2250b05778
Two new classes are added to WebRTC from Chrome: ChannelMixer and ChannelMixingMatrix but they are not yet utilized in the audio path for WebRTC. The idea is to utilize these new classes when adding support for multi- channel encoding/decoding in WebRTC/Chrome. Adds support for a new enumerator call webrtc::ChannelLayout and some helper methods which maps between channel layout and number of channels. These parts are also copied from Chrome. Minor (cosmetic) changes are also done on the AudioFrame to prepare for upcoming work. Bug: webrtc:10783 Change-Id: I6cd7a13a3bc1c8bbfa19bc974c7a011d22d19197 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141674 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28482}
100 lines
3.6 KiB
C++
100 lines
3.6 KiB
C++
/*
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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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