1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "audio/utility/channel_mixer.h"
12
13 #include "audio/utility/channel_mixing_matrix.h"
14 #include "rtc_base/checks.h"
15 #include "rtc_base/logging.h"
16 #include "rtc_base/numerics/safe_conversions.h"
17
18 namespace webrtc {
19
ChannelMixer(ChannelLayout input_layout,ChannelLayout output_layout)20 ChannelMixer::ChannelMixer(ChannelLayout input_layout,
21 ChannelLayout output_layout)
22 : input_layout_(input_layout),
23 output_layout_(output_layout),
24 input_channels_(ChannelLayoutToChannelCount(input_layout)),
25 output_channels_(ChannelLayoutToChannelCount(output_layout)) {
26 // Create the transformation matrix.
27 ChannelMixingMatrix matrix_builder(input_layout_, input_channels_,
28 output_layout_, output_channels_);
29 remapping_ = matrix_builder.CreateTransformationMatrix(&matrix_);
30 }
31
32 ChannelMixer::~ChannelMixer() = default;
33
Transform(AudioFrame * frame)34 void ChannelMixer::Transform(AudioFrame* frame) {
35 RTC_DCHECK(frame);
36 RTC_DCHECK_EQ(matrix_[0].size(), static_cast<size_t>(input_channels_));
37 RTC_DCHECK_EQ(matrix_.size(), static_cast<size_t>(output_channels_));
38
39 // Leave the audio frame intact if the channel layouts for in and out are
40 // identical.
41 if (input_layout_ == output_layout_) {
42 return;
43 }
44
45 if (IsUpMixing()) {
46 RTC_CHECK_LE(frame->samples_per_channel() * output_channels_,
47 frame->max_16bit_samples());
48 }
49
50 // Only change the number of output channels if the audio frame is muted.
51 if (frame->muted()) {
52 frame->num_channels_ = output_channels_;
53 frame->channel_layout_ = output_layout_;
54 return;
55 }
56
57 const int16_t* in_audio = frame->data();
58
59 // Only allocate fresh memory at first access or if the required size has
60 // increased.
61 // TODO(henrika): we might be able to do downmixing in-place and thereby avoid
62 // extra memory allocation and a memcpy.
63 const size_t num_elements = frame->samples_per_channel() * output_channels_;
64 if (audio_vector_ == nullptr || num_elements > audio_vector_size_) {
65 audio_vector_.reset(new int16_t[num_elements]);
66 audio_vector_size_ = num_elements;
67 }
68 int16_t* out_audio = audio_vector_.get();
69
70 // Modify the number of channels by creating a weighted sum of input samples
71 // where the weights (scale factors) for each output sample are given by the
72 // transformation matrix.
73 for (size_t i = 0; i < frame->samples_per_channel(); i++) {
74 for (size_t output_ch = 0; output_ch < output_channels_; ++output_ch) {
75 float acc_value = 0.0f;
76 for (size_t input_ch = 0; input_ch < input_channels_; ++input_ch) {
77 const float scale = matrix_[output_ch][input_ch];
78 // Scale should always be positive.
79 RTC_DCHECK_GE(scale, 0);
80 // Each output sample is a weighted sum of input samples.
81 acc_value += scale * in_audio[i * input_channels_ + input_ch];
82 }
83 const size_t index = output_channels_ * i + output_ch;
84 RTC_CHECK_LE(index, audio_vector_size_);
85 out_audio[index] = rtc::saturated_cast<int16_t>(acc_value);
86 }
87 }
88
89 // Update channel information.
90 frame->num_channels_ = output_channels_;
91 frame->channel_layout_ = output_layout_;
92
93 // Copy the output result to the audio frame in `frame`.
94 memcpy(
95 frame->mutable_data(), out_audio,
96 sizeof(int16_t) * frame->samples_per_channel() * frame->num_channels());
97 }
98
99 } // namespace webrtc
100