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WebRtc_Word32 => int32_t etc. in audio_coding/

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BUG=314

Review URL: https://webrtc-codereview.appspot.com/1271006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2013-04-09 00:28:06 +00:00
parent 6faf71d27b
commit 0946a56023
382 changed files with 8469 additions and 8488 deletions
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54
webrtc/modules/audio_coding/main/test/Channel.h
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@ -26,11 +26,11 @@ namespace webrtc {
struct ACMTestFrameSizeStats
{
WebRtc_UWord16 frameSizeSample;
WebRtc_Word16 maxPayloadLen;
WebRtc_UWord32 numPackets;
WebRtc_UWord64 totalPayloadLenByte;
WebRtc_UWord64 totalEncodedSamples;
uint16_t frameSizeSample;
int16_t maxPayloadLen;
uint32_t numPackets;
uint64_t totalPayloadLenByte;
uint64_t totalEncodedSamples;
double rateBitPerSec;
double usageLenSec;
@ -39,9 +39,9 @@ struct ACMTestFrameSizeStats
struct ACMTestPayloadStats
{
bool newPacket;
WebRtc_Word16 payloadType;
WebRtc_Word16 lastPayloadLenByte;
WebRtc_UWord32 lastTimestamp;
int16_t payloadType;
int16_t lastPayloadLenByte;
uint32_t lastTimestamp;
ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
};
@ -50,15 +50,15 @@ class Channel: public AudioPacketizationCallback
public:
Channel(
WebRtc_Word16 chID = -1);
int16_t chID = -1);
~Channel();
WebRtc_Word32 SendData(
int32_t SendData(
const FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize,
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
const uint16_t payloadSize,
const RTPFragmentationHeader* fragmentation);
void RegisterReceiverACM(
@ -66,16 +66,16 @@ public:
void ResetStats();
WebRtc_Word16 Stats(
int16_t Stats(
CodecInst& codecInst,
ACMTestPayloadStats& payloadStats);
void Stats(
WebRtc_UWord32* numPackets);
uint32_t* numPackets);
void Stats(
WebRtc_UWord8* payloadLenByte,
WebRtc_UWord32* payloadType);
uint8_t* payloadLenByte,
uint32_t* payloadType);
void PrintStats(
CodecInst& codecInst);
@ -85,7 +85,7 @@ public:
_isStereo = isStereo;
}
WebRtc_UWord32 LastInTimestamp();
uint32_t LastInTimestamp();
void SetFECTestWithPacketLoss(bool usePacketLoss)
{
@ -97,27 +97,27 @@ public:
private:
void CalcStatistics(
WebRtcRTPHeader& rtpInfo,
WebRtc_UWord16 payloadSize);
uint16_t payloadSize);
AudioCodingModule* _receiverACM;
WebRtc_UWord16 _seqNo;
uint16_t _seqNo;
// 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2];
uint8_t _payloadData[60 * 32 * 2 * 2];
CriticalSectionWrapper* _channelCritSect;
FILE* _bitStreamFile;
bool _saveBitStream;
WebRtc_Word16 _lastPayloadType;
int16_t _lastPayloadType;
ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
bool _isStereo;
WebRtcRTPHeader _rtpInfo;
bool _leftChannel;
WebRtc_UWord32 _lastInTimestamp;
uint32_t _lastInTimestamp;
// FEC Test variables
WebRtc_Word16 _packetLoss;
int16_t _packetLoss;
bool _useFECTestWithPacketLoss;
WebRtc_UWord64 _beginTime;
WebRtc_UWord64 _totalBytes;
uint64_t _beginTime;
uint64_t _totalBytes;
};
} // namespace webrtc