1 files changed, 111 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
new file mode 100644
index 0000000000..6dbc8ee9fe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
@@ -0,0 +1,111 @@
+/*
+ *  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_coding/codecs/opus/test/audio_ring_buffer.h"
+
+#include <memory>
+
+#include "common_audio/channel_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class AudioRingBufferTest
+    : public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > {
+};
+
+void ReadAndWriteTest(const ChannelBuffer<float>& input,
+                      size_t num_write_chunk_frames,
+                      size_t num_read_chunk_frames,
+                      size_t buffer_frames,
+                      ChannelBuffer<float>* output) {
+  const size_t num_channels = input.num_channels();
+  const size_t total_frames = input.num_frames();
+  AudioRingBuffer buf(num_channels, buffer_frames);
+  std::unique_ptr<float*[]> slice(new float*[num_channels]);
+
+  size_t input_pos = 0;
+  size_t output_pos = 0;
+  while (input_pos + buf.WriteFramesAvailable() < total_frames) {
+    // Write until the buffer is as full as possible.
+    while (buf.WriteFramesAvailable() >= num_write_chunk_frames) {
+      buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+                num_write_chunk_frames);
+      input_pos += num_write_chunk_frames;
+    }
+    // Read until the buffer is as empty as possible.
+    while (buf.ReadFramesAvailable() >= num_read_chunk_frames) {
+      EXPECT_LT(output_pos, total_frames);
+      buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+               num_read_chunk_frames);
+      output_pos += num_read_chunk_frames;
+    }
+  }
+
+  // Write and read the last bit.
+  if (input_pos < total_frames) {
+    buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+              total_frames - input_pos);
+  }
+  if (buf.ReadFramesAvailable()) {
+    buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+             buf.ReadFramesAvailable());
+  }
+  EXPECT_EQ(0u, buf.ReadFramesAvailable());
+}
+
+TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
+  const size_t kFrames = 5000;
+  const size_t num_channels = ::testing::get<3>(GetParam());
+
+  // Initialize the input data to an increasing sequence.
+  ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels));
+  for (size_t i = 0; i < num_channels; ++i)
+    for (size_t j = 0; j < kFrames; ++j)
+      input.channels()[i][j] = (i + 1) * (j + 1);
+
+  ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels));
+  ReadAndWriteTest(input, ::testing::get<0>(GetParam()),
+                   ::testing::get<1>(GetParam()), ::testing::get<2>(GetParam()),
+                   &output);
+
+  // Verify the read data matches the input.
+  for (size_t i = 0; i < num_channels; ++i)
+    for (size_t j = 0; j < kFrames; ++j)
+      EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    AudioRingBufferTest,
+    AudioRingBufferTest,
+    ::testing::Combine(::testing::Values(10, 20, 42),  // num_write_chunk_frames
+                       ::testing::Values(1, 10, 17),   // num_read_chunk_frames
+                       ::testing::Values(100, 256),    // buffer_frames
+                       ::testing::Values(1, 4)));      // num_channels
+
+TEST_F(AudioRingBufferTest, MoveReadPosition) {
+  const size_t kNumChannels = 1;
+  const float kInputArray[] = {1, 2, 3, 4};
+  const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray);
+  ChannelBuffer<float> input(kNumFrames, kNumChannels);
+  input.SetDataForTesting(kInputArray, kNumFrames);
+  AudioRingBuffer buf(kNumChannels, kNumFrames);
+  buf.Write(input.channels(), kNumChannels, kNumFrames);
+
+  buf.MoveReadPositionForward(3);
+  ChannelBuffer<float> output(1, kNumChannels);
+  buf.Read(output.channels(), kNumChannels, 1);
+  EXPECT_EQ(4, output.channels()[0][0]);
+  buf.MoveReadPositionBackward(3);
+  buf.Read(output.channels(), kNumChannels, 1);
+  EXPECT_EQ(2, output.channels()[0][0]);
+}
+
+}  // namespace webrtc