Formatting ACM tests
Pure formatting of all files located in /webrtc/modules/audio_coding/main/test/
Smaller manual modifications done after using Eclipse formatting tool, like wrapping long lines (mostly comments).
BUG=issue1024
Review URL: https://webrtc-codereview.appspot.com/1342004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3946 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/test/Channel.h b/webrtc/modules/audio_coding/main/test/Channel.h
index c0bf7f3..b4b51cf 100644
--- a/webrtc/modules/audio_coding/main/test/Channel.h
+++ b/webrtc/modules/audio_coding/main/test/Channel.h
@@ -23,103 +23,83 @@
#define MAX_NUM_PAYLOADS 50
#define MAX_NUM_FRAMESIZES 6
-
-struct ACMTestFrameSizeStats
-{
- uint16_t frameSizeSample;
- int16_t maxPayloadLen;
- uint32_t numPackets;
- uint64_t totalPayloadLenByte;
- uint64_t totalEncodedSamples;
- double rateBitPerSec;
- double usageLenSec;
-
+struct ACMTestFrameSizeStats {
+ uint16_t frameSizeSample;
+ int16_t maxPayloadLen;
+ uint32_t numPackets;
+ uint64_t totalPayloadLenByte;
+ uint64_t totalEncodedSamples;
+ double rateBitPerSec;
+ double usageLenSec;
};
-struct ACMTestPayloadStats
-{
- bool newPacket;
- int16_t payloadType;
- int16_t lastPayloadLenByte;
- uint32_t lastTimestamp;
- ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
+struct ACMTestPayloadStats {
+ bool newPacket;
+ int16_t payloadType;
+ int16_t lastPayloadLenByte;
+ uint32_t lastTimestamp;
+ ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
};
-class Channel: public AudioPacketizationCallback
-{
-public:
+class Channel : public AudioPacketizationCallback {
+ public:
- Channel(
- int16_t chID = -1);
- ~Channel();
+ Channel(int16_t chID = -1);
+ ~Channel();
- int32_t SendData(
- const FrameType frameType,
- const uint8_t payloadType,
- const uint32_t timeStamp,
- const uint8_t* payloadData,
- const uint16_t payloadSize,
- const RTPFragmentationHeader* fragmentation);
+ int32_t SendData(const FrameType frameType, const uint8_t payloadType,
+ const uint32_t timeStamp, const uint8_t* payloadData,
+ const uint16_t payloadSize,
+ const RTPFragmentationHeader* fragmentation);
- void RegisterReceiverACM(
- AudioCodingModule *acm);
+ void RegisterReceiverACM(AudioCodingModule *acm);
- void ResetStats();
+ void ResetStats();
- int16_t Stats(
- CodecInst& codecInst,
- ACMTestPayloadStats& payloadStats);
+ int16_t Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats);
- void Stats(
- uint32_t* numPackets);
+ void Stats(uint32_t* numPackets);
- void Stats(
- uint8_t* payloadLenByte,
- uint32_t* payloadType);
+ void Stats(uint8_t* payloadLenByte, uint32_t* payloadType);
- void PrintStats(
- CodecInst& codecInst);
+ void PrintStats(CodecInst& codecInst);
- void SetIsStereo(bool isStereo)
- {
- _isStereo = isStereo;
- }
+ void SetIsStereo(bool isStereo) {
+ _isStereo = isStereo;
+ }
- uint32_t LastInTimestamp();
+ uint32_t LastInTimestamp();
- void SetFECTestWithPacketLoss(bool usePacketLoss)
- {
- _useFECTestWithPacketLoss = usePacketLoss;
- }
+ void SetFECTestWithPacketLoss(bool usePacketLoss) {
+ _useFECTestWithPacketLoss = usePacketLoss;
+ }
- double BitRate();
+ double BitRate();
-private:
- void CalcStatistics(
- WebRtcRTPHeader& rtpInfo,
- uint16_t payloadSize);
+ private:
+ void CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize);
- AudioCodingModule* _receiverACM;
- uint16_t _seqNo;
- // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
- uint8_t _payloadData[60 * 32 * 2 * 2];
+ AudioCodingModule* _receiverACM;
+ uint16_t _seqNo;
+ // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
+ uint8_t _payloadData[60 * 32 * 2 * 2];
- CriticalSectionWrapper* _channelCritSect;
- FILE* _bitStreamFile;
- bool _saveBitStream;
- int16_t _lastPayloadType;
- ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
- bool _isStereo;
- WebRtcRTPHeader _rtpInfo;
- bool _leftChannel;
- uint32_t _lastInTimestamp;
- // FEC Test variables
- int16_t _packetLoss;
- bool _useFECTestWithPacketLoss;
- uint64_t _beginTime;
- uint64_t _totalBytes;
+ CriticalSectionWrapper* _channelCritSect;
+ FILE* _bitStreamFile;
+ bool _saveBitStream;
+ int16_t _lastPayloadType;
+ ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
+ bool _isStereo;
+ WebRtcRTPHeader _rtpInfo;
+ bool _leftChannel;
+ uint32_t _lastInTimestamp;
+ // FEC Test variables
+ int16_t _packetLoss;
+ bool _useFECTestWithPacketLoss;
+ uint64_t _beginTime;
+ uint64_t _totalBytes;
};
-} // namespace webrtc
+} // namespace webrtc
#endif