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Reformat the WebRTC code base

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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}
This commit is contained in:
Yves Gerey
2018-06-19 15:03:05 +02:00
parent b602123a5a
commit 665174fdbb
1569 changed files with 30495 additions and 30309 deletions
Download Patch File Download Diff File Expand all files Collapse all files

44
modules/audio_coding/test/RTPFile.h
View File

@ -22,30 +22,40 @@ namespace webrtc {
class RTPStream {
public:
virtual ~RTPStream() {
}
virtual ~RTPStream() {}
virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
const int16_t seqNo, const uint8_t* payloadData,
const size_t payloadSize, uint32_t frequency) = 0;
virtual void Write(const uint8_t payloadType,
const uint32_t timeStamp,
const int16_t seqNo,
const uint8_t* payloadData,
const 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(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
size_t payloadSize, uint32_t* offset) = 0;
virtual size_t Read(WebRtcRTPHeader* rtpInfo,
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 MakeRTPheader(uint8_t* rtpHeader,
uint8_t payloadType,
int16_t seqNo,
uint32_t timeStamp,
uint32_t ssrc);
void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
};
class RTPPacket {
public:
RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
const uint8_t* payloadData, size_t payloadSize,
RTPPacket(uint8_t payloadType,
uint32_t timeStamp,
int16_t seqNo,
const uint8_t* payloadData,
size_t payloadSize,
uint32_t frequency);
~RTPPacket();
@ -80,20 +90,16 @@ class RTPBuffer : public RTPStream {
private:
RWLockWrapper* _queueRWLock;
std::queue<RTPPacket *> _rtpQueue;
std::queue<RTPPacket*> _rtpQueue;
};
class RTPFile : public RTPStream {
public:
~RTPFile() {
}
~RTPFile() {}
RTPFile()
: _rtpFile(NULL),
_rtpEOF(false) {
}
RTPFile() : _rtpFile(NULL), _rtpEOF(false) {}
void Open(const char *outFilename, const char *mode);
void Open(const char* outFilename, const char* mode);
void Close();