1 files changed, 89 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/test_utils.cc b/third_party/libwebrtc/modules/audio_processing/test/test_utils.cc
new file mode 100644
index 0000000000..9aeebe5155
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/test_utils.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2015 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 "modules/audio_processing/test/test_utils.h"
+
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+
+ChannelBufferWavReader::ChannelBufferWavReader(std::unique_ptr<WavReader> file)
+    : file_(std::move(file)) {}
+
+ChannelBufferWavReader::~ChannelBufferWavReader() = default;
+
+bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
+  RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
+  interleaved_.resize(buffer->size());
+  if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
+      interleaved_.size()) {
+    return false;
+  }
+
+  FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+  Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
+               buffer->channels());
+  return true;
+}
+
+ChannelBufferWavWriter::ChannelBufferWavWriter(std::unique_ptr<WavWriter> file)
+    : file_(std::move(file)) {}
+
+ChannelBufferWavWriter::~ChannelBufferWavWriter() = default;
+
+void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
+  RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
+  interleaved_.resize(buffer.size());
+  Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
+             &interleaved_[0]);
+  FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+  file_->WriteSamples(&interleaved_[0], interleaved_.size());
+}
+
+ChannelBufferVectorWriter::ChannelBufferVectorWriter(std::vector<float>* output)
+    : output_(output) {
+  RTC_DCHECK(output_);
+}
+
+ChannelBufferVectorWriter::~ChannelBufferVectorWriter() = default;
+
+void ChannelBufferVectorWriter::Write(const ChannelBuffer<float>& buffer) {
+  // Account for sample rate changes throughout a simulation.
+  interleaved_buffer_.resize(buffer.size());
+  Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
+             interleaved_buffer_.data());
+  size_t old_size = output_->size();
+  output_->resize(old_size + interleaved_buffer_.size());
+  FloatToFloatS16(interleaved_buffer_.data(), interleaved_buffer_.size(),
+                  output_->data() + old_size);
+}
+
+FILE* OpenFile(absl::string_view filename, absl::string_view mode) {
+  std::string filename_str(filename);
+  FILE* file = fopen(filename_str.c_str(), std::string(mode).c_str());
+  if (!file) {
+    printf("Unable to open file %s\n", filename_str.c_str());
+    exit(1);
+  }
+  return file;
+}
+
+void SetFrameSampleRate(Int16FrameData* frame, int sample_rate_hz) {
+  frame->sample_rate_hz = sample_rate_hz;
+  frame->samples_per_channel =
+      AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000;
+}
+
+}  // namespace webrtc