Bug 1338086 - Remove useless else blocks in order to reduce complexity in media/webrtc/signaling/ r?jesup
MozReview-Commit-ID: EU5B0cUYp6c
--- a/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp
+++ b/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp
@@ -2491,33 +2491,27 @@ JsepSessionImpl::EnableOfferMsection(Sdp
return NS_OK;
}
mozilla::Sdp*
JsepSessionImpl::GetParsedLocalDescription() const
{
if (mPendingLocalDescription) {
return mPendingLocalDescription.get();
- } else if (mCurrentLocalDescription) {
- return mCurrentLocalDescription.get();
}
-
- return nullptr;
+ return mCurrentLocalDescription.get();
}
mozilla::Sdp*
JsepSessionImpl::GetParsedRemoteDescription() const
{
if (mPendingRemoteDescription) {
return mPendingRemoteDescription.get();
- } else if (mCurrentRemoteDescription) {
- return mCurrentRemoteDescription.get();
}
-
- return nullptr;
+ return mCurrentRemoteDescription.get();
}
const Sdp*
JsepSessionImpl::GetAnswer() const
{
return mWasOffererLastTime ? mCurrentRemoteDescription.get()
: mCurrentLocalDescription.get();
}
old mode 100755
new mode 100644
--- a/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
+++ b/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
@@ -531,29 +531,26 @@ WebrtcAudioConduit::ConfigureRecvMediaCo
continue;
}
if(mPtrVoECodec->SetRecPayloadType(mChannel,cinst) == -1)
{
error = mPtrVoEBase->LastError();
CSFLogError(logTag, "%s SetRecvCodec Failed %d ",__FUNCTION__, error);
continue;
- } else {
- CSFLogDebug(logTag, "%s Successfully Set RecvCodec %s", __FUNCTION__,
- codec->mName.c_str());
- //copy this to local database
- if(CopyCodecToDB(codec))
- {
- success = true;
- } else {
+ }
+ CSFLogDebug(logTag, "%s Successfully Set RecvCodec %s", __FUNCTION__,
+ codec->mName.c_str());
+
+ //copy this to local database
+ if(!CopyCodecToDB(codec)) {
CSFLogError(logTag,"%s Unable to updated Codec Database", __FUNCTION__);
return kMediaConduitUnknownError;
- }
-
}
+ success = true;
} //end for
if(!success)
{
CSFLogError(logTag, "%s Setting Receive Codec Failed ", __FUNCTION__);
return kMediaConduitInvalidReceiveCodec;
}
@@ -922,65 +919,64 @@ WebrtcAudioConduit::SendRtp(const uint8_
}
}
#endif
ReentrantMonitorAutoEnter enter(mTransportMonitor);
// XXX(pkerr) - the PacketOptions are being ignored. This parameter was added along
// with the Call API update in the webrtc.org codebase.
// The only field in it is the packet_id, which is used when the header
// extension for TransportSequenceNumber is being used, which we don't.
- (void) options;
+ (void)options;
if(mTransmitterTransport &&
(mTransmitterTransport->SendRtpPacket(data, len) == NS_OK))
{
CSFLogDebug(logTag, "%s Sent RTP Packet ", __FUNCTION__);
return true;
- } else {
- CSFLogError(logTag, "%s RTP Packet Send Failed ", __FUNCTION__);
- return false;
}
+ CSFLogError(logTag, "%s RTP Packet Send Failed ", __FUNCTION__);
+ return false;
}
// Called on WebRTC Process thread and perhaps others
bool
WebrtcAudioConduit::SendRtcp(const uint8_t* data, size_t len)
{
- CSFLogDebug(logTag, "%s : len %lu, first rtcp = %u ",
+ CSFLogDebug(logTag, "%s : len %lu, first rtcp = %u ",
__FUNCTION__,
(unsigned long) len,
static_cast<unsigned>(data[1]));
// We come here if we have only one pipeline/conduit setup,
// such as for unidirectional streams.
// We also end up here if we are receiving
ReentrantMonitorAutoEnter enter(mTransportMonitor);
if(mReceiverTransport &&
mReceiverTransport->SendRtcpPacket(data, len) == NS_OK)
{
// Might be a sender report, might be a receiver report, we don't know.
