Line data Source code
1 : /*
2 : * Copyright (c) 2014 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 "webrtc/common_audio/audio_converter.h"
12 :
13 : #include <cstring>
14 : #include <memory>
15 : #include <utility>
16 : #include <vector>
17 :
18 : #include "webrtc/base/checks.h"
19 : #include "webrtc/base/safe_conversions.h"
20 : #include "webrtc/common_audio/channel_buffer.h"
21 : #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
22 :
23 : using rtc::checked_cast;
24 :
25 : namespace webrtc {
26 :
27 : class CopyConverter : public AudioConverter {
28 : public:
29 0 : CopyConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
30 : size_t dst_frames)
31 0 : : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
32 0 : ~CopyConverter() override {};
33 :
34 0 : void Convert(const float* const* src, size_t src_size, float* const* dst,
35 : size_t dst_capacity) override {
36 0 : CheckSizes(src_size, dst_capacity);
37 0 : if (src != dst) {
38 0 : for (size_t i = 0; i < src_channels(); ++i)
39 0 : std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
40 : }
41 0 : }
42 : };
43 :
44 : class UpmixConverter : public AudioConverter {
45 : public:
46 0 : UpmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
47 : size_t dst_frames)
48 0 : : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
49 0 : ~UpmixConverter() override {};
50 :
51 0 : void Convert(const float* const* src, size_t src_size, float* const* dst,
52 : size_t dst_capacity) override {
53 0 : CheckSizes(src_size, dst_capacity);
54 0 : for (size_t i = 0; i < dst_frames(); ++i) {
55 0 : const float value = src[0][i];
56 0 : for (size_t j = 0; j < dst_channels(); ++j)
57 0 : dst[j][i] = value;
58 : }
59 0 : }
60 : };
61 :
62 : class DownmixConverter : public AudioConverter {
63 : public:
64 0 : DownmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
65 : size_t dst_frames)
66 0 : : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
67 0 : }
68 0 : ~DownmixConverter() override {};
69 :
70 0 : void Convert(const float* const* src, size_t src_size, float* const* dst,
71 : size_t dst_capacity) override {
72 0 : CheckSizes(src_size, dst_capacity);
73 0 : float* dst_mono = dst[0];
74 0 : for (size_t i = 0; i < src_frames(); ++i) {
75 0 : float sum = 0;
76 0 : for (size_t j = 0; j < src_channels(); ++j)
77 0 : sum += src[j][i];
78 0 : dst_mono[i] = sum / src_channels();
79 : }
80 0 : }
81 : };
82 :
83 : class ResampleConverter : public AudioConverter {
84 : public:
85 0 : ResampleConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
86 : size_t dst_frames)
87 0 : : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
88 0 : resamplers_.reserve(src_channels);
89 0 : for (size_t i = 0; i < src_channels; ++i)
90 0 : resamplers_.push_back(std::unique_ptr<PushSincResampler>(
91 0 : new PushSincResampler(src_frames, dst_frames)));
92 0 : }
93 0 : ~ResampleConverter() override {};
94 :
95 0 : void Convert(const float* const* src, size_t src_size, float* const* dst,
96 : size_t dst_capacity) override {
97 0 : CheckSizes(src_size, dst_capacity);
98 0 : for (size_t i = 0; i < resamplers_.size(); ++i)
99 0 : resamplers_[i]->Resample(src[i], src_frames(), dst[i], dst_frames());
100 0 : }
101 :
102 : private:
103 : std::vector<std::unique_ptr<PushSincResampler>> resamplers_;
104 : };
105 :
106 : // Apply a vector of converters in serial, in the order given. At least two
107 : // converters must be provided.
108 : class CompositionConverter : public AudioConverter {
109 : public:
110 0 : CompositionConverter(std::vector<std::unique_ptr<AudioConverter>> converters)
111 0 : : converters_(std::move(converters)) {
112 0 : RTC_CHECK_GE(converters_.size(), 2);
113 : // We need an intermediate buffer after every converter.
