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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

87
modules/audio_coding/test/Channel.cc
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@ -30,11 +30,15 @@ int32_t Channel::SendData(FrameType frameType,
rtpInfo.header.markerBit = false;
rtpInfo.header.ssrc = 0;
rtpInfo.header.sequenceNumber = (external_sequence_number_ < 0) ?
_seqNo++ : static_cast<uint16_t>(external_sequence_number_);
rtpInfo.header.sequenceNumber =
(external_sequence_number_ < 0)
? _seqNo++
: static_cast<uint16_t>(external_sequence_number_);
rtpInfo.header.payloadType = payloadType;
rtpInfo.header.timestamp = (external_send_timestamp_ < 0) ? timeStamp :
static_cast<uint32_t>(external_send_timestamp_);
rtpInfo.header.timestamp =
(external_send_timestamp_ < 0)
? timeStamp
: static_cast<uint32_t>(external_send_timestamp_);
if (frameType == kAudioFrameCN) {
rtpInfo.type.Audio.isCNG = true;
@ -57,7 +61,7 @@ int32_t Channel::SendData(FrameType frameType,
// only 0x80 if we have multiple blocks
_payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
size_t REDheader = (fragmentation->fragmentationTimeDiff[1] << 10) +
fragmentation->fragmentationLength[1];
fragmentation->fragmentationLength[1];
_payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
_payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
_payloadData[3] = uint8_t(REDheader & 0x000000FF);
@ -96,7 +100,7 @@ int32_t Channel::SendData(FrameType frameType,
_channelCritSect.Enter();
if (_saveBitStream) {
//fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
// fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
}
if (!_isStereo) {
@ -128,8 +132,8 @@ int32_t Channel::SendData(FrameType frameType,
// TODO(turajs): rewite this method.
void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
int n;
if ((rtpInfo.header.payloadType != _lastPayloadType)
&& (_lastPayloadType != -1)) {
if ((rtpInfo.header.payloadType != _lastPayloadType) &&
(_lastPayloadType != -1)) {
// payload-type is changed.
// we have to terminate the calculations on the previous payload type
// we ignore the last packet in that payload type just to make things
@ -156,14 +160,15 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
if (!newPayload) {
if (!currentPayloadStr->newPacket) {
if (!_useLastFrameSize) {
_lastFrameSizeSample = (uint32_t) ((uint32_t) rtpInfo.header.timestamp -
(uint32_t) currentPayloadStr->lastTimestamp);
_lastFrameSizeSample =
(uint32_t)((uint32_t)rtpInfo.header.timestamp -
(uint32_t)currentPayloadStr->lastTimestamp);
}
assert(_lastFrameSizeSample > 0);
int k = 0;
for (; k < MAX_NUM_FRAMESIZES; ++k) {
if ((currentPayloadStr->frameSizeStats[k].frameSizeSample ==
_lastFrameSizeSample) ||
_lastFrameSizeSample) ||
(currentPayloadStr->frameSizeStats[k].frameSizeSample == 0)) {
break;
}
@ -174,9 +179,9 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
_lastPayloadType, _lastFrameSizeSample);
return;
}
ACMTestFrameSizeStats* currentFrameSizeStats = &(currentPayloadStr
->frameSizeStats[k]);
currentFrameSizeStats->frameSizeSample = (int16_t) _lastFrameSizeSample;
ACMTestFrameSizeStats* currentFrameSizeStats =
&(currentPayloadStr->frameSizeStats[k]);
currentFrameSizeStats->frameSizeSample = (int16_t)_lastFrameSizeSample;
// increment the number of encoded samples.
currentFrameSizeStats->totalEncodedSamples += _lastFrameSizeSample;
@ -185,15 +190,15 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
// increment the total number of bytes (this is based on
// the previous payload we don't know the frame-size of
// the current payload.
currentFrameSizeStats->totalPayloadLenByte += currentPayloadStr
->lastPayloadLenByte;
currentFrameSizeStats->totalPayloadLenByte +=
currentPayloadStr->lastPayloadLenByte;
// store the maximum payload-size (this is based on
// the previous payload we don't know the frame-size of
// the current payload.
