395 lines
14 KiB
C++
395 lines
14 KiB
C++
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/*
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* Copyright (c) 2012 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/audio_buffer.h"
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#include <string.h>
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#include <cstdint>
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#include "common_audio/channel_buffer.h"
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#include "common_audio/resampler/push_sinc_resampler.h"
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#include "modules/audio_processing/splitting_filter.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 size_t kSamplesPer32kHzChannel = 320;
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constexpr size_t kSamplesPer48kHzChannel = 480;
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size_t NumBandsFromFramesPerChannel(size_t num_frames) {
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if (num_frames == kSamplesPer32kHzChannel) {
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return 2;
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}
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if (num_frames == kSamplesPer48kHzChannel) {
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return 3;
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}
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return 1;
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}
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} // namespace
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AudioBuffer::AudioBuffer(size_t input_rate,
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size_t input_num_channels,
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size_t buffer_rate,
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size_t buffer_num_channels,
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size_t output_rate,
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size_t output_num_channels)
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: input_num_frames_(static_cast<int>(input_rate) / 100),
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input_num_channels_(input_num_channels),
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buffer_num_frames_(static_cast<int>(buffer_rate) / 100),
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buffer_num_channels_(buffer_num_channels),
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output_num_frames_(static_cast<int>(output_rate) / 100),
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output_num_channels_(0),
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num_channels_(buffer_num_channels),
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num_bands_(NumBandsFromFramesPerChannel(buffer_num_frames_)),
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num_split_frames_(rtc::CheckedDivExact(buffer_num_frames_, num_bands_)),
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data_(
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new ChannelBuffer<float>(buffer_num_frames_, buffer_num_channels_)) {
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RTC_DCHECK_GT(input_num_frames_, 0);
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RTC_DCHECK_GT(buffer_num_frames_, 0);
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RTC_DCHECK_GT(output_num_frames_, 0);
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RTC_DCHECK_GT(input_num_channels_, 0);
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RTC_DCHECK_GT(buffer_num_channels_, 0);
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RTC_DCHECK_LE(buffer_num_channels_, input_num_channels_);
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const bool input_resampling_needed = input_num_frames_ != buffer_num_frames_;
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const bool output_resampling_needed =
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output_num_frames_ != buffer_num_frames_;
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if (input_resampling_needed) {
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for (size_t i = 0; i < buffer_num_channels_; ++i) {
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input_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
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new PushSincResampler(input_num_frames_, buffer_num_frames_)));
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}
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}
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if (output_resampling_needed) {
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for (size_t i = 0; i < buffer_num_channels_; ++i) {
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output_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
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new PushSincResampler(buffer_num_frames_, output_num_frames_)));
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}
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}
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if (num_bands_ > 1) {
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split_data_.reset(new ChannelBuffer<float>(
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buffer_num_frames_, buffer_num_channels_, num_bands_));
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splitting_filter_.reset(new SplittingFilter(
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buffer_num_channels_, num_bands_, buffer_num_frames_));
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}
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}
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AudioBuffer::~AudioBuffer() {}
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void AudioBuffer::set_downmixing_to_specific_channel(size_t channel) {
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downmix_by_averaging_ = false;
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RTC_DCHECK_GT(input_num_channels_, channel);
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channel_for_downmixing_ = std::min(channel, input_num_channels_ - 1);
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}
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void AudioBuffer::set_downmixing_by_averaging() {
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downmix_by_averaging_ = true;
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}
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void AudioBuffer::CopyFrom(const float* const* stacked_data,
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const StreamConfig& stream_config) {
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RTC_DCHECK_EQ(stream_config.num_frames(), input_num_frames_);
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RTC_DCHECK_EQ(stream_config.num_channels(), input_num_channels_);
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RestoreNumChannels();
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const bool downmix_needed = input_num_channels_ > 1 && num_channels_ == 1;
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const bool resampling_needed = input_num_frames_ != buffer_num_frames_;
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if (downmix_needed) {
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RTC_DCHECK_GE(kMaxSamplesPerChannel10ms, input_num_frames_);
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std::array<float, kMaxSamplesPerChannel10ms> downmix;
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if (downmix_by_averaging_) {
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const float kOneByNumChannels = 1.f / input_num_channels_;
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for (size_t i = 0; i < input_num_frames_; ++i) {
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float value = stacked_data[0][i];
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for (size_t j = 1; j < input_num_channels_; ++j) {
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value += stacked_data[j][i];
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}
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downmix[i] = value * kOneByNumChannels;
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}
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}
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const float* downmixed_data = downmix_by_averaging_
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? downmix.data()
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: stacked_data[channel_for_downmixing_];
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if (resampling_needed) {
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input_resamplers_[0]->Resample(downmixed_data, input_num_frames_,
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data_->channels()[0], buffer_num_frames_);
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}
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const float* data_to_convert =
