webrtc_m130/talk/app/webrtc/sctputils_unittest.cc
deadbeef ab9b2d1516 Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
Reason for reland:
The original CL actually didn't break browser_tests; it was
just a coincidence that it started failing.

Original issue's description:
> Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ )
>
> Reason for revert:
> Broke browser_tests on Mac. Still need to investigate the cause.
>
> Original issue's description:
> > Moving MediaStreamSignaling logic into PeerConnection.
> >
> > This needs to happen because in the future, m-lines will be offered
> > based on the set of RtpSenders/RtpReceivers, rather than the set of
> > tracks that MediaStreamSignaling knows about.
> >
> > Besides that, MediaStreamSignaling was a "glue class" without
> > a clearly defined role, so it going away is good for other
> > reasons as well.
> >
> > Committed: https://crrev.com/97c392935411398b506861601c82e31d95c591f0
> > Cr-Commit-Position: refs/heads/master@{#10268}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/fc648b6d934e936f4d9a32c813364b331536ec3b
> Cr-Commit-Position: refs/heads/master@{#10269}

TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1404473005

Cr-Commit-Position: refs/heads/master@{#10277}
2015-10-14 18:33:20 +00:00

179 lines
6.3 KiB
C++

/*
* libjingle
* Copyright 2013 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/sctputils.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/gunit.h"
class SctpUtilsTest : public testing::Test {
public:
void VerifyOpenMessageFormat(const rtc::Buffer& 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::ByteBuffer buffer(packet.data(), packet.length());
ASSERT_TRUE(buffer.ReadUInt8(&message_type));
EXPECT_EQ(0x03, message_type);
ASSERT_TRUE(buffer.ReadUInt8(&channel_type));
if (config.ordered) {
EXPECT_EQ(config.maxRetransmits > -1 ?
0x01 : (config.maxRetransmitTime > -1 ? 0x02 : 0),
channel_type);
} else {
EXPECT_EQ(config.maxRetransmits > -1 ?
0x81 : (config.maxRetransmitTime > -1 ? 0x82 : 0x80),
channel_type);
}
ASSERT_TRUE(buffer.ReadUInt16(&priority));
ASSERT_TRUE(buffer.ReadUInt32(&reliability));
if (config.maxRetransmits > -1 || config.maxRetransmitTime > -1) {
EXPECT_EQ(config.maxRetransmits > -1 ?
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);
std::string label_output;
ASSERT_TRUE(buffer.ReadString(&label_output, label_length));
EXPECT_EQ(label, label_output);
std::string protocol_output;
ASSERT_TRUE(buffer.ReadString(&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::Buffer 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::Buffer 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(-1, output_config.maxRetransmits);
}
TEST_F(SctpUtilsTest, WriteParseOpenMessageWithMaxRetransmits) {
webrtc::DataChannelInit config;
std::string label = "abc";
config.maxRetransmits = 10;
config.protocol = "y";
rtc::Buffer 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_EQ(-1, output_config.maxRetransmitTime);
}
TEST_F(SctpUtilsTest, WriteParseAckMessage) {
rtc::Buffer packet;
webrtc::WriteDataChannelOpenAckMessage(&packet);
uint8_t message_type;
rtc::ByteBuffer buffer(packet.data(), packet.length());
ASSERT_TRUE(buffer.ReadUInt8(&message_type));
EXPECT_EQ(0x02, message_type);
EXPECT_TRUE(webrtc::ParseDataChannelOpenAckMessage(packet));
}
TEST_F(SctpUtilsTest, TestIsOpenMessage) {
rtc::ByteBuffer open;
open.WriteUInt8(0x03);
EXPECT_TRUE(webrtc::IsOpenMessage(open));
rtc::ByteBuffer openAck;
openAck.WriteUInt8(0x02);
EXPECT_FALSE(webrtc::IsOpenMessage(open));
rtc::ByteBuffer invalid;
openAck.WriteUInt8(0x01);
EXPECT_FALSE(webrtc::IsOpenMessage(invalid));
rtc::ByteBuffer empty;
EXPECT_FALSE(webrtc::IsOpenMessage(empty));
}