CSFLogDebug(logTag, "%s Sent RTCP Packet ", __FUNCTION__);
return true;
- } else if(mTransmitterTransport &&
- (mTransmitterTransport->SendRtcpPacket(data, len) == NS_OK)) {
- CSFLogDebug(logTag, "%s Sent RTCP Packet (sender report) ", __FUNCTION__);
- return true;
- } else {
- CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
- return false;
}
+ if (mTransmitterTransport &&
+ (mTransmitterTransport->SendRtcpPacket(data, len) == NS_OK)) {
+ CSFLogDebug(logTag, "%s Sent RTCP Packet (sender report) ", __FUNCTION__);
+ return true;
+ }
+ CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
+ return false;
}
/**
* Converts between CodecConfig to WebRTC Codec Structure.
*/
bool
WebrtcAudioConduit::CodecConfigToWebRTCCodec(const AudioCodecConfig* codecInfo,
webrtc::CodecInst& cinst)
- {
+{
const unsigned int plNameLength = codecInfo->mName.length();
memset(&cinst, 0, sizeof(webrtc::CodecInst));
if(sizeof(cinst.plname) < plNameLength+1)
{
CSFLogError(logTag, "%s Payload name buffer capacity mismatch ",
__FUNCTION__);
return false;
}
@@ -991,37 +987,32 @@ WebrtcAudioConduit::CodecConfigToWebRTCC
cinst.pacsize = codecInfo->mPacSize;
cinst.plfreq = codecInfo->mFreq;
if (codecInfo->mName == "G722") {
// Compensate for G.722 spec error in RFC 1890
cinst.plfreq = 16000;
}
cinst.channels = codecInfo->mChannels;
return true;
- }
+}
/**
- * Supported Sampling Frequncies.
+ * Supported Sampling Frequencies.
*/
bool
WebrtcAudioConduit::IsSamplingFreqSupported(int freq) const
{
- if(GetNum10msSamplesForFrequency(freq))
- {
- return true;
- } else {
- return false;
- }
+ return GetNum10msSamplesForFrequency(freq) != 0;
}
/* Return block-length of 10 ms audio frame in number of samples */
unsigned int
WebrtcAudioConduit::GetNum10msSamplesForFrequency(int samplingFreqHz) const
{
- switch(samplingFreqHz)
+ switch (samplingFreqHz)
{
case 16000: return 160; //160 samples
case 32000: return 320; //320 samples
case 44100: return 441; //441 samples
case 48000: return 480; //480 samples
default: return 0; // invalid or unsupported
}
}
--- a/media/webrtc/signaling/src/media-conduit/VideoConduit.cpp
+++ b/media/webrtc/signaling/src/media-conduit/VideoConduit.cpp
@@ -308,22 +308,21 @@ WebrtcVideoConduit::ConfigureCodecMode(w
return kMediaConduitMalformedArgument;
}
webrtc::VideoEncoder::EncoderType
PayloadNameToEncoderType(const std::string& name)
{
if ("VP8" == name) {
return webrtc::VideoEncoder::EncoderType::kVp8;
- } else if ("VP9" == name) {
+ } else if ("VP9" == name) { // NOLINT(readability-else-after-return)
return webrtc::VideoEncoder::EncoderType::kVp9;
- } else if ("H264" == name) {
+ } else if ("H264" == name) { // NOLINT(readability-else-after-return)
return webrtc::VideoEncoder::EncoderType::kH264;
}
-
return webrtc::VideoEncoder::EncoderType::kUnsupportedCodec;
}
void
WebrtcVideoConduit::DeleteSendStream()
{
if (mSendStream) {
@@ -377,22 +376,21 @@ WebrtcVideoConduit::CreateSendStream()
return kMediaConduitNoError;
}
webrtc::VideoDecoder::DecoderType
PayloadNameToDecoderType(const std::string& name)
{
if ("VP8" == name) {
return webrtc::VideoDecoder::DecoderType::kVp8;
- } else if ("VP9" == name) {
+ } else if ("VP9" == name) { // NOLINT(readability-else-after-return)
return webrtc::VideoDecoder::DecoderType::kVp9;
- } else if ("H264" == name) {
+ } else if ("H264" == name) { // NOLINT(readability-else-after-return)
return webrtc::VideoDecoder::DecoderType::kH264;
}
-
return webrtc::VideoDecoder::DecoderType::kUnsupportedCodec;
}
void
WebrtcVideoConduit::DeleteRecvStream()
{
if (mRecvStream) {
mCall->Call()->DestroyVideoReceiveStream(mRecvStream);
@@ -1882,17 +1880,18 @@ WebrtcVideoConduit::SendRtcp(const uint8
// We also end up here if we are receiving
ReentrantMonitorAutoEnter enter(mTransportMonitor);
if (mReceiverTransport &&
NS_SUCCEEDED(mReceiverTransport->SendRtcpPacket(packet, length)))
{
// Might be a sender report, might be a receiver report, we don't know.
