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