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665174fdbb
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
131 lines
3.4 KiB
C++
131 lines
3.4 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 "modules/audio_coding/include/audio_coding_module.h"
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#include "rtc_base/synchronization/rw_lock_wrapper.h"
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#include "typedefs.h" // NOLINT(build/include)
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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(const uint8_t payloadType,
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const uint32_t timeStamp,
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const int16_t seqNo,
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const uint8_t* payloadData,
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const 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(WebRtcRTPHeader* rtpInfo,
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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(WebRtcRTPHeader* rtpInfo, 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();
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~RTPBuffer();
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void Write(const uint8_t payloadType,
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const uint32_t timeStamp,
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const int16_t seqNo,
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const uint8_t* payloadData,
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const size_t payloadSize,
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uint32_t frequency) override;
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size_t Read(WebRtcRTPHeader* rtpInfo,
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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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RWLockWrapper* _queueRWLock;
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std::queue<RTPPacket*> _rtpQueue;
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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(const uint8_t payloadType,
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const uint32_t timeStamp,
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const int16_t seqNo,
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const uint8_t* payloadData,
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const size_t payloadSize,
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uint32_t frequency) override;
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size_t Read(WebRtcRTPHeader* rtpInfo,
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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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