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}
179 lines
6.3 KiB
C++
179 lines
6.3 KiB
C++
/*
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* libjingle
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* Copyright 2013 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "talk/app/webrtc/sctputils.h"
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#include "webrtc/base/bytebuffer.h"
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#include "webrtc/base/gunit.h"
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class SctpUtilsTest : public testing::Test {
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public:
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void VerifyOpenMessageFormat(const rtc::Buffer& packet,
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const std::string& label,
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const webrtc::DataChannelInit& config) {
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uint8_t message_type;
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uint8_t channel_type;
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uint32_t reliability;
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uint16_t priority;
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uint16_t label_length;
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uint16_t protocol_length;
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rtc::ByteBuffer buffer(packet.data(), packet.length());
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ASSERT_TRUE(buffer.ReadUInt8(&message_type));
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EXPECT_EQ(0x03, message_type);
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ASSERT_TRUE(buffer.ReadUInt8(&channel_type));
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if (config.ordered) {
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EXPECT_EQ(config.maxRetransmits > -1 ?
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0x01 : (config.maxRetransmitTime > -1 ? 0x02 : 0),
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channel_type);
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} else {
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EXPECT_EQ(config.maxRetransmits > -1 ?
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0x81 : (config.maxRetransmitTime > -1 ? 0x82 : 0x80),
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channel_type);
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}
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ASSERT_TRUE(buffer.ReadUInt16(&priority));
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ASSERT_TRUE(buffer.ReadUInt32(&reliability));
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if (config.maxRetransmits > -1 || config.maxRetransmitTime > -1) {
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EXPECT_EQ(config.maxRetransmits > -1 ?
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config.maxRetransmits : config.maxRetransmitTime,
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static_cast<int>(reliability));
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}
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ASSERT_TRUE(buffer.ReadUInt16(&label_length));
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ASSERT_TRUE(buffer.ReadUInt16(&protocol_length));
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EXPECT_EQ(label.size(), label_length);
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EXPECT_EQ(config.protocol.size(), protocol_length);
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std::string label_output;
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ASSERT_TRUE(buffer.ReadString(&label_output, label_length));
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EXPECT_EQ(label, label_output);
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std::string protocol_output;
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ASSERT_TRUE(buffer.ReadString(&protocol_output, protocol_length));
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EXPECT_EQ(config.protocol, protocol_output);
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}
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};
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TEST_F(SctpUtilsTest, WriteParseOpenMessageWithOrderedReliable) {
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webrtc::DataChannelInit config;
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std::string label = "abc";
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config.protocol = "y";
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rtc::Buffer packet;
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ASSERT_TRUE(webrtc::WriteDataChannelOpenMessage(label, config, &packet));
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VerifyOpenMessageFormat(packet, label, config);
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std::string output_label;
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webrtc::DataChannelInit output_config;
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ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(
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packet, &output_label, &output_config));
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EXPECT_EQ(label, output_label);
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EXPECT_EQ(config.protocol, output_config.protocol);
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EXPECT_EQ(config.ordered, output_config.ordered);
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EXPECT_EQ(config.maxRetransmitTime, output_config.maxRetransmitTime);
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EXPECT_EQ(config.maxRetransmits, output_config.maxRetransmits);
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}
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TEST_F(SctpUtilsTest, WriteParseOpenMessageWithMaxRetransmitTime) {
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webrtc::DataChannelInit config;
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std::string label = "abc";
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config.ordered = false;
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config.maxRetransmitTime = 10;
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config.protocol = "y";
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rtc::Buffer packet;
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ASSERT_TRUE(webrtc::WriteDataChannelOpenMessage(label, config, &packet));
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VerifyOpenMessageFormat(packet, label, config);
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std::string output_label;
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webrtc::DataChannelInit output_config;
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ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(
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packet, &output_label, &output_config));
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EXPECT_EQ(label, output_label);
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EXPECT_EQ(config.protocol, output_config.protocol);
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EXPECT_EQ(config.ordered, output_config.ordered);
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EXPECT_EQ(config.maxRetransmitTime, output_config.maxRetransmitTime);
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EXPECT_EQ(-1, output_config.maxRetransmits);
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}
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TEST_F(SctpUtilsTest, WriteParseOpenMessageWithMaxRetransmits) {
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webrtc::DataChannelInit config;
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std::string label = "abc";
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config.maxRetransmits = 10;
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config.protocol = "y";
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rtc::Buffer packet;
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ASSERT_TRUE(webrtc::WriteDataChannelOpenMessage(label, config, &packet));
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VerifyOpenMessageFormat(packet, label, config);
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std::string output_label;
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webrtc::DataChannelInit output_config;
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ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(
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packet, &output_label, &output_config));
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EXPECT_EQ(label, output_label);
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EXPECT_EQ(config.protocol, output_config.protocol);
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EXPECT_EQ(config.ordered, output_config.ordered);
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EXPECT_EQ(config.maxRetransmits, output_config.maxRetransmits);
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EXPECT_EQ(-1, output_config.maxRetransmitTime);
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}
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TEST_F(SctpUtilsTest, WriteParseAckMessage) {
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rtc::Buffer packet;
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webrtc::WriteDataChannelOpenAckMessage(&packet);
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uint8_t message_type;
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rtc::ByteBuffer buffer(packet.data(), packet.length());
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ASSERT_TRUE(buffer.ReadUInt8(&message_type));
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EXPECT_EQ(0x02, message_type);
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EXPECT_TRUE(webrtc::ParseDataChannelOpenAckMessage(packet));
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}
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TEST_F(SctpUtilsTest, TestIsOpenMessage) {
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rtc::ByteBuffer open;
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open.WriteUInt8(0x03);
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EXPECT_TRUE(webrtc::IsOpenMessage(open));
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rtc::ByteBuffer openAck;
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openAck.WriteUInt8(0x02);
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EXPECT_FALSE(webrtc::IsOpenMessage(open));
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rtc::ByteBuffer invalid;
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openAck.WriteUInt8(0x01);
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EXPECT_FALSE(webrtc::IsOpenMessage(invalid));
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rtc::ByteBuffer empty;
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EXPECT_FALSE(webrtc::IsOpenMessage(empty));
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}
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