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/*
* Copyright (c) 2012 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 MODULES_AUDIO_CODING_TEST_RTPFILE_H_
#define MODULES_AUDIO_CODING_TEST_RTPFILE_H_
#include <stdio.h>
#include <cstdint>
#include <queue>
#include "absl/strings/string_view.h"
#include "api/rtp_headers.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class RTPStream {
public:
virtual ~RTPStream() {}
virtual void Write(uint8_t payloadType,
uint32_t timeStamp,
int16_t seqNo,
const uint8_t* payloadData,
size_t payloadSize,
uint32_t frequency) = 0;
// Returns the packet's payload size. Zero should be treated as an
// end-of-stream (in the case that EndOfFile() is true) or an error.
virtual size_t Read(RTPHeader* rtp_Header,
uint8_t* payloadData,
size_t payloadSize,
uint32_t* offset) = 0;
virtual bool EndOfFile() const = 0;
protected:
void MakeRTPheader(uint8_t* rtpHeader,
uint8_t payloadType,
int16_t seqNo,
uint32_t timeStamp,
uint32_t ssrc);
void ParseRTPHeader(RTPHeader* rtp_header, const uint8_t* rtpHeader);
};
class RTPPacket {
public:
RTPPacket(uint8_t payloadType,
uint32_t timeStamp,
int16_t seqNo,
const uint8_t* payloadData,
size_t payloadSize,
uint32_t frequency);
~RTPPacket();
uint8_t payloadType;
uint32_t timeStamp;
int16_t seqNo;
uint8_t* payloadData;
size_t payloadSize;
uint32_t frequency;
};
class RTPBuffer : public RTPStream {
public:
RTPBuffer() = default;
~RTPBuffer() = default;
void Write(uint8_t payloadType,
uint32_t timeStamp,
int16_t seqNo,
const uint8_t* payloadData,
size_t payloadSize,
uint32_t frequency) override;
size_t Read(RTPHeader* rtp_header,
uint8_t* payloadData,
size_t payloadSize,
uint32_t* offset) override;
bool EndOfFile() const override;
private:
mutable Mutex mutex_;
std::queue<RTPPacket*> _rtpQueue RTC_GUARDED_BY(&mutex_);
};
class RTPFile : public RTPStream {
public:
~RTPFile() {}
RTPFile() : _rtpFile(NULL), _rtpEOF(false) {}
void Open(absl::string_view outFilename, absl::string_view mode);
void Close();
void WriteHeader();
void ReadHeader();
void Write(uint8_t payloadType,
uint32_t timeStamp,
int16_t seqNo,
const uint8_t* payloadData,
size_t payloadSize,
uint32_t frequency) override;
size_t Read(RTPHeader* rtp_header,
uint8_t* payloadData,
size_t payloadSize,
uint32_t* offset) override;
bool EndOfFile() const override { return _rtpEOF; }
private:
FILE* _rtpFile;
bool _rtpEOF;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_TEST_RTPFILE_H_
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