
Adding original_length to the Packet struct. This is populated with the plen value from the RTP dump file. In the case of reading a pcap file, original_length will be equal to length. Also increasing the maximum packet size to 3500 bytes. This is to accomodate some test files that contain PCM16b audio encoding. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7333 4adac7df-926f-26a2-2b94-8c16560cd09d
48 lines
1.3 KiB
C++
48 lines
1.3 KiB
C++
/*
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* Copyright (c) 2014 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_TEST_RTP_FILE_READER_H_
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#define WEBRTC_TEST_RTP_FILE_READER_H_
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#include <string>
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#include "webrtc/common_types.h"
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namespace webrtc {
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namespace test {
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class RtpFileReader {
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public:
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enum FileFormat {
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kPcap,
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kRtpDump,
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};
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struct Packet {
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// Accommodate for 50 ms packets of 32 kHz PCM16 samples (3200 bytes) plus
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// some overhead.
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static const size_t kMaxPacketBufferSize = 3500;
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uint8_t data[kMaxPacketBufferSize];
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size_t length;
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// The length the packet had on wire. Will be different from |length| when
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// reading a header-only RTP dump.
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size_t original_length;
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uint32_t time_ms;
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};
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virtual ~RtpFileReader() {}
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static RtpFileReader* Create(FileFormat format,
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const std::string& filename);
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virtual bool NextPacket(Packet* packet) = 0;
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_TEST_RTP_FILE_READER_H_
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