mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
99 lines
3.3 KiB
C++
99 lines
3.3 KiB
C++
/*
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* Copyright (c) 2016 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 "modules/audio_processing/agc2/down_sampler.h"
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#include <string.h>
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#include <algorithm>
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#include "modules/audio_processing/agc2/biquad_filter.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace {
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constexpr int kChunkSizeMs = 10;
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constexpr int kSampleRate8kHz = 8000;
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constexpr int kSampleRate16kHz = 16000;
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constexpr int kSampleRate32kHz = 32000;
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constexpr int kSampleRate48kHz = 48000;
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// Bandlimiter coefficients computed based on that only
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// the first 40 bins of the spectrum for the downsampled
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// signal are used.
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// [B,A] = butter(2,(41/64*4000)/8000)
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const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_16kHz = {
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{0.1455f, 0.2911f, 0.1455f},
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{-0.6698f, 0.2520f}};
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// [B,A] = butter(2,(41/64*4000)/16000)
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const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_32kHz = {
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{0.0462f, 0.0924f, 0.0462f},
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{-1.3066f, 0.4915f}};
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// [B,A] = butter(2,(41/64*4000)/24000)
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const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_48kHz = {
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{0.0226f, 0.0452f, 0.0226f},
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{-1.5320f, 0.6224f}};
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} // namespace
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DownSampler::DownSampler(ApmDataDumper* data_dumper)
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: data_dumper_(data_dumper) {
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Initialize(48000);
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}
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void DownSampler::Initialize(int sample_rate_hz) {
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RTC_DCHECK(
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sample_rate_hz == kSampleRate8kHz || sample_rate_hz == kSampleRate16kHz ||
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sample_rate_hz == kSampleRate32kHz || sample_rate_hz == kSampleRate48kHz);
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sample_rate_hz_ = sample_rate_hz;
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down_sampling_factor_ = rtc::CheckedDivExact(sample_rate_hz_, 8000);
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/// Note that the down sampling filter is not used if the sample rate is 8
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/// kHz.
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if (sample_rate_hz_ == kSampleRate16kHz) {
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low_pass_filter_.Initialize(kLowPassFilterCoefficients_16kHz);
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} else if (sample_rate_hz_ == kSampleRate32kHz) {
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low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz);
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} else if (sample_rate_hz_ == kSampleRate48kHz) {
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low_pass_filter_.Initialize(kLowPassFilterCoefficients_48kHz);
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}
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}
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void DownSampler::DownSample(rtc::ArrayView<const float> in,
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rtc::ArrayView<float> out) {
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data_dumper_->DumpWav("lc_down_sampler_input", in, sample_rate_hz_, 1);
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RTC_DCHECK_EQ(sample_rate_hz_ * kChunkSizeMs / 1000, in.size());
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RTC_DCHECK_EQ(kSampleRate8kHz * kChunkSizeMs / 1000, out.size());
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const size_t kMaxNumFrames = kSampleRate48kHz * kChunkSizeMs / 1000;
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float x[kMaxNumFrames];
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// Band-limit the signal to 4 kHz.
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if (sample_rate_hz_ != kSampleRate8kHz) {
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low_pass_filter_.Process(in, rtc::ArrayView<float>(x, in.size()));
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// Downsample the signal.
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size_t k = 0;
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for (size_t j = 0; j < out.size(); ++j) {
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RTC_DCHECK_GT(kMaxNumFrames, k);
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out[j] = x[k];
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k += down_sampling_factor_;
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}
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} else {
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std::copy(in.data(), in.data() + in.size(), out.data());
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}
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data_dumper_->DumpWav("lc_down_sampler_output", out, kSampleRate8kHz, 1);
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}
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} // namespace webrtc
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