Reformat the WebRTC code base
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}diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index 7d5e6e2..8fdb677 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -30,11 +30,15 @@
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 @@
// 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 @@
_channelCritSect.Enter();
if (_saveBitStream) {
- //fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
+ // fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
}
if (!_isStereo) {
@@ -128,8 +132,8 @@
// 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 @@
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 @@
_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 @@
// 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 @@
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 @@
_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() {
-}
+Channel::~Channel() {}
void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
_receiverACM = acm;
@@ -311,13 +315,13 @@
_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 @@
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 @@
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 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;
}