webrtc/pc/sctp_utils_unittest.cc
Victor Boivie cd3d29b6fb pc: Simplify StreamId class
Before this CL, the StreamId class represented either a valid SCTP
stream ID, or "nothing", which means that it was a wrapped
absl::optional. Since created data channels don't have a SCTP stream ID
until it's known whether this peer will use odd or even numbers, the
"nothing" value was used for that state.

This unfortunately made it a bit hard to work with objects of this type,
as one always had to check if it contained a value. And even if a caller
would check this, and then pass the StreamId to a different function,
that function would have to do the check itself (often as a RTC_DCHECK)
since the passed StreamId always could have that state.

This CL simply extracts the "absl::optional" part of it, forcing holders
to wrap it in an optional type - when it can be "nothing". But allowing
the other code to just pass StreamId that can't be "nothing". That
simplifies the code a bit, potentially removing some bugs.

Bug: chromium:41221056
Change-Id: I93104cdd5d2f5fc1dbeb9d9dfc4cf361f11a9d68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342440
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41880}
2024-03-12 10:57:56 +00:00

211 lines
7 KiB
C++

/*
* Copyright 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/sctp_utils.h"
#include <stdint.h>
#include <limits>
#include "absl/types/optional.h"
#include "api/priority.h"
#include "media/sctp/sctp_transport_internal.h"
#include "rtc_base/byte_buffer.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "test/gtest.h"
using webrtc::StreamId;
class SctpUtilsTest : public ::testing::Test {
public:
void VerifyOpenMessageFormat(const rtc::CopyOnWriteBuffer& packet,
const std::string& label,
const webrtc::DataChannelInit& config) {
uint8_t message_type;
uint8_t channel_type;
uint32_t reliability;
uint16_t priority;
uint16_t label_length;
uint16_t protocol_length;
rtc::ByteBufferReader buffer(packet);
ASSERT_TRUE(buffer.ReadUInt8(&message_type));
EXPECT_EQ(0x03, message_type);
ASSERT_TRUE(buffer.ReadUInt8(&channel_type));
if (config.ordered) {
EXPECT_EQ(
config.maxRetransmits ? 0x01 : (config.maxRetransmitTime ? 0x02 : 0),
channel_type);
} else {
EXPECT_EQ(config.maxRetransmits
? 0x81
: (config.maxRetransmitTime ? 0x82 : 0x80),
channel_type);
}
ASSERT_TRUE(buffer.ReadUInt16(&priority));
if (config.priority) {
// Exact values are checked by round-trip conversion, but
// all values defined are greater than zero.
EXPECT_GT(priority, 0);
} else {
EXPECT_EQ(priority, 0);
}
ASSERT_TRUE(buffer.ReadUInt32(&reliability));
if (config.maxRetransmits || config.maxRetransmitTime) {
EXPECT_EQ(config.maxRetransmits ? *config.maxRetransmits
: *config.maxRetransmitTime,
static_cast<int>(reliability));
}
ASSERT_TRUE(buffer.ReadUInt16(&label_length));
ASSERT_TRUE(buffer.ReadUInt16(&protocol_length));
EXPECT_EQ(label.size(), label_length);
EXPECT_EQ(config.protocol.size(), protocol_length);
absl::string_view label_output;
ASSERT_TRUE(buffer.ReadStringView(&label_output, label_length));
EXPECT_EQ(label, label_output);
absl::string_view protocol_output;
ASSERT_TRUE(buffer.ReadStringView(&protocol_output, protocol_length));
EXPECT_EQ(config.protocol, protocol_output);
}
};
TEST_F(SctpUtilsTest, WriteParseOpenMessageWithOrderedReliable) {
webrtc::DataChannelInit config;
std::string label = "abc";
config.protocol = "y";
rtc::CopyOnWriteBuffer packet;
ASSERT_TRUE(webrtc::WriteDataChannelOpenMessage(label, config, &packet));
VerifyOpenMessageFormat(packet, label, config);
std::string output_label;
webrtc::DataChannelInit output_config;
ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(packet, &output_label,
&output_config));
EXPECT_EQ(label, output_label);
EXPECT_EQ(config.protocol, output_config.protocol);
EXPECT_EQ(config.ordered, output_config.ordered);
EXPECT_EQ(config.maxRetransmitTime, output_config.maxRetransmitTime);
EXPECT_EQ(config.maxRetransmits, output_config.maxRetransmits);
}
TEST_F(SctpUtilsTest, WriteParseOpenMessageWithMaxRetransmitTime) {
webrtc::DataChannelInit config;
std::string label = "abc";
config.ordered = false;
config.maxRetransmitTime = 10;
config.protocol = "y";
rtc::CopyOnWriteBuffer packet;
ASSERT_TRUE(webrtc::WriteDataChannelOpenMessage(label, config, &packet));
VerifyOpenMessageFormat(packet, label, config);
std::string output_label;
webrtc::DataChannelInit output_config;
ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(packet, &output_label,
&output_config));
EXPECT_EQ(label, output_label);
EXPECT_EQ(config.protocol, output_config.protocol);
EXPECT_EQ(config.ordered, output_config.ordered);
EXPECT_EQ(*config.maxRetransmitTime, *output_config.maxRetransmitTime);
EXPECT_FALSE(output_config.maxRetransmits);
}
TEST_F(SctpUtilsTest, WriteParseOpenMessageWithMaxRetransmits) {
webrtc::DataChannelInit config;
std::string label = "abc";
config.maxRetransmits = 10;
config.protocol = "y";
rtc::CopyOnWriteBuffer packet;
ASSERT_TRUE(webrtc::WriteDataChannelOpenMessage(label, config, &packet));
VerifyOpenMessageFormat(packet, label, config);
std::string output_label;
webrtc::DataChannelInit output_config;
ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(packet, &output_label,
&output_config));
EXPECT_EQ(label, output_label);
EXPECT_EQ(config.protocol, output_config.protocol);
EXPECT_EQ(config.ordered, output_config.ordered);
EXPECT_EQ(config.maxRetransmits, output_config.maxRetransmits);
EXPECT_FALSE(output_config.maxRetransmitTime);
}
TEST_F(SctpUtilsTest, WriteParseOpenMessageWithPriority) {
webrtc::DataChannelInit config;
std::string label = "abc";
config.protocol = "y";
config.priority = webrtc::Priority::kVeryLow;
rtc::CopyOnWriteBuffer packet;
ASSERT_TRUE(webrtc::WriteDataChannelOpenMessage(label, config, &packet));
VerifyOpenMessageFormat(packet, label, config);
std::string output_label;
webrtc::DataChannelInit output_config;
ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(packet, &output_label,
&output_config));
EXPECT_EQ(label, output_label);
ASSERT_TRUE(output_config.priority);
EXPECT_EQ(*config.priority, *output_config.priority);
}
TEST_F(SctpUtilsTest, WriteParseAckMessage) {
rtc::CopyOnWriteBuffer packet;
webrtc::WriteDataChannelOpenAckMessage(&packet);
uint8_t message_type;
rtc::ByteBufferReader buffer(packet);
ASSERT_TRUE(buffer.ReadUInt8(&message_type));
EXPECT_EQ(0x02, message_type);
EXPECT_TRUE(webrtc::ParseDataChannelOpenAckMessage(packet));
}
TEST_F(SctpUtilsTest, TestIsOpenMessage) {
rtc::CopyOnWriteBuffer open(1);
open.MutableData()[0] = 0x03;
EXPECT_TRUE(webrtc::IsOpenMessage(open));
rtc::CopyOnWriteBuffer openAck(1);
openAck.MutableData()[0] = 0x02;
EXPECT_FALSE(webrtc::IsOpenMessage(openAck));
rtc::CopyOnWriteBuffer invalid(1);
invalid.MutableData()[0] = 0x01;
EXPECT_FALSE(webrtc::IsOpenMessage(invalid));
rtc::CopyOnWriteBuffer empty;
EXPECT_FALSE(webrtc::IsOpenMessage(empty));
}
TEST(SctpSidTest, Basics) {
// These static asserts are mostly here to aid with readability (i.e. knowing
// what these constants represent).
static_assert(cricket::kMinSctpSid == 0, "Min stream id should be 0");
static_assert(cricket::kMaxSctpSid <= cricket::kSpecMaxSctpSid, "");
static_assert(
cricket::kSpecMaxSctpSid == std::numeric_limits<uint16_t>::max(),
"Max legal sctp stream value should be 0xffff");
}