48ac226b9a
This is useful for debugging h264 input when we don't have an h264 decoder, as the resulting file should be possible to play back using mplayer. It is also often convenient to dump rtp packets in an interleaved format where the size of a packet is inserted before the actual payload. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42139004 Cr-Commit-Position: refs/heads/master@{#8558} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8558 4adac7df-926f-26a2-2b94-8c16560cd09d
46 lines
1.3 KiB
C++
46 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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struct RtpPacket {
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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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class RtpFileReader {
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public:
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enum FileFormat { kPcap, kRtpDump, kLengthPacketInterleaved };
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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(RtpPacket* 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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