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resampling_needed ? data_->channels()[0] : downmixed_data;
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FloatToFloatS16(data_to_convert, buffer_num_frames_, data_->channels()[0]);
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} else {
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if (resampling_needed) {
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for (size_t i = 0; i < num_channels_; ++i) {
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input_resamplers_[i]->Resample(stacked_data[i], input_num_frames_,
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data_->channels()[i],
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buffer_num_frames_);
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FloatToFloatS16(data_->channels()[i], buffer_num_frames_,
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data_->channels()[i]);
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}
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} else {
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for (size_t i = 0; i < num_channels_; ++i) {
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FloatToFloatS16(stacked_data[i], buffer_num_frames_,
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data_->channels()[i]);
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}
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}
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}
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}
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void AudioBuffer::CopyTo(const StreamConfig& stream_config,
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float* const* stacked_data) {
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RTC_DCHECK_EQ(stream_config.num_frames(), output_num_frames_);
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const bool resampling_needed = output_num_frames_ != buffer_num_frames_;
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if (resampling_needed) {
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for (size_t i = 0; i < num_channels_; ++i) {
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FloatS16ToFloat(data_->channels()[i], buffer_num_frames_,
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data_->channels()[i]);
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output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_,
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stacked_data[i], output_num_frames_);
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}
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} else {
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for (size_t i = 0; i < num_channels_; ++i) {
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FloatS16ToFloat(data_->channels()[i], buffer_num_frames_,
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stacked_data[i]);
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}
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}
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for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) {
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memcpy(stacked_data[i], stacked_data[0],
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output_num_frames_ * sizeof(**stacked_data));
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}
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}
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void AudioBuffer::CopyTo(AudioBuffer* buffer) const {
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RTC_DCHECK_EQ(buffer->num_frames(), output_num_frames_);
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const bool resampling_needed = output_num_frames_ != buffer_num_frames_;
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if (resampling_needed) {
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for (size_t i = 0; i < num_channels_; ++i) {
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output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_,
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buffer->channels()[i],
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buffer->num_frames());
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}
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} else {
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for (size_t i = 0; i < num_channels_; ++i) {
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memcpy(buffer->channels()[i], data_->channels()[i],
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buffer_num_frames_ * sizeof(**buffer->channels()));
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}
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}
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for (size_t i = num_channels_; i < buffer->num_channels(); ++i) {
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memcpy(buffer->channels()[i], buffer->channels()[0],
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output_num_frames_ * sizeof(**buffer->channels()));
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}
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}
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void AudioBuffer::RestoreNumChannels() {
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num_channels_ = buffer_num_channels_;
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data_->set_num_channels(buffer_num_channels_);
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if (split_data_.get()) {
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split_data_->set_num_channels(buffer_num_channels_);
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}
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}
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void AudioBuffer::set_num_channels(size_t num_channels) {
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RTC_DCHECK_GE(buffer_num_channels_, num_channels);
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num_channels_ = num_channels;
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data_->set_num_channels(num_channels);
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if (split_data_.get()) {
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split_data_->set_num_channels(num_channels);
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}
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}
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// The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
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void AudioBuffer::CopyFrom(const int16_t* const interleaved_data,
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const StreamConfig& stream_config) {
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RTC_DCHECK_EQ(stream_config.num_channels(), input_num_channels_);
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RTC_DCHECK_EQ(stream_config.num_frames(), input_num_frames_);
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RestoreNumChannels();
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const bool resampling_required = input_num_frames_ != buffer_num_frames_;
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const int16_t* interleaved = interleaved_data;
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if (num_channels_ == 1) {
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if (input_num_channels_ == 1) {
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if (resampling_required) {
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std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
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S16ToFloatS16(interleaved, input_num_frames_, float_buffer.data());
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input_resamplers_[0]->Resample(float_buffer.data(), input_num_frames_,
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data_->channels()[0],
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buffer_num_frames_);
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} else {
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S16ToFloatS16(interleaved, input_num_frames_, data_->channels()[0]);
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}
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} else {
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std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
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float* downmixed_data =
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resampling_required ? float_buffer.data() : data_->channels()[0];
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if (downmix_by_averaging_) {
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for (size_t j = 0, k = 0; j < input_num_frames_; ++j) {
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int32_t sum = 0;
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for (size_t i = 0; i < input_num_channels_; ++i, ++k) {
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sum += interleaved[k];
