webrtc_audio_source_adapter_unittest.cc 4.1 KB

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  1. // Copyright 2016 The Chromium Authors. All rights reserved.
  2. // Use of this source code is governed by a BSD-style license that can be
  3. // found in the LICENSE file.
  4. #include "remoting/protocol/webrtc_audio_source_adapter.h"
  5. #include <numeric>
  6. #include <vector>
  7. #include "base/memory/ptr_util.h"
  8. #include "base/memory/raw_ptr.h"
  9. #include "base/run_loop.h"
  10. #include "base/test/task_environment.h"
  11. #include "remoting/proto/audio.pb.h"
  12. #include "remoting/protocol/fake_audio_source.h"
  13. #include "testing/gmock/include/gmock/gmock.h"
  14. #include "testing/gtest/include/gtest/gtest.h"
  15. #include "third_party/webrtc/api/media_stream_interface.h"
  16. #include "third_party/webrtc/rtc_base/ref_count.h"
  17. #include "third_party/webrtc/rtc_base/ref_counted_object.h"
  18. namespace remoting {
  19. namespace protocol {
  20. namespace {
  21. const int kSampleRate = 48000;
  22. const int kBytesPerSample = 2;
  23. const int kChannels = 2;
  24. constexpr auto kFrameDuration = base::Milliseconds(10);
  25. class FakeAudioSink : public webrtc::AudioTrackSinkInterface{
  26. public:
  27. FakeAudioSink() = default;
  28. ~FakeAudioSink() override = default;
  29. void OnData(const void* audio_data,
  30. int bits_per_sample,
  31. int sample_rate,
  32. size_t number_of_channels,
  33. size_t number_of_samples) override {
  34. EXPECT_EQ(kSampleRate, sample_rate);
  35. EXPECT_EQ(kBytesPerSample * 8, bits_per_sample);
  36. EXPECT_EQ(kChannels, static_cast<int>(number_of_channels));
  37. EXPECT_EQ((kSampleRate * kFrameDuration).InSeconds(),
  38. static_cast<int>(number_of_samples));
  39. const int16_t* samples = reinterpret_cast<const int16_t*>(audio_data);
  40. samples_.insert(samples_.end(), samples,
  41. samples + number_of_samples * kChannels);
  42. }
  43. const std::vector<int16_t>& samples() { return samples_; }
  44. private:
  45. std::vector<int16_t> samples_;
  46. };
  47. } // namespace
  48. class WebrtcAudioSourceAdapterTest : public testing::Test {
  49. public:
  50. void SetUp() override {
  51. audio_source_adapter_ = new rtc::RefCountedObject<WebrtcAudioSourceAdapter>(
  52. task_environment_.GetMainThreadTaskRunner());
  53. audio_source_ = new FakeAudioSource();
  54. audio_source_adapter_->Start(base::WrapUnique(audio_source_.get()));
  55. audio_source_adapter_->AddSink(&sink_);
  56. base::RunLoop().RunUntilIdle();
  57. }
  58. void TearDown() override {
  59. audio_source_adapter_ = nullptr;
  60. base::RunLoop().RunUntilIdle();
  61. }
  62. protected:
  63. base::test::SingleThreadTaskEnvironment task_environment_;
  64. raw_ptr<FakeAudioSource> audio_source_;
  65. scoped_refptr<WebrtcAudioSourceAdapter> audio_source_adapter_;
  66. FakeAudioSink sink_;
  67. };
  68. TEST_F(WebrtcAudioSourceAdapterTest, PartialFrames) {
  69. int16_t sample_value = 1;
  70. std::vector<int> frame_sizes_ms = {10, 12, 18, 2, 5, 7, 55, 13, 8};
  71. for (int frame_size_ms : frame_sizes_ms) {
  72. int num_samples = frame_size_ms * kSampleRate / 1000;
  73. std::vector<int16_t> data(num_samples * kChannels);
  74. for (int i = 0; i < num_samples; ++i) {
  75. data[i * kChannels] = sample_value;
  76. data[i * kChannels + 1] = -sample_value;
  77. ++sample_value;
  78. }
  79. std::unique_ptr<AudioPacket> packet(new AudioPacket());
  80. packet->add_data(reinterpret_cast<char*>(&(data[0])),
  81. num_samples * kChannels * sizeof(int16_t));
  82. packet->set_encoding(AudioPacket::ENCODING_RAW);
  83. packet->set_sampling_rate(AudioPacket::SAMPLING_RATE_48000);
  84. packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2);
  85. packet->set_channels(AudioPacket::CHANNELS_STEREO);
  86. audio_source_->callback().Run(std::move(packet));
  87. }
  88. int total_length_ms =
  89. std::accumulate(frame_sizes_ms.begin(), frame_sizes_ms.end(), 0,
  90. [](int sum, int x) { return sum + x; });
  91. const std::vector<int16_t>& received = sink_.samples();
  92. int total_samples = total_length_ms * kSampleRate / 1000;
  93. ASSERT_EQ(total_samples * kChannels, static_cast<int>(received.size()));
  94. sample_value = 1;
  95. for (int i = 0; i < total_samples; ++i) {
  96. ASSERT_EQ(sample_value, received[i * kChannels]) << i;
  97. ASSERT_EQ(-sample_value, received[i * kChannels + 1]);
  98. ++sample_value;
  99. }
  100. }
  101. } // namespace protocol
  102. } // namespace remoting