| /* |
| * 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/level_controller/down_sampler.h" |
| |
| #include <string.h> |
| #include <algorithm> |
| #include <vector> |
| |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "modules/audio_processing/level_controller/biquad_filter.h" |
| #include "modules/audio_processing/logging/apm_data_dumper.h" |
| #include "rtc_base/checks.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 |