blob: a941c307bb329e04edd9c28ff91c3c51c1968d90 [file] [log] [blame]
// Copyright 2012 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/client/audio/audio_player.h"
#include <cstdint>
#include <memory>
#include "base/callback.h"
#include "base/compiler_specific.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace {
const int kAudioSamplesPerFrame = 25;
const int kAudioSampleBytes = 4;
const int kAudioFrameBytes = kAudioSamplesPerFrame * kAudioSampleBytes;
const int kPaddingBytes = 16;
// TODO(nicholss): Update legacy audio player to use new audio buffer code.
// TODO(garykac): Generate random audio data in the tests rather than having
// a single constant value.
const uint8_t kDefaultBufferData = 0x5A;
const uint8_t kDummyAudioData = 0x8B;
} // namespace
namespace remoting {
class FakeAudioPlayer : public AudioPlayer {
public:
FakeAudioPlayer() = default;
bool ResetAudioPlayer(AudioPacket::SamplingRate) override { return true; }
uint32_t GetSamplesPerFrame() override { return kAudioSamplesPerFrame; }
};
class AudioPlayerTest : public ::testing::Test {
protected:
void SetUp() override {
audio_.reset(new FakeAudioPlayer());
buffer_.reset(new char[kAudioFrameBytes + kPaddingBytes]);
}
void TearDown() override {}
void ConsumeAudioFrame() {
uint8_t* buffer = reinterpret_cast<uint8_t*>(buffer_.get());
memset(buffer, kDefaultBufferData, kAudioFrameBytes + kPaddingBytes);
AudioPlayer::AudioPlayerCallback(reinterpret_cast<void*>(buffer_.get()),
kAudioFrameBytes,
reinterpret_cast<void*>(audio_.get()));
// Verify we haven't written beyond the end of the buffer.
for (int i = 0; i < kPaddingBytes; i++)
ASSERT_EQ(kDefaultBufferData, *(buffer + kAudioFrameBytes + i));
}
// Check that the first |num_bytes| bytes are filled with audio data and
// the rest of the buffer is zero-filled.
void CheckAudioFrameBytes(int num_bytes) {
uint8_t* buffer = reinterpret_cast<uint8_t*>(buffer_.get());
int i = 0;
for (; i < num_bytes; i++) {
ASSERT_EQ(kDummyAudioData, *(buffer + i));
}
// Rest of audio frame must be filled with '0's.
for (; i < kAudioFrameBytes; i++) {
ASSERT_EQ(0, *(buffer + i));
}
}
int GetNumQueuedSamples() {
return audio_->queued_bytes_ / kAudioSampleBytes;
}
int GetNumQueuedPackets() {
return static_cast<int>(audio_->queued_packets_.size());
}
int GetBytesConsumed() { return static_cast<int>(audio_->bytes_consumed_); }
std::unique_ptr<AudioPlayer> audio_;
std::unique_ptr<char[]> buffer_;
};
std::unique_ptr<AudioPacket> CreatePacketWithSamplingRate(
AudioPacket::SamplingRate rate,
int samples) {
std::unique_ptr<AudioPacket> packet(new AudioPacket());
packet->set_encoding(AudioPacket::ENCODING_RAW);
packet->set_sampling_rate(rate);
packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2);
packet->set_channels(AudioPacket::CHANNELS_STEREO);
// The data must be a multiple of 4 bytes (channels x bytes_per_sample).
std::string data;
data.resize(samples * kAudioSampleBytes, kDummyAudioData);
packet->add_data(data);
return packet;
}
std::unique_ptr<AudioPacket> CreatePacket44100Hz(int samples) {
return CreatePacketWithSamplingRate(AudioPacket::SAMPLING_RATE_44100,
samples);
}
std::unique_ptr<AudioPacket> CreatePacket48000Hz(int samples) {
return CreatePacketWithSamplingRate(AudioPacket::SAMPLING_RATE_48000,
samples);
}
TEST_F(AudioPlayerTest, Init) {
ASSERT_EQ(0, GetNumQueuedPackets());
audio_->ProcessAudioPacket(CreatePacket44100Hz(10), base::Closure());
ASSERT_EQ(1, GetNumQueuedPackets());
}
TEST_F(AudioPlayerTest, MultipleSamples) {
audio_->ProcessAudioPacket(CreatePacket44100Hz(10), base::Closure());
ASSERT_EQ(10, GetNumQueuedSamples());
ASSERT_EQ(1, GetNumQueuedPackets());
audio_->ProcessAudioPacket(CreatePacket44100Hz(20), base::Closure());
ASSERT_EQ(30, GetNumQueuedSamples());
ASSERT_EQ(2, GetNumQueuedPackets());
}
TEST_F(AudioPlayerTest, ChangeSampleRate) {
audio_->ProcessAudioPacket(CreatePacket44100Hz(10), base::Closure());
ASSERT_EQ(10, GetNumQueuedSamples());
ASSERT_EQ(1, GetNumQueuedPackets());
// New packet with different sampling rate causes previous samples to
// be removed.