CSFLogDebug(logTag, "%s Sent RTCP Packet ", __FUNCTION__);
return true;
- } else if (mTransmitterTransport &&
+ }
+ if (mTransmitterTransport &&
NS_SUCCEEDED(mTransmitterTransport->SendRtcpPacket(packet, length))) {
CSFLogDebug(logTag, "%s Sent RTCP Packet (sender report) ", __FUNCTION__);
return true;
}
CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
return false;
}
@@ -2009,17 +2008,18 @@ WebrtcVideoConduit::MozVideoLatencyAvg()
return mVideoLatencyAvg / sRoundingPadding;
}
uint64_t
WebrtcVideoConduit::CodecPluginID()
{
if (mSendCodecPlugin) {
return mSendCodecPlugin->PluginID();
- } else if (mRecvCodecPlugin) {
+ }
+ if (mRecvCodecPlugin) {
return mRecvCodecPlugin->PluginID();
}
return 0;
}
bool
WebrtcVideoConduit::RequiresNewSendStream(const VideoCodecConfig& newConfig) const
--- a/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.cpp
+++ b/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.cpp
@@ -19,22 +19,21 @@ MediaPipelineFilter::MediaPipelineFilter
bool MediaPipelineFilter::Filter(const webrtc::RTPHeader& header,
uint32_t correlator) {
if (correlator) {
// This special correlator header takes precedence. It also lets us learn
// about SSRC mappings if we don't know about them yet.
if (correlator == correlator_) {
AddRemoteSSRC(header.ssrc);
return true;
- } else {
- // Some other stream; it is possible that an SSRC has moved, so make sure
- // we don't have that SSRC in our filter any more.
- remote_ssrc_set_.erase(header.ssrc);
- return false;
}
+ // Some other stream; it is possible that an SSRC has moved, so make sure
+ // we don't have that SSRC in our filter any more.
+ remote_ssrc_set_.erase(header.ssrc);
+ return false;
}
if (remote_ssrc_set_.count(header.ssrc)) {
return true;
}
// Last ditch effort...
if (payload_type_set_.count(header.payloadType)) {
--- a/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp
+++ b/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp
@@ -415,17 +415,18 @@ RunStatsQuery(
return rv;
}
nsCOMPtr<nsIEventTarget> stsThread =
do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
if (NS_FAILED(rv)) {
return rv;
- } else if (!stsThread) {
+ }
+ if (!stsThread) {
return NS_ERROR_FAILURE;
}
rv = RUN_ON_THREAD(stsThread,
WrapRunnableNM(&GetAllStats_s,
aThisChild,
aRequestId,
queries),
@@ -537,17 +538,18 @@ RunLogQuery(const nsCString& aPattern,
const int aRequestId)
{
nsresult rv;
nsCOMPtr<nsIEventTarget> stsThread =
do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
if (NS_FAILED(rv)) {
return rv;
- } else if (!stsThread) {
+ }
+ if (!stsThread) {
return NS_ERROR_FAILURE;
}
rv = RUN_ON_THREAD(stsThread,
WrapRunnableNM(&GetLogging_s,
aThisChild,
aRequestId,
aPattern.get()),