100 lines
3.6 KiB
C++
100 lines
3.6 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 "audio/utility/channel_mixer.h"
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#include "audio/utility/channel_mixing_matrix.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_conversions.h"
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namespace webrtc {
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ChannelMixer::ChannelMixer(ChannelLayout input_layout,
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ChannelLayout output_layout)
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: input_layout_(input_layout),
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output_layout_(output_layout),
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input_channels_(ChannelLayoutToChannelCount(input_layout)),
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output_channels_(ChannelLayoutToChannelCount(output_layout)) {
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// Create the transformation matrix.
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ChannelMixingMatrix matrix_builder(input_layout_, input_channels_,
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output_layout_, output_channels_);
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remapping_ = matrix_builder.CreateTransformationMatrix(&matrix_);
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}
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ChannelMixer::~ChannelMixer() = default;
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void ChannelMixer::Transform(AudioFrame* frame) {
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RTC_DCHECK(frame);
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RTC_DCHECK_EQ(matrix_[0].size(), static_cast<size_t>(input_channels_));
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RTC_DCHECK_EQ(matrix_.size(), static_cast<size_t>(output_channels_));
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// Leave the audio frame intact if the channel layouts for in and out are
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// identical.
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if (input_layout_ == output_layout_) {
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return;
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}
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if (IsUpMixing()) {
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RTC_CHECK_LE(frame->samples_per_channel() * output_channels_,
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frame->max_16bit_samples());
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}
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// Only change the number of output channels if the audio frame is muted.
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if (frame->muted()) {
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frame->num_channels_ = output_channels_;
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frame->channel_layout_ = output_layout_;
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return;
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}
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const int16_t* in_audio = frame->data();
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// Only allocate fresh memory at first access or if the required size has
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// increased.
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// TODO(henrika): we might be able to do downmixing in-place and thereby avoid
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// extra memory allocation and a memcpy.
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const size_t num_elements = frame->samples_per_channel() * output_channels_;
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if (audio_vector_ == nullptr || num_elements > audio_vector_size_) {
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audio_vector_.reset(new int16_t[num_elements]);
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audio_vector_size_ = num_elements;
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}
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int16_t* out_audio = audio_vector_.get();
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// Modify the number of channels by creating a weighted sum of input samples
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// where the weights (scale factors) for each output sample are given by the
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// transformation matrix.
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for (size_t i = 0; i < frame->samples_per_channel(); i++) {
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for (size_t output_ch = 0; output_ch < output_channels_; ++output_ch) {
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float acc_value = 0.0f;
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for (size_t input_ch = 0; input_ch < input_channels_; ++input_ch) {
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const float scale = matrix_[output_ch][input_ch];
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// Scale should always be positive.
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RTC_DCHECK_GE(scale, 0);
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// Each output sample is a weighted sum of input samples.
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acc_value += scale * in_audio[i * input_channels_ + input_ch];
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}
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const size_t index = output_channels_ * i + output_ch;
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RTC_CHECK_LE(index, audio_vector_size_);
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out_audio[index] = rtc::saturated_cast<int16_t>(acc_value);
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}
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}
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// Update channel information.
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frame->num_channels_ = output_channels_;
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frame->channel_layout_ = output_layout_;
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// Copy the output result to the audio frame in |frame|.
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memcpy(
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frame->mutable_data(), out_audio,
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sizeof(int16_t) * frame->samples_per_channel() * frame->num_channels());
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}
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} // namespace webrtc
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