audio_->ProcessAudioPacket(CreatePacket48000Hz(20), base::Closure());
ASSERT_EQ(20, GetNumQueuedSamples());
ASSERT_EQ(1, GetNumQueuedPackets());
}
TEST_F(AudioPlayerTest, ExceedLatency) {
// Push about 4 seconds worth of samples.
for (int i = 0; i < 100; ++i) {
audio_->ProcessAudioPacket(CreatePacket48000Hz(2000), base::Closure());
}
// Verify that we don't have more than 0.5s.
EXPECT_LT(GetNumQueuedSamples(), 24000);
}
// Incoming packets: 100
// Consume: 25 (w/ 75 remaining, offset 25 into packet)
TEST_F(AudioPlayerTest, ConsumePartialPacket) {
int total_samples = 0;
int bytes_consumed = 0;
// Process 100 samples.
int packet1_samples = 100;
total_samples += packet1_samples;
audio_->ProcessAudioPacket(CreatePacket44100Hz(packet1_samples),
base::Closure());
ASSERT_EQ(total_samples, GetNumQueuedSamples());
ASSERT_EQ(1, GetNumQueuedPackets());
ASSERT_EQ(bytes_consumed, GetBytesConsumed());
// Consume one frame (=25) of samples.
ConsumeAudioFrame();
total_samples -= kAudioSamplesPerFrame;
bytes_consumed += kAudioFrameBytes;
ASSERT_EQ(total_samples, GetNumQueuedSamples());
ASSERT_EQ(1, GetNumQueuedPackets());
ASSERT_EQ(bytes_consumed, GetBytesConsumed());
CheckAudioFrameBytes(kAudioFrameBytes);
// Remaining samples.
ASSERT_EQ(75, total_samples);
ASSERT_EQ(25 * kAudioSampleBytes, bytes_consumed);
}
// Incoming packets: 20, 70
// Consume: 25, 25 (w/ 40 remaining, offset 30 into packet)
TEST_F(AudioPlayerTest, ConsumeAcrossPackets) {
int total_samples = 0;
int bytes_consumed = 0;
// Packet 1.
int packet1_samples = 20;
total_samples += packet1_samples;
audio_->ProcessAudioPacket(CreatePacket44100Hz(packet1_samples),
base::Closure());
ASSERT_EQ(total_samples, GetNumQueuedSamples());
// Packet 2.
int packet2_samples = 70;
total_samples += packet2_samples;
audio_->ProcessAudioPacket(CreatePacket44100Hz(packet2_samples),
base::Closure());
ASSERT_EQ(total_samples, GetNumQueuedSamples());
ASSERT_EQ(bytes_consumed, GetBytesConsumed());
// Consume 1st frame of 25 samples.
// This will consume the entire 1st packet.
ConsumeAudioFrame();
total_samples -= kAudioSamplesPerFrame;
bytes_consumed += kAudioFrameBytes - (packet1_samples * kAudioSampleBytes);
ASSERT_EQ(total_samples, GetNumQueuedSamples());
ASSERT_EQ(1, GetNumQueuedPackets());
ASSERT_EQ(bytes_consumed, GetBytesConsumed());
CheckAudioFrameBytes(kAudioFrameBytes);
// Consume 2nd frame of 25 samples.
ConsumeAudioFrame();
total_samples -= kAudioSamplesPerFrame;
bytes_consumed += kAudioFrameBytes;
ASSERT_EQ(total_samples, GetNumQueuedSamples());
ASSERT_EQ(1, GetNumQueuedPackets());
ASSERT_EQ(bytes_consumed, GetBytesConsumed());
CheckAudioFrameBytes(kAudioFrameBytes);
// Remaining samples.
