blob: ebc94b4e43b6ff401f9ad59d946d94c337c3e408 [file] [log] [blame]
// Copyright 2015 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/audio_pump.h"
#include <stddef.h>
#include <memory>
#include <utility>
#include <vector>
#include "base/macros.h"
#include "base/memory/ptr_util.h"
#include "base/message_loop/message_loop.h"
#include "base/run_loop.h"
#include "remoting/codec/audio_encoder.h"
#include "remoting/proto/audio.pb.h"
#include "remoting/protocol/audio_source.h"
#include "remoting/protocol/audio_stub.h"
#include "remoting/protocol/fake_audio_source.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace remoting {
namespace protocol {
namespace {
// Creates a dummy packet with 1k data
std::unique_ptr<AudioPacket> MakeAudioPacket(int channel_count = 2) {
std::unique_ptr<AudioPacket> packet(new AudioPacket);
packet->add_data()->resize(1024);
packet->set_encoding(AudioPacket::ENCODING_RAW);
packet->set_sampling_rate(AudioPacket::SAMPLING_RATE_44100);
packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2);
packet->set_channels(static_cast<AudioPacket::Channels>(channel_count));
return packet;
}
} // namespace
class FakeAudioEncoder : public AudioEncoder {
public:
FakeAudioEncoder() = default;
~FakeAudioEncoder() override = default;
std::unique_ptr<AudioPacket> Encode(
std::unique_ptr<AudioPacket> packet) override {
EXPECT_TRUE(!!packet);
EXPECT_EQ(packet->encoding(), AudioPacket::ENCODING_RAW);
EXPECT_EQ(packet->sampling_rate(), AudioPacket::SAMPLING_RATE_44100);
EXPECT_EQ(packet->bytes_per_sample(), AudioPacket::BYTES_PER_SAMPLE_2);
EXPECT_LE(packet->channels(), AudioPacket::CHANNELS_STEREO);
return packet;
}
int GetBitrate() override { return 160000; }
private:
DISALLOW_COPY_AND_ASSIGN(FakeAudioEncoder);
};
class AudioPumpTest : public testing::Test, public protocol::AudioStub {
public:
AudioPumpTest() = default;
void SetUp() override;
void TearDown() override;
// protocol::AudioStub interface.
void ProcessAudioPacket(std::unique_ptr<AudioPacket> audio_packet,
const base::Closure& done) override;
protected:
base::MessageLoop message_loop_;
// |source_| and |encoder_| are owned by the |pump_|.
FakeAudioSource* source_;
FakeAudioEncoder* encoder_;
std::unique_ptr<AudioPump> pump_;
std::vector<std::unique_ptr<AudioPacket>> sent_packets_;
std::vector<base::Closure> done_closures_;
private:
DISALLOW_COPY_AND_ASSIGN(AudioPumpTest);
};
void AudioPumpTest::SetUp() {
source_ = new FakeAudioSource();
encoder_ = new FakeAudioEncoder();
pump_.reset(new AudioPump(message_loop_.task_runner(),
base::WrapUnique(source_),
base::WrapUnique(encoder_), this));
}
void AudioPumpTest::TearDown() {
pump_.reset();
// Let the message loop run to finish destroying the capturer.
base::RunLoop().RunUntilIdle();
}
void AudioPumpTest::ProcessAudioPacket(
std::unique_ptr<AudioPacket> audio_packet,
const base::Closure& done) {
sent_packets_.push_back(std::move(audio_packet));
done_closures_.push_back(done);
}
// Verify that the pump pauses pumping when the network is congested.
TEST_F(AudioPumpTest, BufferSizeLimit) {
// Run message loop to let the pump start the capturer.
base::RunLoop().RunUntilIdle();
ASSERT_FALSE(source_->callback().is_null());
// Try sending 100 packets, 1k each. The pump should stop pumping and start
// dropping the data at some point.
for (size_t i = 0; i < 100; ++i) {
source_->callback().Run(MakeAudioPacket());
base::RunLoop().RunUntilIdle();
}
size_t num_sent_packets = sent_packets_.size();
EXPECT_LT(num_sent_packets, 100U);
EXPECT_GT(num_sent_packets, 0U);
// Call done closure for the first packet. This should allow one more packet
// to be sent below.
done_closures_.front().Run();
base::RunLoop().RunUntilIdle();
// Verify that the pump continues to send captured audio.
source_->callback().Run(MakeAudioPacket());
base::RunLoop().RunUntilIdle();
EXPECT_EQ(num_sent_packets + 1, sent_packets_.size());
}
TEST_F(AudioPumpTest, DownmixAudioPacket) {
// Run message loop to let the pump start the capturer.
base::RunLoop().RunUntilIdle();
ASSERT_TRUE(source_->callback());
// Generate several audio packets with different channel counts.
static const int kChannels[] = {
AudioPacket::CHANNELS_7_1,
AudioPacket::CHANNELS_6_1,
AudioPacket::CHANNELS_5_1,
AudioPacket::CHANNELS_STEREO,
AudioPacket::CHANNELS_MONO,
AudioPacket::CHANNELS_7_1,
AudioPacket::CHANNELS_7_1,
AudioPacket::CHANNELS_7_1,
AudioPacket::CHANNELS_7_1,
AudioPacket::CHANNELS_6_1,
AudioPacket::CHANNELS_6_1,
AudioPacket::CHANNELS_6_1,
AudioPacket::CHANNELS_6_1,
AudioPacket::CHANNELS_5_1,
AudioPacket::CHANNELS_5_1,
AudioPacket::CHANNELS_5_1,
AudioPacket::CHANNELS_5_1,
AudioPacket::CHANNELS_STEREO,
AudioPacket::CHANNELS_STEREO,
AudioPacket::CHANNELS_STEREO,
AudioPacket::CHANNELS_STEREO,
AudioPacket::CHANNELS_MONO,
AudioPacket::CHANNELS_MONO,
AudioPacket::CHANNELS_MONO,
AudioPacket::CHANNELS_MONO,
};
for (size_t i = 0; i < arraysize(kChannels); i++) {
source_->callback().Run(MakeAudioPacket(kChannels[i]));
// Run message loop to let the pump processes the audio packet and send it
// to the encoder.
base::RunLoop().RunUntilIdle();
// Call done closure to allow one more packet to be sent.
ASSERT_EQ(done_closures_.size(), 1U);
done_closures_.front().Run();
done_closures_.pop_back();
base::RunLoop().RunUntilIdle();
}
ASSERT_EQ(sent_packets_.size(), arraysize(kChannels));
}
} // namespace protocol
} // namespace remoting