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- // Copyright 2016 The Chromium Authors. All rights reserved.
- // 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 {
- namespace 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> 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 protocol
- } // namespace remoting
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