f048872e91
migrating talk/base to webrtc/base. BUG=N/A R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17479005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
80 lines
2.7 KiB
C++
80 lines
2.7 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_MULTIPART_H__
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#define WEBRTC_BASE_MULTIPART_H__
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#include <string>
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#include <vector>
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#include "webrtc/base/sigslot.h"
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#include "webrtc/base/stream.h"
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namespace rtc {
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///////////////////////////////////////////////////////////////////////////////
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// MultipartStream - Implements an RFC2046 multipart stream by concatenating
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// the supplied parts together, and adding the correct boundaries.
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///////////////////////////////////////////////////////////////////////////////
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class MultipartStream : public StreamInterface, public sigslot::has_slots<> {
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public:
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MultipartStream(const std::string& type, const std::string& boundary);
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virtual ~MultipartStream();
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void GetContentType(std::string* content_type);
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// Note: If content_disposition and/or content_type are the empty string,
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// they will be omitted.
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bool AddPart(StreamInterface* data_stream,
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const std::string& content_disposition,
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const std::string& content_type);
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bool AddPart(const std::string& data,
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const std::string& content_disposition,
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const std::string& content_type);
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void EndParts();
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// Calculates the size of a part before actually adding the part.
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size_t GetPartSize(const std::string& data,
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const std::string& content_disposition,
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const std::string& content_type) const;
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size_t GetEndPartSize() const;
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// StreamInterface
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virtual StreamState GetState() const;
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virtual StreamResult Read(void* buffer, size_t buffer_len,
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size_t* read, int* error);
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virtual StreamResult Write(const void* data, size_t data_len,
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size_t* written, int* error);
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virtual void Close();
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virtual bool SetPosition(size_t position);
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virtual bool GetPosition(size_t* position) const;
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virtual bool GetSize(size_t* size) const;
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virtual bool GetAvailable(size_t* size) const;
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private:
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typedef std::vector<StreamInterface*> PartList;
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// StreamInterface Slots
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void OnEvent(StreamInterface* stream, int events, int error);
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std::string type_, boundary_;
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PartList parts_;
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bool adding_;
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size_t current_; // The index into parts_ of the current read position.
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size_t position_; // The current read position in bytes.
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DISALLOW_COPY_AND_ASSIGN(MultipartStream);
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};
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} // namespace rtc
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#endif // WEBRTC_BASE_MULTIPART_H__
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