Index: remoting/protocol/webrtc_audio_source_adapter_unittest.cc |
diff --git a/remoting/protocol/webrtc_audio_source_adapter_unittest.cc b/remoting/protocol/webrtc_audio_source_adapter_unittest.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..8b4407cefa646104a81bf3b3fda39de7e3e2ca70 |
--- /dev/null |
+++ b/remoting/protocol/webrtc_audio_source_adapter_unittest.cc |
@@ -0,0 +1,120 @@ |
+// 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/message_loop/message_loop.h" |
+#include "base/run_loop.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/mediastreaminterface.h" |
+#include "third_party/webrtc/base/refcount.h" |
+ |
+namespace remoting { |
+namespace protocol { |
+ |
+namespace { |
+ |
+const int kSampleRate = 48000; |
+const int kBytesPerSample = 2; |
+const int kChannels = 2; |
+constexpr base::TimeDelta kFrameDuration = |
+ base::TimeDelta::FromMilliseconds(10); |
+ |
+class FakeAudioSink : public webrtc::AudioTrackSinkInterface{ |
+ public: |
+ FakeAudioSink() {} |
+ ~FakeAudioSink() override {} |
+ |
+ 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 / base::TimeDelta::FromSeconds(1), |
+ 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>( |
+ message_loop_.task_runner()); |
+ audio_source_ = new FakeAudioSource(); |
+ audio_source_adapter_->Start(base::WrapUnique(audio_source_)); |
+ audio_source_adapter_->AddSink(&sink_); |
+ base::RunLoop().RunUntilIdle(); |
+ } |
+ |
+ void TearDown() override { |
+ audio_source_adapter_ = nullptr; |
+ base::RunLoop().RunUntilIdle(); |
+ } |
+ |
+ protected: |
+ base::MessageLoop message_loop_; |
+ 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 |