blob: ab089b10ce14863e8bde82f8e7c62ad73d82dff7 [file] [log] [blame]
/*
* Copyright (c) 2025 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/post_filter.h"
#include <array>
#include <cstddef>
#include <memory>
#include <vector>
#include "api/array_view.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/utility/cascaded_biquad_filter.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
// Removes frequencies above 19.5kHz.
// sos = signal.iirdesign(
// 19200 * 2 / 48000, 19500 * 2 / 48000,
// 3, 20, ftype='cheby2', output="sos")
constexpr std::array<CascadedBiQuadFilter::BiQuadCoefficients, 4>
kPostFilterCoefficients48kHz = {{
{.b = {0.56142156f, 1.11499931f, 0.56142156f},
.a = {1.57914249f, 0.63379496f}},
{.b = {1.00000000f, 1.88944170f, 1.00000000f},
.a = {1.55130066f, 0.68708719f}},
{.b = {1.00000000f, 1.76057310f, 1.00000000f},
.a = {1.53001328f, 0.78591224f}},
{.b = {1.00000000f, 1.67448535f, 1.00000000f},
.a = {1.56506670f, 0.92096576f}},
}};
} // namespace
std::unique_ptr<PostFilter> PostFilter::CreateIfNeeded(int sample_rate_hz,
size_t num_channels) {
if (sample_rate_hz != 48000) {
return nullptr;
}
return std::unique_ptr<PostFilter>(
new PostFilter(kPostFilterCoefficients48kHz, num_channels));
}
PostFilter::PostFilter(
ArrayView<const CascadedBiQuadFilter::BiQuadCoefficients> coefficients,
size_t num_channels) {
RTC_DCHECK(!coefficients.empty());
filters_.resize(num_channels);
for (size_t k = 0; k < filters_.size(); ++k) {
filters_[k].reset(new CascadedBiQuadFilter(coefficients));
}
}
void PostFilter::Process(AudioBuffer& audio) {
RTC_DCHECK_EQ(filters_.size(), audio.num_channels());
for (size_t k = 0; k < audio.num_channels(); ++k) {
ArrayView<float> channel_data =
ArrayView<float>(&audio.channels()[k][0], audio.num_frames());
filters_[k]->Process(channel_data);
}
}
} // namespace webrtc