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Diffstat (limited to 'webrtc/common_audio/audio_converter.cc')
| -rw-r--r-- | webrtc/common_audio/audio_converter.cc | 200 |
1 files changed, 200 insertions, 0 deletions
diff --git a/webrtc/common_audio/audio_converter.cc b/webrtc/common_audio/audio_converter.cc new file mode 100644 index 0000000000..f1709ae653 --- /dev/null +++ b/webrtc/common_audio/audio_converter.cc @@ -0,0 +1,200 @@ +/* + * 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 |