114 0 : for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
115 0 : buffers_.push_back(
116 0 : std::unique_ptr<ChannelBuffer<float>>(new ChannelBuffer<float>(
117 0 : (*it)->dst_frames(), (*it)->dst_channels())));
118 0 : }
119 0 : ~CompositionConverter() override {};
120 :
121 0 : void Convert(const float* const* src, size_t src_size, float* const* dst,
122 : size_t dst_capacity) override {
123 0 : converters_.front()->Convert(src, src_size, buffers_.front()->channels(),
124 0 : buffers_.front()->size());
125 0 : for (size_t i = 2; i < converters_.size(); ++i) {
126 0 : auto& src_buffer = buffers_[i - 2];
127 0 : auto& dst_buffer = buffers_[i - 1];
128 0 : converters_[i]->Convert(src_buffer->channels(),
129 : src_buffer->size(),
130 : dst_buffer->channels(),
131 0 : dst_buffer->size());
132 : }
133 0 : converters_.back()->Convert(buffers_.back()->channels(),
134 0 : buffers_.back()->size(), dst, dst_capacity);
135 0 : }
136 :
137 : private:
138 : std::vector<std::unique_ptr<AudioConverter>> converters_;
139 : std::vector<std::unique_ptr<ChannelBuffer<float>>> buffers_;
140 : };
141 :
142 0 : std::unique_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
143 : size_t src_frames,
144 : size_t dst_channels,
145 : size_t dst_frames) {
146 0 : std::unique_ptr<AudioConverter> sp;
147 0 : if (src_channels > dst_channels) {
148 0 : if (src_frames != dst_frames) {
149 0 : std::vector<std::unique_ptr<AudioConverter>> converters;
150 0 : converters.push_back(std::unique_ptr<AudioConverter>(new DownmixConverter(
151 0 : src_channels, src_frames, dst_channels, src_frames)));
152 : converters.push_back(
153 0 : std::unique_ptr<AudioConverter>(new ResampleConverter(
154 0 : dst_channels, src_frames, dst_channels, dst_frames)));
155 0 : sp.reset(new CompositionConverter(std::move(converters)));
156 : } else {
157 : sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
158 0 : dst_frames));
159 : }
160 0 : } else if (src_channels < dst_channels) {
161 0 : if (src_frames != dst_frames) {
162 0 : std::vector<std::unique_ptr<AudioConverter>> converters;
163 : converters.push_back(
164 0 : std::unique_ptr<AudioConverter>(new ResampleConverter(
165 0 : src_channels, src_frames, src_channels, dst_frames)));
166 0 : converters.push_back(std::unique_ptr<AudioConverter>(new UpmixConverter(
167 0 : src_channels, dst_frames, dst_channels, dst_frames)));
168 0 : sp.reset(new CompositionConverter(std::move(converters)));
169 : } else {
170 : sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
171 0 : dst_frames));
172 : }
173 0 : } else if (src_frames != dst_frames) {
174 : sp.reset(new ResampleConverter(src_channels, src_frames, dst_channels,
175 0 : dst_frames));
176 : } else {
177 : sp.reset(new CopyConverter(src_channels, src_frames, dst_channels,
178 0 : dst_frames));
179 : }
180 :
181 0 : return sp;
182 : }
183 :
184 : // For CompositionConverter.
185 0 : AudioConverter::AudioConverter()
186 : : src_channels_(0),
187 : src_frames_(0),
188 : dst_channels_(0),
189 0 : dst_frames_(0) {}
190 :
191 0 : AudioConverter::AudioConverter(size_t src_channels, size_t src_frames,
192 0 : size_t dst_channels, size_t dst_frames)
193 : : src_channels_(src_channels),
194 : src_frames_(src_frames),
195 : dst_channels_(dst_channels),
196 0 : dst_frames_(dst_frames) {
197 0 : RTC_CHECK(dst_channels == src_channels || dst_channels == 1 ||
198 0 : src_channels == 1);
199 0 : }
200 :
201 0 : void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
202 0 : RTC_CHECK_EQ(src_size, src_channels() * src_frames());
203 0 : RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames());
204 0 : }
205 :
206 : } // namespace webrtc
|
