webrtc/modules/audio_processing/agc2/down_sampler.cc
Alex Loiko 4ed47d0190 Noise level estimation for AGC2.
We put back the old noise estimator from LevelController. We add a few
new unit tests. We also re-arrange the code so that it fits with how
it is used in AGC2. The differences are:

1. The NoiseLevelEstimator is now fully self-contained.
2. The NoiseLevelEstimator is responsible for calling SignalClassifier
   and computing the signal energy. Previously the signal type and
   energy were used in several places. It made sense to compute the
   values independently of the noise calculation.
3. Re-initialization doesn't have to be done by the caller.
4. The interface is AudioFrameView instead of AudioBuffer.

# Bots are green, nothing should break internal stuff
NOTRY=True

Bug: webrtc:7494
Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d
Reviewed-on: https://webrtc-review.googlesource.com/66380
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22738}
2018-04-04 18:23:55 +00:00

98 lines
3.3 KiB
C++

/*
* Copyright (c) 2016 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/agc2/down_sampler.h"
#include <string.h>
#include <algorithm>
#include "modules/audio_processing/agc2/biquad_filter.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
constexpr int kChunkSizeMs = 10;
constexpr int kSampleRate8kHz = 8000;
constexpr int kSampleRate16kHz = 16000;
constexpr int kSampleRate32kHz = 32000;
constexpr int kSampleRate48kHz = 48000;
// Bandlimiter coefficients computed based on that only
// the first 40 bins of the spectrum for the downsampled
// signal are used.
// [B,A] = butter(2,(41/64*4000)/8000)
const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_16kHz = {
{0.1455f, 0.2911f, 0.1455f},
{-0.6698f, 0.2520f}};
// [B,A] = butter(2,(41/64*4000)/16000)
const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_32kHz = {
{0.0462f, 0.0924f, 0.0462f},
{-1.3066f, 0.4915f}};
// [B,A] = butter(2,(41/64*4000)/24000)
const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_48kHz = {
{0.0226f, 0.0452f, 0.0226f},
{-1.5320f, 0.6224f}};
} // namespace
DownSampler::DownSampler(ApmDataDumper* data_dumper)
: data_dumper_(data_dumper) {
Initialize(48000);
}
void DownSampler::Initialize(int sample_rate_hz) {
RTC_DCHECK(
sample_rate_hz == kSampleRate8kHz || sample_rate_hz == kSampleRate16kHz ||
sample_rate_hz == kSampleRate32kHz || sample_rate_hz == kSampleRate48kHz);
sample_rate_hz_ = sample_rate_hz;
down_sampling_factor_ = rtc::CheckedDivExact(sample_rate_hz_, 8000);
/// Note that the down sampling filter is not used if the sample rate is 8
/// kHz.
if (sample_rate_hz_ == kSampleRate16kHz) {
low_pass_filter_.Initialize(kLowPassFilterCoefficients_16kHz);
} else if (sample_rate_hz_ == kSampleRate32kHz) {
low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz);
} else if (sample_rate_hz_ == kSampleRate48kHz) {
low_pass_filter_.Initialize(kLowPassFilterCoefficients_48kHz);
}
}
void DownSampler::DownSample(rtc::ArrayView<const float> in,
rtc::ArrayView<float> out) {
data_dumper_->DumpWav("lc_down_sampler_input", in, sample_rate_hz_, 1);
RTC_DCHECK_EQ(sample_rate_hz_ * kChunkSizeMs / 1000, in.size());
RTC_DCHECK_EQ(kSampleRate8kHz * kChunkSizeMs / 1000, out.size());
const size_t kMaxNumFrames = kSampleRate48kHz * kChunkSizeMs / 1000;
float x[kMaxNumFrames];
// Band-limit the signal to 4 kHz.
if (sample_rate_hz_ != kSampleRate8kHz) {
low_pass_filter_.Process(in, rtc::ArrayView<float>(x, in.size()));
// Downsample the signal.
size_t k = 0;
for (size_t j = 0; j < out.size(); ++j) {
RTC_DCHECK_GT(kMaxNumFrames, k);
out[j] = x[k];
k += down_sampling_factor_;
}
} else {
std::copy(in.data(), in.data() + in.size(), out.data());
}
data_dumper_->DumpWav("lc_down_sampler_output", out, kSampleRate8kHz, 1);
}
} // namespace webrtc