| // Copyright 2016 The Chromium Authors |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "remoting/protocol/webrtc_audio_source_adapter.h" |
| |
| #include <numeric> |
| #include <vector> |
| |
| #include "base/memory/ptr_util.h" |
| #include "base/memory/raw_ptr.h" |
| #include "base/run_loop.h" |
| #include "base/test/task_environment.h" |
| #include "remoting/proto/audio.pb.h" |
| #include "remoting/protocol/fake_audio_source.h" |
| #include "testing/gmock/include/gmock/gmock.h" |
| #include "testing/gtest/include/gtest/gtest.h" |
| #include "third_party/webrtc/api/media_stream_interface.h" |
| #include "third_party/webrtc/rtc_base/ref_count.h" |
| #include "third_party/webrtc/rtc_base/ref_counted_object.h" |
| |
| namespace remoting::protocol { |
| |
| namespace { |
| |
| const int kSampleRate = 48000; |
| const int kBytesPerSample = 2; |
| const int kChannels = 2; |
| constexpr auto kFrameDuration = base::Milliseconds(10); |
| |
| class FakeAudioSink : public webrtc::AudioTrackSinkInterface { |
| public: |
| FakeAudioSink() = default; |
| ~FakeAudioSink() override = default; |
| |
| void OnData(const void* audio_data, |
| int bits_per_sample, |
| int sample_rate, |
| size_t number_of_channels, |
| size_t number_of_samples) override { |
| EXPECT_EQ(kSampleRate, sample_rate); |
| EXPECT_EQ(kBytesPerSample * 8, bits_per_sample); |
| EXPECT_EQ(kChannels, static_cast<int>(number_of_channels)); |
| EXPECT_EQ((kSampleRate * kFrameDuration).InSeconds(), |
| static_cast<int>(number_of_samples)); |
| const int16_t* samples = reinterpret_cast<const int16_t*>(audio_data); |
| samples_.insert(samples_.end(), samples, |
| samples + number_of_samples * kChannels); |
| } |
| |
| const std::vector<int16_t>& samples() { return samples_; } |
| |
| private: |
| std::vector<int16_t> samples_; |
| }; |
| |
| } // namespace |
| |
| class WebrtcAudioSourceAdapterTest : public testing::Test { |
| public: |
| void SetUp() override { |
| audio_source_adapter_ = new rtc::RefCountedObject<WebrtcAudioSourceAdapter>( |
| task_environment_.GetMainThreadTaskRunner()); |
| audio_source_ = new FakeAudioSource(); |
| audio_source_adapter_->Start(base::WrapUnique(audio_source_.get())); |
| audio_source_adapter_->AddSink(&sink_); |
| base::RunLoop().RunUntilIdle(); |
| } |
| |
| void TearDown() override { |
| audio_source_adapter_ = nullptr; |
| base::RunLoop().RunUntilIdle(); |
| } |
| |
| protected: |
| base::test::SingleThreadTaskEnvironment task_environment_; |
| raw_ptr<FakeAudioSource, AcrossTasksDanglingUntriaged> audio_source_; |
| scoped_refptr<WebrtcAudioSourceAdapter> audio_source_adapter_; |
| FakeAudioSink sink_; |
| }; |
| |
| TEST_F(WebrtcAudioSourceAdapterTest, PartialFrames) { |
| int16_t sample_value = 1; |
| std::vector<int> frame_sizes_ms = {10, 12, 18, 2, 5, 7, 55, 13, 8}; |
| for (int frame_size_ms : frame_sizes_ms) { |
| int num_samples = frame_size_ms * kSampleRate / 1000; |
| std::vector<int16_t> data(num_samples * kChannels); |
| for (int i = 0; i < num_samples; ++i) { |
| data[i * kChannels] = sample_value; |
| data[i * kChannels + 1] = -sample_value; |
| ++sample_value; |
| } |
| |
| std::unique_ptr<AudioPacket> packet(new AudioPacket()); |
| packet->add_data(reinterpret_cast<char*>(&(data[0])), |
| num_samples * kChannels * sizeof(int16_t)); |
| packet->set_encoding(AudioPacket::ENCODING_RAW); |
| packet->set_sampling_rate(AudioPacket::SAMPLING_RATE_48000); |
| packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2); |
| packet->set_channels(AudioPacket::CHANNELS_STEREO); |
| audio_source_->callback().Run(std::move(packet)); |
| } |
| |
| int total_length_ms = |
| std::accumulate(frame_sizes_ms.begin(), frame_sizes_ms.end(), 0, |
| [](int sum, int x) { return sum + x; }); |
| const std::vector<int16_t>& received = sink_.samples(); |
| int total_samples = total_length_ms * kSampleRate / 1000; |
| ASSERT_EQ(total_samples * kChannels, static_cast<int>(received.size())); |
| sample_value = 1; |
| for (int i = 0; i < total_samples; ++i) { |
| ASSERT_EQ(sample_value, received[i * kChannels]) << i; |
| ASSERT_EQ(-sample_value, received[i * kChannels + 1]); |
| ++sample_value; |
| } |
| } |
| |
| } // namespace remoting::protocol |