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/*
* 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 "webrtc/modules/audio_processing/level_controller/down_sampler.h"
#include <string.h>
#include <algorithm>
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/level_controller/biquad_filter.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
namespace webrtc {
namespace {
// 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 == AudioProcessing::kSampleRate8kHz ||
sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
sample_rate_hz == AudioProcessing::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_ == AudioProcessing::kSampleRate16kHz) {
low_pass_filter_.Initialize(kLowPassFilterCoefficients_16kHz);
} else if (sample_rate_hz_ == AudioProcessing::kSampleRate32kHz) {
low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz);
} else if (sample_rate_hz_ == AudioProcessing::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_ * AudioProcessing::kChunkSizeMs / 1000,
in.size());
RTC_DCHECK_EQ(
AudioProcessing::kSampleRate8kHz * AudioProcessing::kChunkSizeMs / 1000,
out.size());
const size_t kMaxNumFrames =
AudioProcessing::kSampleRate48kHz * AudioProcessing::kChunkSizeMs / 1000;
float x[kMaxNumFrames];
// Band-limit the signal to 4 kHz.
if (sample_rate_hz_ != AudioProcessing::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,
AudioProcessing::kSampleRate8kHz, 1);
}
} // namespace webrtc
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