} // namespace webrtc
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