6776518bea
This refactoring takes a careful approach to avoid rushing the change: * stub headers are left in all the old locations of webrtc/base * existing GN targets are kept and now just forward to the moved ones using public_deps. The only exception to the above is the base_java target and its .java files, which were moved to webrtc/rtc_base right away since it's not possible to use public_deps for android_library. To avoid breaking builds, a temporary Dummy.java file was added to the new intermediate target in webrtc/rtc_base:base_java as well to avoid hitting a GN assert in the android_library template. The above approach should make the transition smooth without breaking downstream. A helper script was created (https://codereview.webrtc.org/2879203002/) and was run like this: stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634 stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634 Fixed invalid header guards in the following files: webrtc/base/base64.h webrtc/base/cryptstring.h webrtc/base/event.h webrtc/base/flags.h webrtc/base/httpbase.h webrtc/base/httpcommon-inl.h webrtc/base/httpcommon.h webrtc/base/httpserver.h webrtc/base/logsinks.h webrtc/base/macutils.h webrtc/base/nattypes.h webrtc/base/openssladapter.h webrtc/base/opensslstreamadapter.h webrtc/base/pathutils.h webrtc/base/physicalsocketserver.h webrtc/base/proxyinfo.h webrtc/base/sigslot.h webrtc/base/sigslotrepeater.h webrtc/base/socket.h webrtc/base/socketaddresspair.h webrtc/base/socketfactory.h webrtc/base/stringutils.h webrtc/base/testbase64.h webrtc/base/testutils.h webrtc/base/transformadapter.h webrtc/base/win32filesystem.h Added new header guards to: sslroots.h testbase64.h BUG=webrtc:7634 NOTRY=True NOPRESUBMIT=True R=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2877023002 . Cr-Commit-Position: refs/heads/master@{#18816}
62 lines
1.8 KiB
C++
62 lines
1.8 KiB
C++
/*
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* Copyright 2015 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_RTC_BASE_BUFFERQUEUE_H_
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#define WEBRTC_RTC_BASE_BUFFERQUEUE_H_
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#include <deque>
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#include <vector>
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#include "webrtc/base/buffer.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/criticalsection.h"
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namespace rtc {
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class BufferQueue {
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public:
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// Creates a buffer queue with a given capacity and default buffer size.
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BufferQueue(size_t capacity, size_t default_size);
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virtual ~BufferQueue();
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// Return number of queued buffers.
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size_t size() const;
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// Clear the BufferQueue by moving all Buffers from |queue_| to |free_list_|.
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void Clear();
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// ReadFront will only read one buffer at a time and will truncate buffers
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// that don't fit in the passed memory.
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// Returns true unless no data could be returned.
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bool ReadFront(void* data, size_t bytes, size_t* bytes_read);
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// WriteBack always writes either the complete memory or nothing.
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// Returns true unless no data could be written.
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bool WriteBack(const void* data, size_t bytes, size_t* bytes_written);
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protected:
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// These methods are called when the state of the queue changes.
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virtual void NotifyReadableForTest() {}
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virtual void NotifyWritableForTest() {}
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private:
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size_t capacity_;
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size_t default_size_;
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CriticalSection crit_;
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std::deque<Buffer*> queue_ GUARDED_BY(crit_);
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std::vector<Buffer*> free_list_ GUARDED_BY(crit_);
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RTC_DISALLOW_COPY_AND_ASSIGN(BufferQueue);
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};
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} // namespace rtc
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#endif // WEBRTC_RTC_BASE_BUFFERQUEUE_H_
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