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This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
86 lines
2.9 KiB
C++
86 lines
2.9 KiB
C++
/*
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* Copyright (c) 2019 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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#ifndef AUDIO_UTILITY_CHANNEL_MIXER_H_
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#define AUDIO_UTILITY_CHANNEL_MIXER_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <memory>
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#include <vector>
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#include "api/audio/audio_frame.h"
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#include "api/audio/channel_layout.h"
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namespace webrtc {
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// ChannelMixer is for converting audio between channel layouts. The conversion
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// matrix is built upon construction and used during each Transform() call. The
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// algorithm works by generating a conversion matrix mapping each output channel
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// to list of input channels. The transform renders all of the output channels,
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// with each output channel rendered according to a weighted sum of the relevant
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// input channels as defined in the matrix.
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// This file is derived from Chromium's media/base/channel_mixer.h.
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class ChannelMixer {
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public:
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// To mix two channels into one and preserve loudness, we must apply
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// (1 / sqrt(2)) gain to each.
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static constexpr float kHalfPower = 0.707106781186547524401f;
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ChannelMixer(ChannelLayout input_layout, ChannelLayout output_layout);
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~ChannelMixer();
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// Transforms all input channels corresponding to the selected |input_layout|
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// to the number of channels in the selected |output_layout|.
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// Example usage (downmix from stereo to mono):
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//
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// ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_MONO);
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// AudioFrame frame;
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// frame.samples_per_channel_ = 160;
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// frame.num_channels_ = 2;
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// EXPECT_EQ(2u, frame.channels());
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// mixer.Transform(&frame);
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// EXPECT_EQ(1u, frame.channels());
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//
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void Transform(AudioFrame* frame);
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private:
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bool IsUpMixing() const { return output_channels_ > input_channels_; }
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// Selected channel layouts.
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const ChannelLayout input_layout_;
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const ChannelLayout output_layout_;
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// Channel counts for input and output.
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const size_t input_channels_;
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const size_t output_channels_;
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// 2D matrix of output channels to input channels.
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std::vector<std::vector<float> > matrix_;
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// 1D array used as temporary storage during the transformation.
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std::unique_ptr<int16_t[]> audio_vector_;
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// Number of elements allocated for |audio_vector_|.
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size_t audio_vector_size_ = 0;
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// Optimization case for when we can simply remap the input channels to output
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// channels, i.e., when all scaling factors in |matrix_| equals 1.0.
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bool remapping_;
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// Delete the copy constructor and assignment operator.
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ChannelMixer(const ChannelMixer& other) = delete;
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ChannelMixer& operator=(const ChannelMixer& other) = delete;
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};
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} // namespace webrtc
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#endif // AUDIO_UTILITY_CHANNEL_MIXER_H_
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