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Diffstat (limited to 'third_party/libwebrtc/common_audio/audio_converter.cc')
-rw-r--r-- | third_party/libwebrtc/common_audio/audio_converter.cc | 219 |
1 files changed, 219 insertions, 0 deletions
diff --git a/third_party/libwebrtc/common_audio/audio_converter.cc b/third_party/libwebrtc/common_audio/audio_converter.cc new file mode 100644 index 0000000000..485ec80c56 --- /dev/null +++ b/third_party/libwebrtc/common_audio/audio_converter.cc @@ -0,0 +1,219 @@ +/* + * 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 "common_audio/audio_converter.h" + +#include <cstring> +#include <memory> +#include <utility> +#include <vector> + +#include "common_audio/channel_buffer.h" +#include "common_audio/resampler/push_sinc_resampler.h" +#include "rtc_base/checks.h" +#include "rtc_base/numerics/safe_conversions.h" + +namespace webrtc { + +class CopyConverter : public AudioConverter { + public: + CopyConverter(size_t src_channels, + size_t src_frames, + size_t 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 (size_t i = 0; i < src_channels(); ++i) + std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i])); + } + } +}; + +class UpmixConverter : public AudioConverter { + public: + UpmixConverter(size_t src_channels, + size_t src_frames, + size_t 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 (size_t j = 0; j < dst_channels(); ++j) + dst[j][i] = value; + } + } +}; + +class DownmixConverter : public AudioConverter { + public: + DownmixConverter(size_t src_channels, + size_t src_frames, + size_t 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 (size_t j = 0; j < src_channels(); ++j) + sum += src[j][i]; + dst_mono[i] = sum / src_channels(); + } + } +}; + +class ResampleConverter : public AudioConverter { + public: + ResampleConverter(size_t src_channels, + size_t src_frames, + size_t dst_channels, + size_t dst_frames) + : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) { + resamplers_.reserve(src_channels); + for (size_t i = 0; i < src_channels; ++i) + resamplers_.push_back(std::unique_ptr<PushSincResampler>( + 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: + std::vector<std::unique_ptr<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: + explicit CompositionConverter( + std::vector<std::unique_ptr<AudioConverter>> converters) + : converters_(std::move(converters)) { + RTC_CHECK_GE(converters_.size(), 2); + // We need an intermediate buffer after every converter. + for (auto it = converters_.begin(); it != converters_.end() - 1; ++it) + buffers_.push_back( + std::unique_ptr<ChannelBuffer<float>>(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: + std::vector<std::unique_ptr<AudioConverter>> converters_; + std::vector<std::unique_ptr<ChannelBuffer<float>>> buffers_; +}; + +std::unique_ptr<AudioConverter> AudioConverter::Create(size_t src_channels, + size_t src_frames, + size_t dst_channels, + size_t dst_frames) { + std::unique_ptr<AudioConverter> sp; + if (src_channels > dst_channels) { + if (src_frames != dst_frames) { + std::vector<std::unique_ptr<AudioConverter>> converters; + converters.push_back(std::unique_ptr<AudioConverter>(new DownmixConverter( + src_channels, src_frames, dst_channels, src_frames))); + converters.push_back( + std::unique_ptr<AudioConverter>(new ResampleConverter( + dst_channels, src_frames, dst_channels, dst_frames))); + sp.reset(new CompositionConverter(std::move(converters))); + } else { + sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels, + dst_frames)); + } + } else if (src_channels < dst_channels) { + if (src_frames != dst_frames) { + std::vector<std::unique_ptr<AudioConverter>> converters; + converters.push_back( + std::unique_ptr<AudioConverter>(new ResampleConverter( + src_channels, src_frames, src_channels, dst_frames))); + converters.push_back(std::unique_ptr<AudioConverter>(new UpmixConverter( + src_channels, dst_frames, dst_channels, dst_frames))); + sp.reset(new CompositionConverter(std::move(converters))); + } 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; +} + +// For CompositionConverter. +AudioConverter::AudioConverter() + : src_channels_(0), src_frames_(0), dst_channels_(0), dst_frames_(0) {} + +AudioConverter::AudioConverter(size_t src_channels, + size_t src_frames, + size_t 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 |