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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/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
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