if (currentFrameSizeStats->maxPayloadLen
< currentPayloadStr->lastPayloadLenByte) {
currentFrameSizeStats->maxPayloadLen = currentPayloadStr
->lastPayloadLenByte;
if (currentFrameSizeStats->maxPayloadLen <
currentPayloadStr->lastPayloadLenByte) {
currentFrameSizeStats->maxPayloadLen =
currentPayloadStr->lastPayloadLenByte;
}
// store the current values for the next time
currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
@ -203,8 +208,8 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
currentPayloadStr->lastPayloadLenByte = payloadSize;
currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
currentPayloadStr->payloadType = rtpInfo.header.payloadType;
memset(currentPayloadStr->frameSizeStats, 0, MAX_NUM_FRAMESIZES *
sizeof(ACMTestFrameSizeStats));
memset(currentPayloadStr->frameSizeStats, 0,
MAX_NUM_FRAMESIZES * sizeof(ACMTestFrameSizeStats));
}
} else {
n = 0;
@ -216,8 +221,8 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
_payloadStats[n].lastPayloadLenByte = payloadSize;
_payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
_payloadStats[n].payloadType = rtpInfo.header.payloadType;
memset(_payloadStats[n].frameSizeStats, 0, MAX_NUM_FRAMESIZES *
sizeof(ACMTestFrameSizeStats));
memset(_payloadStats[n].frameSizeStats, 0,
MAX_NUM_FRAMESIZES * sizeof(ACMTestFrameSizeStats));
}
}
@ -262,8 +267,7 @@ Channel::Channel(int16_t chID)
}
}
Channel::~Channel() {
}
Channel::~Channel() {}
void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
_receiverACM = acm;
@ -311,13 +315,13 @@ int16_t Channel::Stats(CodecInst& codecInst,
_channelCritSect.Leave();
return 0;
}
payloadStats.frameSizeStats[n].usageLenSec = (double) payloadStats
.frameSizeStats[n].totalEncodedSamples / (double) codecInst.plfreq;
payloadStats.frameSizeStats[n].usageLenSec =
(double)payloadStats.frameSizeStats[n].totalEncodedSamples /
(double)codecInst.plfreq;
payloadStats.frameSizeStats[n].rateBitPerSec =
payloadStats.frameSizeStats[n].totalPayloadLenByte * 8
/ payloadStats.frameSizeStats[n].usageLenSec;
payloadStats.frameSizeStats[n].totalPayloadLenByte * 8 /
payloadStats.frameSizeStats[n].usageLenSec;
}
_channelCritSect.Leave();
return 0;
@ -353,14 +357,14 @@ void Channel::Stats(uint8_t* payloadType, uint32_t* payloadLenByte) {
if (_payloadStats[k].payloadType == -1) {
break;
}
payloadType[k] = (uint8_t) _payloadStats[k].payloadType;
payloadType[k] = (uint8_t)_payloadStats[k].payloadType;
payloadLenByte[k] = 0;
for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
break;
}
payloadLenByte[k] += (uint16_t) _payloadStats[k].frameSizeStats[n]
.totalPayloadLenByte;
payloadLenByte[k] +=
(uint16_t)_payloadStats[k].frameSizeStats[n].totalPayloadLenByte;
}
}
@ -387,18 +391,15 @@ void Channel::PrintStats(CodecInst& codecInst) {
payloadStats.frameSizeStats[k].rateBitPerSec);
printf("Maximum Payload-Size.......... %" PRIuS " Bytes\n",
payloadStats.frameSizeStats[k].maxPayloadLen);
printf(
"Maximum Instantaneous Rate.... %.0f bits/sec\n",
((double) payloadStats.frameSizeStats[k].maxPayloadLen * 8.0
* (double) codecInst.plfreq)
/ (double) payloadStats.frameSizeStats[k].frameSizeSample);
printf("Maximum Instantaneous Rate.... %.0f bits/sec\n",
((double)payloadStats.frameSizeStats[k].maxPayloadLen * 8.0 *
(double)codecInst.plfreq) /
(double)payloadStats.frameSizeStats[k].frameSizeSample);
printf("Number of Packets............. %u\n",
(unsigned int) payloadStats.frameSizeStats[k].numPackets);
(unsigned int)payloadStats.frameSizeStats[k].numPackets);
printf("Duration...................... %0.3f sec\n\n",
payloadStats.frameSizeStats[k].usageLenSec);
}
}
uint32_t Channel::LastInTimestamp() {
@ -413,7 +414,7 @@ double Channel::BitRate() {
double rate;
uint64_t currTime = rtc::TimeMillis();
_channelCritSect.Enter();
rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
rate = ((double)_totalBytes * 8.0) / (double)(currTime - _beginTime);
_channelCritSect.Leave();
return rate;
}