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}
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downmixed_data[j] = sum / static_cast<int16_t>(input_num_channels_);
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}
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} else {
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for (size_t j = 0, k = channel_for_downmixing_; j < input_num_frames_;
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++j, k += input_num_channels_) {
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downmixed_data[j] = interleaved[k];
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}
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}
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if (resampling_required) {
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input_resamplers_[0]->Resample(downmixed_data, input_num_frames_,
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data_->channels()[0],
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buffer_num_frames_);
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}
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}
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} else {
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auto deinterleave_channel = [](size_t channel, size_t num_channels,
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size_t samples_per_channel, const int16_t* x,
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float* y) {
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for (size_t j = 0, k = channel; j < samples_per_channel;
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++j, k += num_channels) {
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y[j] = x[k];
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}
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};
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if (resampling_required) {
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std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
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for (size_t i = 0; i < num_channels_; ++i) {
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deinterleave_channel(i, num_channels_, input_num_frames_, interleaved,
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float_buffer.data());
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input_resamplers_[i]->Resample(float_buffer.data(), input_num_frames_,
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data_->channels()[i],
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buffer_num_frames_);
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}
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} else {
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for (size_t i = 0; i < num_channels_; ++i) {
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deinterleave_channel(i, num_channels_, input_num_frames_, interleaved,
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data_->channels()[i]);
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}
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}
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}
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}
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void AudioBuffer::CopyTo(const StreamConfig& stream_config,
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int16_t* const interleaved_data) {
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const size_t config_num_channels = stream_config.num_channels();
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RTC_DCHECK(config_num_channels == num_channels_ || num_channels_ == 1);
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RTC_DCHECK_EQ(stream_config.num_frames(), output_num_frames_);
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const bool resampling_required = buffer_num_frames_ != output_num_frames_;
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int16_t* interleaved = interleaved_data;
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if (num_channels_ == 1) {
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std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
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if (resampling_required) {
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output_resamplers_[0]->Resample(data_->channels()[0], buffer_num_frames_,
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float_buffer.data(), output_num_frames_);
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}
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const float* deinterleaved =
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resampling_required ? float_buffer.data() : data_->channels()[0];
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if (config_num_channels == 1) {
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for (size_t j = 0; j < output_num_frames_; ++j) {
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interleaved[j] = FloatS16ToS16(deinterleaved[j]);
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}
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} else {
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for (size_t i = 0, k = 0; i < output_num_frames_; ++i) {
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float tmp = FloatS16ToS16(deinterleaved[i]);
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for (size_t j = 0; j < config_num_channels; ++j, ++k) {
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interleaved[k] = tmp;
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}
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}
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}
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} else {
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auto interleave_channel = [](size_t channel, size_t num_channels,
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size_t samples_per_channel, const float* x,
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int16_t* y) {
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for (size_t k = 0, j = channel; k < samples_per_channel;
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++k, j += num_channels) {
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y[j] = FloatS16ToS16(x[k]);
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}
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};
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if (resampling_required) {
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for (size_t i = 0; i < num_channels_; ++i) {
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std::array<float, kMaxSamplesPerChannel10ms> float_buffer;
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output_resamplers_[i]->Resample(data_->channels()[i],
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buffer_num_frames_, float_buffer.data(),
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output_num_frames_);
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interleave_channel(i, config_num_channels, output_num_frames_,
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float_buffer.data(), interleaved);
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||
|
|
}
|
||
|
|
} else {
|
||
|
|
for (size_t i = 0; i < num_channels_; ++i) {
|
||
|
|
interleave_channel(i, config_num_channels, output_num_frames_,
|
||
|
|
data_->channels()[i], interleaved);
|
||
|
|
}
|
||
|
|
}
|
||
|
|
|
||
|
|
for (size_t i = num_channels_; i < config_num_channels; ++i) {
|
||
|
|
for (size_t j = 0, k = i, n = num_channels_; j < output_num_frames_;
|
||
|
|
++j, k += config_num_channels, n += config_num_channels) {
|
||
|
|
interleaved[k] = interleaved[n];
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
|
||
|
|
void AudioBuffer::SplitIntoFrequencyBands() {
|
||
|
|
splitting_filter_->Analysis(data_.get(), split_data_.get());
|
||
|
|
}
|
||
|
|
|
||
|
|
void AudioBuffer::MergeFrequencyBands() {
|
||
|
|
splitting_filter_->Synthesis(split_data_.get(), data_.get());
|
||
|
|
}
|
||
|
|
|
||
|
|
void AudioBuffer::ExportSplitChannelData(
|
||
|
|
size_t channel,
|
||
|
|
int16_t* const* split_band_data) const {
|
||
|
|
for (size_t k = 0; k < num_bands(); ++k) {
|
||
|
|
const float* band_data = split_bands_const(channel)[k];
|
||
|
|
|
||
|
|
RTC_DCHECK(split_band_data[k]);
|
||
|
|
RTC_DCHECK(band_data);
|
||
|
|
for (size_t i = 0; i < num_frames_per_band(); ++i) {
|
||
|
|
split_band_data[k][i] = FloatS16ToS16(band_data[i]);
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
|
||
|
|
void AudioBuffer::ImportSplitChannelData(
|
||
|
|
size_t channel,
|
||
|
|
const int16_t* const* split_band_data) {
|
||
|
|
for (size_t k = 0; k < num_bands(); ++k) {
|
||
|
|
float* band_data = split_bands(channel)[k];
|
||
|
|
RTC_DCHECK(split_band_data[k]);
|
||
|
|
RTC_DCHECK(band_data);
|
||
|
|
for (size_t i = 0; i < num_frames_per_band(); ++i) {
|
||
|
|
band_data[i] = split_band_data[k][i];
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
|
||
|
|
} // namespace webrtc
|