Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
This commit is contained in:
Arun Raghavan
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webrtc/common_audio/audio_converter.cc
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webrtc/common_audio/audio_converter.cc
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/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/common_audio/audio_converter.h"
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#include <cstring>
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#include "webrtc/base/checks.h"
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#include "webrtc/base/safe_conversions.h"
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#include "webrtc/common_audio/channel_buffer.h"
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#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
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#include "webrtc/system_wrappers/interface/scoped_vector.h"
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using rtc::checked_cast;
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namespace webrtc {
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class CopyConverter : public AudioConverter {
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public:
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CopyConverter(int src_channels, size_t src_frames, int dst_channels,
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size_t dst_frames)
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: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
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~CopyConverter() override {};
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void Convert(const float* const* src, size_t src_size, float* const* dst,
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size_t dst_capacity) override {
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CheckSizes(src_size, dst_capacity);
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if (src != dst) {
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for (int i = 0; i < src_channels(); ++i)
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std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
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}
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}
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};
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class UpmixConverter : public AudioConverter {
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public:
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UpmixConverter(int src_channels, size_t src_frames, int dst_channels,
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size_t dst_frames)
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: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
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~UpmixConverter() override {};
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void Convert(const float* const* src, size_t src_size, float* const* dst,
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size_t dst_capacity) override {
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CheckSizes(src_size, dst_capacity);
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for (size_t i = 0; i < dst_frames(); ++i) {
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const float value = src[0][i];
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for (int j = 0; j < dst_channels(); ++j)
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dst[j][i] = value;
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}
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}
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};
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class DownmixConverter : public AudioConverter {
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public:
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DownmixConverter(int src_channels, size_t src_frames, int dst_channels,
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size_t dst_frames)
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: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
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}
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~DownmixConverter() override {};
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void Convert(const float* const* src, size_t src_size, float* const* dst,
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size_t dst_capacity) override {
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CheckSizes(src_size, dst_capacity);
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float* dst_mono = dst[0];
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for (size_t i = 0; i < src_frames(); ++i) {
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float sum = 0;
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for (int j = 0; j < src_channels(); ++j)
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sum += src[j][i];
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dst_mono[i] = sum / src_channels();
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}
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}
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};
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class ResampleConverter : public AudioConverter {
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public:
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ResampleConverter(int src_channels, size_t src_frames, int dst_channels,
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size_t dst_frames)
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: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
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resamplers_.reserve(src_channels);
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for (int i = 0; i < src_channels; ++i)
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resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
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}
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~ResampleConverter() override {};
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void Convert(const float* const* src, size_t src_size, float* const* dst,
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size_t dst_capacity) override {
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CheckSizes(src_size, dst_capacity);
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for (size_t i = 0; i < resamplers_.size(); ++i)
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resamplers_[i]->Resample(src[i], src_frames(), dst[i], dst_frames());
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}
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private:
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ScopedVector<PushSincResampler> resamplers_;
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};
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// Apply a vector of converters in serial, in the order given. At least two
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// converters must be provided.
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class CompositionConverter : public AudioConverter {
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public:
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CompositionConverter(ScopedVector<AudioConverter> converters)
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: converters_(converters.Pass()) {
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RTC_CHECK_GE(converters_.size(), 2u);
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// We need an intermediate buffer after every converter.
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for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
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buffers_.push_back(new ChannelBuffer<float>((*it)->dst_frames(),
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(*it)->dst_channels()));
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}
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~CompositionConverter() override {};
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void Convert(const float* const* src, size_t src_size, float* const* dst,
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size_t dst_capacity) override {
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converters_.front()->Convert(src, src_size, buffers_.front()->channels(),
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buffers_.front()->size());
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for (size_t i = 2; i < converters_.size(); ++i) {
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auto src_buffer = buffers_[i - 2];
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auto dst_buffer = buffers_[i - 1];
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converters_[i]->Convert(src_buffer->channels(),
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src_buffer->size(),
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dst_buffer->channels(),
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dst_buffer->size());
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}
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converters_.back()->Convert(buffers_.back()->channels(),
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buffers_.back()->size(), dst, dst_capacity);
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}
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private:
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ScopedVector<AudioConverter> converters_;
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ScopedVector<ChannelBuffer<float>> buffers_;
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};
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rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
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size_t src_frames,
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int dst_channels,
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size_t dst_frames) {
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rtc::scoped_ptr<AudioConverter> sp;
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if (src_channels > dst_channels) {
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if (src_frames != dst_frames) {
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ScopedVector<AudioConverter> converters;
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converters.push_back(new DownmixConverter(src_channels, src_frames,
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dst_channels, src_frames));
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converters.push_back(new ResampleConverter(dst_channels, src_frames,
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dst_channels, dst_frames));
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sp.reset(new CompositionConverter(converters.Pass()));
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} else {
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sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
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dst_frames));
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}
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} else if (src_channels < dst_channels) {
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if (src_frames != dst_frames) {
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ScopedVector<AudioConverter> converters;
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converters.push_back(new ResampleConverter(src_channels, src_frames,
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src_channels, dst_frames));
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converters.push_back(new UpmixConverter(src_channels, dst_frames,
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dst_channels, dst_frames));
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sp.reset(new CompositionConverter(converters.Pass()));
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} else {
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sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
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dst_frames));
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}
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} else if (src_frames != dst_frames) {
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sp.reset(new ResampleConverter(src_channels, src_frames, dst_channels,
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dst_frames));
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} else {
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sp.reset(new CopyConverter(src_channels, src_frames, dst_channels,
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dst_frames));
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}
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return sp.Pass();
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}
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// For CompositionConverter.
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AudioConverter::AudioConverter()
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: src_channels_(0),
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src_frames_(0),
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dst_channels_(0),
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dst_frames_(0) {}
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AudioConverter::AudioConverter(int src_channels, size_t src_frames,
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int dst_channels, size_t dst_frames)
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: src_channels_(src_channels),
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src_frames_(src_frames),
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dst_channels_(dst_channels),
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dst_frames_(dst_frames) {
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RTC_CHECK(dst_channels == src_channels || dst_channels == 1 ||
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src_channels == 1);
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}
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void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
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RTC_CHECK_EQ(src_size, src_channels() * src_frames());
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RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames());
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}
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} // namespace webrtc
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