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62238097c9
This is a safe cleanup change since top-level const applied to parameters in function declarations (that are not also definitions) are ignored by the compiler. Hence, such changes do not change the type of the declared functions and are simply no-ops. Bug: webrtc:13610 Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35802}
132 lines
3.3 KiB
C++
132 lines
3.3 KiB
C++
/*
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* Copyright (c) 2012 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 MODULES_AUDIO_CODING_TEST_RTPFILE_H_
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#define MODULES_AUDIO_CODING_TEST_RTPFILE_H_
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#include <stdio.h>
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#include <queue>
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#include "api/rtp_headers.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class RTPStream {
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public:
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virtual ~RTPStream() {}
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virtual void Write(uint8_t payloadType,
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uint32_t timeStamp,
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int16_t seqNo,
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const uint8_t* payloadData,
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size_t payloadSize,
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uint32_t frequency) = 0;
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// Returns the packet's payload size. Zero should be treated as an
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// end-of-stream (in the case that EndOfFile() is true) or an error.
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virtual size_t Read(RTPHeader* rtp_Header,
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uint8_t* payloadData,
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size_t payloadSize,
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uint32_t* offset) = 0;
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virtual bool EndOfFile() const = 0;
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protected:
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void MakeRTPheader(uint8_t* rtpHeader,
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uint8_t payloadType,
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int16_t seqNo,
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uint32_t timeStamp,
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uint32_t ssrc);
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void ParseRTPHeader(RTPHeader* rtp_header, const uint8_t* rtpHeader);
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};
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class RTPPacket {
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public:
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RTPPacket(uint8_t payloadType,
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uint32_t timeStamp,
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int16_t seqNo,
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const uint8_t* payloadData,
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size_t payloadSize,
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uint32_t frequency);
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~RTPPacket();
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uint8_t payloadType;
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uint32_t timeStamp;
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int16_t seqNo;
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uint8_t* payloadData;
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size_t payloadSize;
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uint32_t frequency;
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};
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class RTPBuffer : public RTPStream {
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public:
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RTPBuffer() = default;
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~RTPBuffer() = default;
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void Write(uint8_t payloadType,
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uint32_t timeStamp,
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int16_t seqNo,
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const uint8_t* payloadData,
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size_t payloadSize,
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uint32_t frequency) override;
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size_t Read(RTPHeader* rtp_header,
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uint8_t* payloadData,
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size_t payloadSize,
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uint32_t* offset) override;
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bool EndOfFile() const override;
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private:
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mutable Mutex mutex_;
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std::queue<RTPPacket*> _rtpQueue RTC_GUARDED_BY(&mutex_);
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};
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class RTPFile : public RTPStream {
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public:
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~RTPFile() {}
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RTPFile() : _rtpFile(NULL), _rtpEOF(false) {}
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void Open(const char* outFilename, const char* mode);
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void Close();
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void WriteHeader();
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void ReadHeader();
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void Write(uint8_t payloadType,
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uint32_t timeStamp,
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int16_t seqNo,
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const uint8_t* payloadData,
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size_t payloadSize,
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uint32_t frequency) override;
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size_t Read(RTPHeader* rtp_header,
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uint8_t* payloadData,
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size_t payloadSize,
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uint32_t* offset) override;
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bool EndOfFile() const override { return _rtpEOF; }
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private:
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FILE* _rtpFile;
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bool _rtpEOF;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_TEST_RTPFILE_H_
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