blob: f03aaceabc653a739f736629e8aee1473b7a43db [file] [log] [blame]
// Copyright 2014 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "media/cast/net/rtp/rtp_packetizer.h"
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include "base/test/simple_test_tick_clock.h"
#include "media/base/fake_single_thread_task_runner.h"
#include "media/cast/common/encoded_frame.h"
#include "media/cast/net/pacing/paced_sender.h"
#include "media/cast/net/rtp/packet_storage.h"
#include "media/cast/net/rtp/rtp_parser.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "third_party/openscreen/src/cast/streaming/encoded_frame.h"
namespace media {
namespace cast {
namespace {
constexpr int kPayload = 127;
constexpr uint32_t kTimestampMs = 10;
constexpr uint16_t kSeqNum = 33;
constexpr int kSsrc = 0x12345;
constexpr unsigned int kFrameSize = 5000;
// The maximum packet length is the internet MTU (1500) minus the IP and ICMP
// header size.
constexpr int kMaxPacketLength = 1472;
// The maximum payload size is the maximum packet size (set using the internet
// standard MTU of 1500) minus the size of all layers' headers combined.
constexpr int kMaxPayloadSize = 1449;
// Allows for enough time to execute packets for a video frame of size
// |kFrameSize|.
static int kTaskExecutionMilliseconds = 34;
} // namespace
class TestRtpPacketTransport : public PacketTransport {
public:
explicit TestRtpPacketTransport(RtpPacketizerConfig config)
: config_(config),
sequence_number_(kSeqNum),
packets_sent_(0),
expected_number_of_packets_(0),
expected_packet_id_(0),
expected_frame_id_(FrameId::first() + 1) {}
TestRtpPacketTransport(const TestRtpPacketTransport&) = delete;
TestRtpPacketTransport& operator=(const TestRtpPacketTransport&) = delete;
void VerifyRtpHeader(const RtpCastHeader& rtp_header) {
VerifyCommonRtpHeader(rtp_header);
VerifyCastRtpHeader(rtp_header);
}
void VerifyCommonRtpHeader(const RtpCastHeader& rtp_header) {
EXPECT_EQ(kPayload, rtp_header.payload_type);
EXPECT_EQ(sequence_number_, rtp_header.sequence_number);
EXPECT_EQ(expected_rtp_timestamp_, rtp_header.rtp_timestamp);
EXPECT_EQ(config_.ssrc, rtp_header.sender_ssrc);
EXPECT_EQ(0, rtp_header.num_csrcs);
}
void VerifyCastRtpHeader(const RtpCastHeader& rtp_header) {
EXPECT_FALSE(rtp_header.is_key_frame);
EXPECT_EQ(expected_frame_id_, rtp_header.frame_id);
EXPECT_EQ(expected_packet_id_, rtp_header.packet_id);
EXPECT_EQ(expected_number_of_packets_ - 1, rtp_header.max_packet_id);
EXPECT_TRUE(rtp_header.is_reference);
EXPECT_EQ(expected_frame_id_ - 1, rtp_header.reference_frame_id);
if (rtp_header.packet_id != 0) {
EXPECT_EQ(rtp_header.num_extensions, 0)
<< "Extensions only allowed on first packet of a frame";
}
}
bool SendPacket(PacketRef packet, base::OnceClosure cb) final {
++packets_sent_;
RtpParser parser(kSsrc, kPayload);
RtpCastHeader rtp_header;
const uint8_t* payload_data;
size_t payload_size;
parser.ParsePacket(&packet->data[0], packet->data.size(), &rtp_header,
&payload_data, &payload_size);
VerifyRtpHeader(rtp_header);
++sequence_number_;
++expected_packet_id_;
return true;
}
int64_t GetBytesSent() final { return 0; }
void StartReceiving(PacketReceiverCallbackWithStatus packet_receiver) final {}
void StopReceiving() final {}
size_t number_of_packets_received() const { return packets_sent_; }
void set_expected_number_of_packets(size_t expected_number_of_packets) {
expected_number_of_packets_ = expected_number_of_packets;
}
void set_rtp_timestamp(RtpTimeTicks rtp_timestamp) {
expected_rtp_timestamp_ = rtp_timestamp;
}
RtpPacketizerConfig config_;
uint32_t sequence_number_;
size_t packets_sent_;
size_t number_of_packets_;
size_t expected_number_of_packets_;
// Assuming packets arrive in sequence.
