webrtc/modules/audio_processing/noise_suppression_impl.cc
Per Åhgren 928146f546 Removing all external access to the integer sample data in AudioBuffer
This CL removes all external access to the integer sample data in the
AudioBuffer class. It also removes the API in AudioBuffer that provides this.

The purpose of this is to pave the way for removing the sample
duplicating and implicit conversions between integer and floating point
sample formats which is done inside the AudioBuffer.

Bug: webrtc:10882
Change-Id: I1438b691bcef98278aef8e3c63624c367c2d12e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149162
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28912}
2019-08-20 08:36:47 +00:00

219 lines
6.3 KiB
C++

/*
* 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 "modules/audio_processing/noise_suppression_impl.h"
#include "modules/audio_processing/audio_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#if defined(WEBRTC_NS_FLOAT)
#include "modules/audio_processing/ns/noise_suppression.h"
#define NS_CREATE WebRtcNs_Create
#define NS_FREE WebRtcNs_Free
#define NS_INIT WebRtcNs_Init
#define NS_SET_POLICY WebRtcNs_set_policy
typedef NsHandle NsState;
#elif defined(WEBRTC_NS_FIXED)
#include "modules/audio_processing/ns/noise_suppression_x.h"
#define NS_CREATE WebRtcNsx_Create
#define NS_FREE WebRtcNsx_Free
#define NS_INIT WebRtcNsx_Init
#define NS_SET_POLICY WebRtcNsx_set_policy
typedef NsxHandle NsState;
#endif
namespace webrtc {
class NoiseSuppressionImpl::Suppressor {
public:
explicit Suppressor(int sample_rate_hz) {
state_ = NS_CREATE();
RTC_CHECK(state_);
int error = NS_INIT(state_, sample_rate_hz);
RTC_DCHECK_EQ(0, error);
}
~Suppressor() { NS_FREE(state_); }
NsState* state() { return state_; }
private:
NsState* state_ = nullptr;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Suppressor);
};
NoiseSuppressionImpl::NoiseSuppressionImpl(rtc::CriticalSection* crit)
: crit_(crit) {
RTC_DCHECK(crit);
}
NoiseSuppressionImpl::~NoiseSuppressionImpl() {}
void NoiseSuppressionImpl::Initialize(size_t channels, int sample_rate_hz) {
rtc::CritScope cs(crit_);
channels_ = channels;
sample_rate_hz_ = sample_rate_hz;
std::vector<std::unique_ptr<Suppressor>> new_suppressors;
if (enabled_) {
new_suppressors.resize(channels);
for (size_t i = 0; i < channels; i++) {
new_suppressors[i].reset(new Suppressor(sample_rate_hz));
}
}
suppressors_.swap(new_suppressors);
set_level(level_);
}
void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
RTC_DCHECK(audio);
#if defined(WEBRTC_NS_FLOAT)
rtc::CritScope cs(crit_);
if (!enabled_) {
return;
}
RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
for (size_t i = 0; i < suppressors_.size(); i++) {
WebRtcNs_Analyze(suppressors_[i]->state(),
audio->split_bands_const_f(i)[kBand0To8kHz]);
}
#endif
}
void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
RTC_DCHECK(audio);
rtc::CritScope cs(crit_);
if (!enabled_) {
return;
}
RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
for (size_t i = 0; i < suppressors_.size(); i++) {
#if defined(WEBRTC_NS_FLOAT)
WebRtcNs_Process(suppressors_[i]->state(), audio->split_bands_const_f(i),
audio->num_bands(), audio->split_bands_f(i));
#elif defined(WEBRTC_NS_FIXED)
int16_t split_band_data[AudioBuffer::kMaxNumBands]
[AudioBuffer::kMaxSplitFrameLength];
int16_t* split_bands[AudioBuffer::kMaxNumBands] = {
split_band_data[0], split_band_data[1], split_band_data[2]};
audio->CopySplitChannelDataTo(i, split_bands);
WebRtcNsx_Process(suppressors_[i]->state(), split_bands, audio->num_bands(),
split_bands);
audio->CopySplitChannelDataFrom(i, split_bands);
#endif
}
}
int NoiseSuppressionImpl::Enable(bool enable) {
rtc::CritScope cs(crit_);
if (enabled_ != enable) {
enabled_ = enable;
Initialize(channels_, sample_rate_hz_);
}
return AudioProcessing::kNoError;
}
bool NoiseSuppressionImpl::is_enabled() const {
rtc::CritScope cs(crit_);
return enabled_;
}
int NoiseSuppressionImpl::set_level(Level level) {
int policy = 1;
switch (level) {
case NoiseSuppression::kLow:
policy = 0;
break;
case NoiseSuppression::kModerate:
policy = 1;
break;
case NoiseSuppression::kHigh:
policy = 2;
break;
case NoiseSuppression::kVeryHigh:
policy = 3;
break;
default:
RTC_NOTREACHED();
}
rtc::CritScope cs(crit_);
level_ = level;
for (auto& suppressor : suppressors_) {
int error = NS_SET_POLICY(suppressor->state(), policy);
RTC_DCHECK_EQ(0, error);
}
return AudioProcessing::kNoError;
}
NoiseSuppression::Level NoiseSuppressionImpl::level() const {
rtc::CritScope cs(crit_);
return level_;
}
float NoiseSuppressionImpl::speech_probability() const {
rtc::CritScope cs(crit_);
#if defined(WEBRTC_NS_FLOAT)
float probability_average = 0.0f;
for (auto& suppressor : suppressors_) {
probability_average +=
WebRtcNs_prior_speech_probability(suppressor->state());
}
if (!suppressors_.empty()) {
probability_average /= suppressors_.size();
}
return probability_average;
#elif defined(WEBRTC_NS_FIXED)
// TODO(peah): Returning error code as a float! Remove this.
// Currently not available for the fixed point implementation.
return AudioProcessing::kUnsupportedFunctionError;
#endif
}
std::vector<float> NoiseSuppressionImpl::NoiseEstimate() {
rtc::CritScope cs(crit_);
std::vector<float> noise_estimate;
#if defined(WEBRTC_NS_FLOAT)
const float kNumChannelsFraction = 1.f / suppressors_.size();
noise_estimate.assign(WebRtcNs_num_freq(), 0.f);
for (auto& suppressor : suppressors_) {
const float* noise = WebRtcNs_noise_estimate(suppressor->state());
for (size_t i = 0; i < noise_estimate.size(); ++i) {
noise_estimate[i] += kNumChannelsFraction * noise[i];
}
}
#elif defined(WEBRTC_NS_FIXED)
noise_estimate.assign(WebRtcNsx_num_freq(), 0.f);
for (auto& suppressor : suppressors_) {
int q_noise;
const uint32_t* noise =
WebRtcNsx_noise_estimate(suppressor->state(), &q_noise);
const float kNormalizationFactor =
1.f / ((1 << q_noise) * suppressors_.size());
for (size_t i = 0; i < noise_estimate.size(); ++i) {
noise_estimate[i] += kNormalizationFactor * noise[i];
}
}
#endif
return noise_estimate;
}
size_t NoiseSuppressionImpl::num_noise_bins() {
#if defined(WEBRTC_NS_FLOAT)
return WebRtcNs_num_freq();
#elif defined(WEBRTC_NS_FIXED)
return WebRtcNsx_num_freq();
#endif
}
} // namespace webrtc