
Reason for revert: Still breaking CallAndModifyStream. Chromium CL intended to fix it (https://codereview.chromium.org/1435713002/) wasn't sufficient, because I forgot to call addStream/removeStream on the second connection. Original issue's description: > Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) > > Reason for revert: > Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream. > > Original issue's description: > > Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) > > > > Reason for revert: > > Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail. > > > > Original issue's description: > > > Adding the ability to create an RtpSender without a track. > > > > > > This CL also changes AddStream to immediately create a sender, rather > > > than waiting until the track is seen in SDP. And the PeerConnection now > > > builds the list of "send streams" from the list of senders, rather than > > > the collection of local media streams. > > > > > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5 > > > Cr-Commit-Position: refs/heads/master@{#10414} > > > > TBR=pthatcher@webrtc.org,pthatcher@chromium.org > > NOPRESUBMIT=true > > NOTREECHECKS=true > > NOTRY=true > > > > Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb > > Cr-Commit-Position: refs/heads/master@{#10417} > > TBR=pthatcher@webrtc.org,pthatcher@chromium.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/6834fa10f142bf5e2275142acb834898911d09ae > Cr-Commit-Position: refs/heads/master@{#10730} TBR=pthatcher@webrtc.org,pthatcher@chromium.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1460323002 Cr-Commit-Position: refs/heads/master@{#10732}
70 lines
1.9 KiB
C++
70 lines
1.9 KiB
C++
/*
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* Copyright 2004 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <string>
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#include "webrtc/base/gunit.h"
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#include "webrtc/base/helpers.h"
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#include "webrtc/base/ssladapter.h"
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namespace rtc {
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class RandomTest : public testing::Test {};
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TEST_F(RandomTest, TestCreateRandomId) {
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CreateRandomId();
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}
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TEST_F(RandomTest, TestCreateRandomDouble) {
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for (int i = 0; i < 100; ++i) {
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double r = CreateRandomDouble();
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EXPECT_GE(r, 0.0);
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EXPECT_LT(r, 1.0);
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}
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}
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TEST_F(RandomTest, TestCreateNonZeroRandomId) {
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EXPECT_NE(0U, CreateRandomNonZeroId());
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}
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TEST_F(RandomTest, TestCreateRandomString) {
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std::string random = CreateRandomString(256);
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EXPECT_EQ(256U, random.size());
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std::string random2;
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EXPECT_TRUE(CreateRandomString(256, &random2));
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EXPECT_NE(random, random2);
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EXPECT_EQ(256U, random2.size());
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}
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TEST_F(RandomTest, TestCreateRandomForTest) {
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// Make sure we get the output we expect.
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SetRandomTestMode(true);
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EXPECT_EQ(2154761789U, CreateRandomId());
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EXPECT_EQ("h0ISP4S5SJKH/9EY", CreateRandomString(16));
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// Reset and make sure we get the same output.
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SetRandomTestMode(true);
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EXPECT_EQ(2154761789U, CreateRandomId());
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EXPECT_EQ("h0ISP4S5SJKH/9EY", CreateRandomString(16));
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// Test different character sets.
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SetRandomTestMode(true);
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std::string str;
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EXPECT_TRUE(CreateRandomString(16, "a", &str));
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EXPECT_EQ("aaaaaaaaaaaaaaaa", str);
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EXPECT_TRUE(CreateRandomString(16, "abc", &str));
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EXPECT_EQ("acbccaaaabbaacbb", str);
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// Turn off test mode for other tests.
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SetRandomTestMode(false);
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}
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} // namespace rtc
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