zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Move talk/app/webrtc to webrtc/api

Browse Source 
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc

The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.

I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002

BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1610243002 .

Cr-Commit-Position: refs/heads/master@{#11545}
This commit is contained in:
Henrik Kjellander
2016-02-10 10:53:12 +01:00
parent f5368ab81d
commit 15583c19d7
263 changed files with 1171 additions and 996 deletions
Download Patch File Download Diff File Expand all files Collapse all files

178
webrtc/api/sctputils_unittest.cc Normal file
View File

@ -0,0 +1,178 @@
/*
* 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 "webrtc/api/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));
}