aosp12/external/webrtc/test/rtp_file_reader.h

50 lines
1.6 KiB
C++

/*
* Copyright (c) 2014 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 TEST_RTP_FILE_READER_H_
#define TEST_RTP_FILE_READER_H_
#include <set>
#include <string>
namespace webrtc {
namespace test {
struct RtpPacket {
// Accommodate for 50 ms packets of 32 kHz PCM16 samples (3200 bytes) plus
// some overhead.
static const size_t kMaxPacketBufferSize = 3500;
uint8_t data[kMaxPacketBufferSize];
size_t length;
// The length the packet had on wire. Will be different from |length| when
// reading a header-only RTP dump.
size_t original_length;
uint32_t time_ms;
};
class RtpFileReader {
public:
enum FileFormat { kPcap, kRtpDump, kLengthPacketInterleaved };
virtual ~RtpFileReader() {}
static RtpFileReader* Create(FileFormat format,
const uint8_t* data,
size_t size,
const std::set<uint32_t>& ssrc_filter);
static RtpFileReader* Create(FileFormat format, const std::string& filename);
static RtpFileReader* Create(FileFormat format,
const std::string& filename,
const std::set<uint32_t>& ssrc_filter);
virtual bool NextPacket(RtpPacket* packet) = 0;
};
} // namespace test
} // namespace webrtc
#endif // TEST_RTP_FILE_READER_H_