5def7b9fde
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}
57 lines
1.8 KiB
C++
57 lines
1.8 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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#ifndef WEBRTC_BASE_HELPERS_H_
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#define WEBRTC_BASE_HELPERS_H_
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#include <string>
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#include "webrtc/base/basictypes.h"
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namespace rtc {
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// For testing, we can return predictable data.
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void SetRandomTestMode(bool test);
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// Initializes the RNG, and seeds it with the specified entropy.
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bool InitRandom(int seed);
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bool InitRandom(const char* seed, size_t len);
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// Generates a (cryptographically) random string of the given length.
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// We generate base64 values so that they will be printable.
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// WARNING: could silently fail. Use the version below instead.
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std::string CreateRandomString(size_t length);
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// Generates a (cryptographically) random string of the given length.
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// We generate base64 values so that they will be printable.
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// Return false if the random number generator failed.
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bool CreateRandomString(size_t length, std::string* str);
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// Generates a (cryptographically) random string of the given length,
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// with characters from the given table. Return false if the random
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// number generator failed.
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bool CreateRandomString(size_t length, const std::string& table,
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std::string* str);
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// Generates a random id.
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uint32_t CreateRandomId();
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// Generates a 64 bit random id.
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uint64_t CreateRandomId64();
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// Generates a random id > 0.
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uint32_t CreateRandomNonZeroId();
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// Generates a random double between 0.0 (inclusive) and 1.0 (exclusive).
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double CreateRandomDouble();
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} // namespace rtc
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#endif // WEBRTC_BASE_HELPERS_H_
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