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/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/audio_converter.h"
#include <cstring>
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include "webrtc/system_wrappers/include/scoped_vector.h"
using rtc::checked_cast;
namespace webrtc {
class CopyConverter : public AudioConverter {
public:
CopyConverter(int src_channels, size_t src_frames, int dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
~CopyConverter() override {};
void Convert(const float* const* src, size_t src_size, float* const* dst,
size_t dst_capacity) override {
CheckSizes(src_size, dst_capacity);
if (src != dst) {
for (int i = 0; i < src_channels(); ++i)
std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
}
}
};
class UpmixConverter : public AudioConverter {
public:
UpmixConverter(int src_channels, size_t src_frames, int dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
~UpmixConverter() override {};
void Convert(const float* const* src, size_t src_size, float* const* dst,
size_t dst_capacity) override {
CheckSizes(src_size, dst_capacity);
for (size_t i = 0; i < dst_frames(); ++i) {
const float value = src[0][i];
for (int j = 0; j < dst_channels(); ++j)
dst[j][i] = value;
}
}
};
class DownmixConverter : public AudioConverter {
public:
DownmixConverter(int src_channels, size_t src_frames, int dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
}
~DownmixConverter() override {};
void Convert(const float* const* src, size_t src_size, float* const* dst,
size_t dst_capacity) override {
CheckSizes(src_size, dst_capacity);
float* dst_mono = dst[0];
for (size_t i = 0; i < src_frames(); ++i) {
float sum = 0;
for (int j = 0; j < src_channels(); ++j)
sum += src[j][i];
dst_mono[i] = sum / src_channels();
}
}
};
class ResampleConverter : public AudioConverter {
public:
ResampleConverter(int src_channels, size_t src_frames, int dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
resamplers_.reserve(src_channels);
for (int i = 0; i < src_channels; ++i)
resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
}
~ResampleConverter() override {};
void Convert(const float* const* src, size_t src_size, float* const* dst,
size_t dst_capacity) override {
CheckSizes(src_size, dst_capacity);
for (size_t i = 0; i < resamplers_.size(); ++i)
resamplers_[i]->Resample(src[i], src_frames(), dst[i], dst_frames());
}
private:
ScopedVector<PushSincResampler> resamplers_;
};
// Apply a vector of converters in serial, in the order given. At least two
// converters must be provided.
class CompositionConverter : public AudioConverter {
public:
CompositionConverter(ScopedVector<AudioConverter> converters)
: converters_(converters.Pass()) {
RTC_CHECK_GE(converters_.size(), 2u);
// We need an intermediate buffer after every converter.
for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
buffers_.push_back(new ChannelBuffer<float>((*it)->dst_frames(),
(*it)->dst_channels()));
}
~CompositionConverter() override {};
void Convert(const float* const* src, size_t src_size, float* const* dst,
size_t dst_capacity) override {
converters_.front()->Convert(src, src_size, buffers_.front()->channels(),
buffers_.front()->size());
for (size_t i = 2; i < converters_.size(); ++i) {
auto src_buffer = buffers_[i - 2];
auto dst_buffer = buffers_[i - 1];
converters_[i]->Convert(src_buffer->channels(),
src_buffer->size(),
dst_buffer->channels(),
dst_buffer->size());
}
converters_.back()->Convert(buffers_.back()->channels(),
buffers_.back()->size(), dst, dst_capacity);
}
private:
ScopedVector<AudioConverter> converters_;
ScopedVector<ChannelBuffer<float>> buffers_;
};
rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
size_t src_frames,
int dst_channels,
size_t dst_frames) {
rtc::scoped_ptr<AudioConverter> sp;
if (src_channels > dst_channels) {
if (src_frames != dst_frames) {
ScopedVector<AudioConverter> converters;
converters.push_back(new DownmixConverter(src_channels, src_frames,
dst_channels, src_frames));
converters.push_back(new ResampleConverter(dst_channels, src_frames,
dst_channels, dst_frames));
sp.reset(new CompositionConverter(converters.Pass()));
} else {
sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
dst_frames));
}
} else if (src_channels < dst_channels) {
if (src_frames != dst_frames) {
ScopedVector<AudioConverter> converters;
converters.push_back(new ResampleConverter(src_channels, src_frames,
src_channels, dst_frames));
converters.push_back(new UpmixConverter(src_channels, dst_frames,
dst_channels, dst_frames));
sp.reset(new CompositionConverter(converters.Pass()));
} else {
sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
dst_frames));
}
} else if (src_frames != dst_frames) {
sp.reset(new ResampleConverter(src_channels, src_frames, dst_channels,
dst_frames));
} else {
sp.reset(new CopyConverter(src_channels, src_frames, dst_channels,
dst_frames));
}
return sp.Pass();
}
// For CompositionConverter.
AudioConverter::AudioConverter()
: src_channels_(0),
src_frames_(0),
dst_channels_(0),
dst_frames_(0) {}
AudioConverter::AudioConverter(int src_channels, size_t src_frames,
int dst_channels, size_t dst_frames)
: src_channels_(src_channels),
src_frames_(src_frames),
dst_channels_(dst_channels),
dst_frames_(dst_frames) {
RTC_CHECK(dst_channels == src_channels || dst_channels == 1 ||
src_channels == 1);
}
void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
RTC_CHECK_EQ(src_size, src_channels() * src_frames());
RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames());
}
} // namespace webrtc
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