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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc | 111 |
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 |