Keep track of DTLS packet sizes to prevent partial reads.
The current use of rtc::FifoBuffer can lead to reading across DTLS packet
boundaries which could cause packets to not being processed correctly.
This CL introduces the new class rtc::BufferQueue and changes the
StreamInterfaceChannel to use it instead of the rtc::FifoBuffer.
BUG=chromium:447431
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/52509004
Cr-Commit-Position: refs/heads/master@{#9254}
diff --git a/webrtc/base/bufferqueue.h b/webrtc/base/bufferqueue.h
new file mode 100644
index 0000000..4adae41
--- /dev/null
+++ b/webrtc/base/bufferqueue.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_BUFFERQUEUE_H_
+#define WEBRTC_BASE_BUFFERQUEUE_H_
+
+#include <deque>
+#include <vector>
+
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/criticalsection.h"
+
+namespace rtc {
+
+class BufferQueue {
+ public:
+ // Creates a buffer queue queue with a given capacity and default buffer size.
+ BufferQueue(size_t capacity, size_t default_size);
+ ~BufferQueue();
+
+ // Return number of queued buffers.
+ size_t size() const;
+
+ // ReadFront will only read one buffer at a time and will truncate buffers
+ // that don't fit in the passed memory.
+ bool ReadFront(void* data, size_t bytes, size_t* bytes_read);
+
+ // WriteBack always writes either the complete memory or nothing.
+ bool WriteBack(const void* data, size_t bytes, size_t* bytes_written);
+
+ private:
+ size_t capacity_;
+ size_t default_size_;
+ std::deque<Buffer*> queue_;
+ std::vector<Buffer*> free_list_;
+ mutable CriticalSection crit_; // object lock
+
+ DISALLOW_COPY_AND_ASSIGN(BufferQueue);
+};
+
+} // namespace rtc
+
+#endif // WEBRTC_BASE_BUFFERQUEUE_H_