ASSERT_EQ(40, total_samples);
ASSERT_EQ(30 * kAudioSampleBytes, bytes_consumed);
}
// Incoming packets: 50, 30
// Consume: 25, 25, 25 (w/ 5 remaining, offset 25 into packet)
TEST_F(AudioPlayerTest, ConsumeEntirePacket) {
int total_samples = 0;
int bytes_consumed = 0;
// Packet 1.
int packet1_samples = 50;
total_samples += packet1_samples;
audio_->ProcessAudioPacket(CreatePacket44100Hz(packet1_samples),
base::Closure());
ASSERT_EQ(total_samples, GetNumQueuedSamples());
ASSERT_EQ(bytes_consumed, GetBytesConsumed());
// Packet 2.
int packet2_samples = 30;
total_samples += packet2_samples;
audio_->ProcessAudioPacket(CreatePacket44100Hz(packet2_samples),
base::Closure());
ASSERT_EQ(total_samples, GetNumQueuedSamples());
ASSERT_EQ(bytes_consumed, GetBytesConsumed());
// Consume 1st frame of 25 samples.
ConsumeAudioFrame();
total_samples -= kAudioSamplesPerFrame;
bytes_consumed += kAudioFrameBytes;
ASSERT_EQ(total_samples, GetNumQueuedSamples());
ASSERT_EQ(2, GetNumQueuedPackets());
ASSERT_EQ(bytes_consumed, GetBytesConsumed());
CheckAudioFrameBytes(kAudioFrameBytes);
// Consume 2nd frame of 25 samples.
// This will consume the entire first packet (exactly), but the entry for
// this packet will stick around (empty) until the next audio chunk is
// consumed.
ConsumeAudioFrame();
total_samples -= kAudioSamplesPerFrame;
bytes_consumed += kAudioFrameBytes;
ASSERT_EQ(total_samples, GetNumQueuedSamples());
ASSERT_EQ(2, GetNumQueuedPackets());
ASSERT_EQ(bytes_consumed, GetBytesConsumed());
CheckAudioFrameBytes(kAudioFrameBytes);
// Consume 3rd frame of 25 samples.
ConsumeAudioFrame();
total_samples -= kAudioSamplesPerFrame;
bytes_consumed += kAudioFrameBytes - (packet1_samples * kAudioSampleBytes);
ASSERT_EQ(total_samples, GetNumQueuedSamples());
ASSERT_EQ(1, GetNumQueuedPackets());
ASSERT_EQ(bytes_consumed, GetBytesConsumed());
CheckAudioFrameBytes(kAudioFrameBytes);
// Remaining samples.
ASSERT_EQ(5, total_samples);
ASSERT_EQ(25 * kAudioSampleBytes, bytes_consumed);
}
// Incoming packets: <none>
// Consume: 25
TEST_F(AudioPlayerTest, NoDataToConsume) {
// Attempt to consume a frame of 25 samples.
ConsumeAudioFrame();
ASSERT_EQ(0, GetNumQueuedSamples());
ASSERT_EQ(0, GetNumQueuedPackets());
ASSERT_EQ(0, GetBytesConsumed());
CheckAudioFrameBytes(0);
}
// Incoming packets: 10
// Consume: 25
TEST_F(AudioPlayerTest, NotEnoughDataToConsume) {
int total_samples = 0;
int bytes_consumed = 0;
// Packet 1.
int packet1_samples = 10;
total_samples += packet1_samples;
audio_->ProcessAudioPacket(CreatePacket44100Hz(packet1_samples),
base::Closure());
ASSERT_EQ(total_samples, GetNumQueuedSamples());
ASSERT_EQ(bytes_consumed, GetBytesConsumed());
// Attempt to consume a frame of 25 samples.
ConsumeAudioFrame();
ASSERT_EQ(0, GetNumQueuedSamples());
ASSERT_EQ(0, GetNumQueuedPackets());
ASSERT_EQ(0, GetBytesConsumed());
CheckAudioFrameBytes(packet1_samples * kAudioSampleBytes);
}
} // namespace remoting