Move webrtc/{base => rtc_base}
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}
diff --git a/webrtc/base/bufferqueue.h b/webrtc/base/bufferqueue.h
index bc9fc84..3142ae3 100644
--- a/webrtc/base/bufferqueue.h
+++ b/webrtc/base/bufferqueue.h
@@ -11,51 +11,9 @@
#ifndef WEBRTC_BASE_BUFFERQUEUE_H_
#define WEBRTC_BASE_BUFFERQUEUE_H_
-#include <deque>
-#include <vector>
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/criticalsection.h"
-
-namespace rtc {
-
-class BufferQueue {
- public:
- // Creates a buffer queue with a given capacity and default buffer size.
- BufferQueue(size_t capacity, size_t default_size);
- virtual ~BufferQueue();
-
- // Return number of queued buffers.
- size_t size() const;
-
- // Clear the BufferQueue by moving all Buffers from |queue_| to |free_list_|.
- void Clear();
-
- // ReadFront will only read one buffer at a time and will truncate buffers
- // that don't fit in the passed memory.
- // Returns true unless no data could be returned.
- bool ReadFront(void* data, size_t bytes, size_t* bytes_read);
-
- // WriteBack always writes either the complete memory or nothing.
- // Returns true unless no data could be written.
- bool WriteBack(const void* data, size_t bytes, size_t* bytes_written);
-
- protected:
- // These methods are called when the state of the queue changes.
- virtual void NotifyReadableForTest() {}
- virtual void NotifyWritableForTest() {}
-
- private:
- size_t capacity_;
- size_t default_size_;
- CriticalSection crit_;
- std::deque<Buffer*> queue_ GUARDED_BY(crit_);
- std::vector<Buffer*> free_list_ GUARDED_BY(crit_);
-
- RTC_DISALLOW_COPY_AND_ASSIGN(BufferQueue);
-};
-
-} // namespace rtc
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/bufferqueue.h"
#endif // WEBRTC_BASE_BUFFERQUEUE_H_