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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
100 lines
3.6 KiB
C++
100 lines
3.6 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/level_controller/down_sampler.h"
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#include <string.h>
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#include <algorithm>
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#include <vector>
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/level_controller/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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// 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(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
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sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
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sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
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sample_rate_hz == AudioProcessing::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_ == AudioProcessing::kSampleRate16kHz) {
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low_pass_filter_.Initialize(kLowPassFilterCoefficients_16kHz);
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} else if (sample_rate_hz_ == AudioProcessing::kSampleRate32kHz) {
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low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz);
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} else if (sample_rate_hz_ == AudioProcessing::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_ * AudioProcessing::kChunkSizeMs / 1000,
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in.size());
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RTC_DCHECK_EQ(
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AudioProcessing::kSampleRate8kHz * AudioProcessing::kChunkSizeMs / 1000,
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out.size());
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const size_t kMaxNumFrames =
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AudioProcessing::kSampleRate48kHz * AudioProcessing::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_ != AudioProcessing::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,
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AudioProcessing::kSampleRate8kHz, 1);
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}
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} // namespace webrtc
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