int expected_packet_id_;
FrameId expected_frame_id_;
RtpTimeTicks expected_rtp_timestamp_;
};
class RtpPacketizerTest : public ::testing::Test {
public:
RtpPacketizerTest(const RtpPacketizerTest&) = delete;
RtpPacketizerTest& operator=(const RtpPacketizerTest&) = delete;
protected:
RtpPacketizerTest()
: task_runner_(new FakeSingleThreadTaskRunner(&testing_clock_)) {
config_.sequence_number = kSeqNum;
config_.ssrc = kSsrc;
config_.payload_type = kPayload;
config_.max_payload_length = kMaxPacketLength;
transport_ = std::make_unique<TestRtpPacketTransport>(config_);
pacer_ = std::make_unique<PacedSender>(kTargetBurstSize, kMaxBurstSize,
&testing_clock_, nullptr,
transport_.get(), task_runner_);
pacer_->RegisterSsrc(config_.ssrc, false);
rtp_packetizer_ = std::make_unique<RtpPacketizer>(
pacer_.get(), &packet_storage_, config_);
video_frame_.dependency =
openscreen::cast::EncodedFrame::Dependency::kDependent;
video_frame_.frame_id = FrameId::first() + 1;
video_frame_.referenced_frame_id = video_frame_.frame_id - 1;
video_frame_.data.assign(kFrameSize, 123);
video_frame_.rtp_timestamp = RtpTimeTicks().Expand(UINT32_C(0x0055aa11));
}
void RunTasks() {
for (int i = 0; i < kTaskExecutionMilliseconds; ++i) {
// Call process the timers every 1 ms.
testing_clock_.Advance(base::Milliseconds(1));
task_runner_->RunTasks();
}
}
// Sends |video_frame_| to the |rtp_packetizer_|.
void SendFrame() {
transport_->set_rtp_timestamp(video_frame_.rtp_timestamp);
testing_clock_.Advance(base::Milliseconds(kTimestampMs));
video_frame_.reference_time = testing_clock_.NowTicks();
rtp_packetizer_->SendFrameAsPackets(video_frame_);
RunTasks();
}
base::SimpleTestTickClock testing_clock_;
scoped_refptr<FakeSingleThreadTaskRunner> task_runner_;
EncodedFrame video_frame_;
PacketStorage packet_storage_;
RtpPacketizerConfig config_;
std::unique_ptr<TestRtpPacketTransport> transport_;
std::unique_ptr<PacedSender> pacer_;
std::unique_ptr<RtpPacketizer> rtp_packetizer_;
};
TEST_F(RtpPacketizerTest, SendStandardPackets) {
size_t expected_num_of_packets = kFrameSize / kMaxPacketLength + 1;
transport_->set_expected_number_of_packets(expected_num_of_packets);
SendFrame();
EXPECT_EQ(expected_num_of_packets, transport_->number_of_packets_received());
}
TEST_F(RtpPacketizerTest, SendEmptyPacket) {
video_frame_.data.clear();
transport_->set_expected_number_of_packets(0);
SendFrame();
EXPECT_EQ(0u, transport_->number_of_packets_received());
}
TEST_F(RtpPacketizerTest, SendPacketOfMaximumSize) {
video_frame_.data.assign(kMaxPayloadSize, 88);
transport_->set_expected_number_of_packets(1);
SendFrame();
EXPECT_EQ(1u, transport_->number_of_packets_received());
}
TEST_F(RtpPacketizerTest, SendPacketJustAboveMaximumSize) {
video_frame_.data.assign(kMaxPayloadSize + 1, 88);
transport_->set_expected_number_of_packets(2);
SendFrame();
EXPECT_EQ(2u, transport_->number_of_packets_received());
}
TEST_F(RtpPacketizerTest, SendPacketExactMultipleOfMaximumSize) {
video_frame_.data.assign(kMaxPayloadSize * 3, 88);
transport_->set_expected_number_of_packets(3);
SendFrame();
EXPECT_EQ(3u, transport_->number_of_packets_received());
}
TEST_F(RtpPacketizerTest, SendPacketsWithAdaptivePlayoutExtension) {
size_t expected_num_of_packets = kFrameSize / kMaxPacketLength + 1;
transport_->set_expected_number_of_packets(expected_num_of_packets);
video_frame_.new_playout_delay_ms = 500;
SendFrame();
EXPECT_EQ(expected_num_of_packets, transport_->number_of_packets_received());
}
TEST_F(RtpPacketizerTest, Stats) {
EXPECT_FALSE(rtp_packetizer_->send_packet_count());
EXPECT_FALSE(rtp_packetizer_->send_octet_count());
constexpr size_t kExpectedNumOfPackets = kFrameSize / kMaxPacketLength + 1;
transport_->set_expected_number_of_packets(kExpectedNumOfPackets);
SendFrame();
EXPECT_EQ(kExpectedNumOfPackets, rtp_packetizer_->send_packet_count());
EXPECT_EQ(kFrameSize, rtp_packetizer_->send_octet_count());
EXPECT_EQ(kExpectedNumOfPackets, transport_->number_of_packets_received());
}
} // namespace cast
} // namespace media