自学教程：C++ CSFLogError函数代码示例

nsresult SrtpFlow::ProtectRtp(void *in, int in_len,                              int max_len, int *out_len) {  nsresult res = CheckInputs(true, in, in_len, max_len, out_len);  if (NS_FAILED(res))    return res;  int len = in_len;  srtp_err_status_t r = srtp_protect(session_, in, &len);  if (r != srtp_err_status_ok) {    CSFLogError(LOGTAG, "Error protecting SRTP packet");    return NS_ERROR_FAILURE;  }  MOZ_ASSERT(len <= max_len);  *out_len = len;  CSFLogDebug(LOGTAG, "Successfully protected an SRTP packet of len %d",              *out_len);  return NS_OK;}

//WebRTC::RTP Callback Implementationint WebrtcAudioConduit::SendPacket(int channel, const void* data, int len){  CSFLogDebug(logTag,  "%s : channel %d %s", __FUNCTION__, channel,              (mEngineReceiving && mOtherDirection) ? "(using mOtherDirection)" : "");  if (mEngineReceiving)  {    if (mOtherDirection)    {      return mOtherDirection->SendPacket(channel, data, len);    }    CSFLogDebug(logTag,  "%s : Asked to send RTP without an RTP sender on channel %d",                __FUNCTION__, channel);    return -1;  } else {#ifdef MOZILLA_INTERNAL_API    if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG)) {      if (mProcessing.Length() > 0) {        TimeStamp started = mProcessing[0].mTimeStamp;        mProcessing.RemoveElementAt(0);        mProcessing.RemoveElementAt(0); // 20ms packetization!  Could automate this by watching sizes        TimeDuration t = TimeStamp::Now() - started;        int64_t delta = t.ToMilliseconds();        LogTime(AsyncLatencyLogger::AudioSendRTP, ((uint64_t) this), delta);      }    }#endif    if(mTransport && (mTransport->SendRtpPacket(data, len) == NS_OK))    {      CSFLogDebug(logTag, "%s Sent RTP Packet ", __FUNCTION__);      return len;    } else {      CSFLogError(logTag, "%s RTP Packet Send Failed ", __FUNCTION__);      return -1;    }  }}

voidRemoteSourceStreamInfo::StorePipeline(int aTrack,                                      bool aIsVideo,                                      mozilla::RefPtr<mozilla::MediaPipeline> aPipeline){  MOZ_ASSERT(mPipelines.find(aTrack) == mPipelines.end());  if (mPipelines.find(aTrack) != mPipelines.end()) {    CSFLogError(logTag, "%s: Request to store duplicate track %d", __FUNCTION__, aTrack);    return;  }  CSFLogDebug(logTag, "%s track %d %s = %p", __FUNCTION__, aTrack, aIsVideo ? "video" : "audio",              aPipeline.get());  // See if we have both audio and video here, and if so cross the streams and sync them  // XXX Needs to be adjusted when we support multiple streams of the same type  for (std::map<int, bool>::iterator it = mTypes.begin(); it != mTypes.end(); ++it) {    if (it->second != aIsVideo) {      // Ok, we have one video, one non-video - cross the streams!      mozilla::WebrtcAudioConduit *audio_conduit = static_cast<mozilla::WebrtcAudioConduit*>                                                   (aIsVideo ?                                                    mPipelines[it->first]->Conduit() :                                                    aPipeline->Conduit());      mozilla::WebrtcVideoConduit *video_conduit = static_cast<mozilla::WebrtcVideoConduit*>                                                   (aIsVideo ?                                                    aPipeline->Conduit() :                                                    mPipelines[it->first]->Conduit());      video_conduit->SyncTo(audio_conduit);      CSFLogDebug(logTag, "Syncing %p to %p, %d to %d", video_conduit, audio_conduit,                  aTrack, it->first);    }  }  //TODO: Revisit once we start supporting multiple streams or multiple tracks  // of same type  mPipelines[aTrack] = aPipeline;  //TODO: move to attribute on Pipeline  mTypes[aTrack] = aIsVideo;}

int WebrtcAudioConduit::SendRTCPPacket(int channel, const void* data, int len){  CSFLogDebug(logTag,  "%s : channel %d", __FUNCTION__, channel);  if (mEngineTransmitting)  {    if (mOtherDirection)    {      return mOtherDirection->SendRTCPPacket(channel, data, len);    }    CSFLogDebug(logTag,  "%s : Asked to send RTCP without an RTP receiver on channel %d",                __FUNCTION__, channel);    return -1;  } else {    if(mTransport && mTransport->SendRtcpPacket(data, len) == NS_OK)    {      CSFLogDebug(logTag, "%s Sent RTCP Packet ", __FUNCTION__);      return len;    } else {      CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);      return -1;    }  }}

nsresultPeerConnectionMedia::InitProxy(){#if !defined(MOZILLA_EXTERNAL_LINKAGE)  // Allow mochitests to disable this, since mochitest configures a fake proxy  // that serves up content.  bool disable = Preferences::GetBool("media.peerconnection.disable_http_proxy",                                      false);  if (disable) {    mProxyResolveCompleted = true;    return NS_OK;  }#endif  nsresult rv;  nsCOMPtr<nsIProtocolProxyService> pps =    do_GetService(NS_PROTOCOLPROXYSERVICE_CONTRACTID, &rv);  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to get proxy service: %d", __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }  // We use the following URL to find the "default" proxy address for all HTTPS  // connections.  We will only attempt one HTTP(S) CONNECT per peer connection.  // "example.com" is guaranteed to be unallocated and should return the best default.  nsCOMPtr<nsIURI> fakeHttpsLocation;  rv = NS_NewURI(getter_AddRefs(fakeHttpsLocation), "https://example.com");  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to set URI: %d", __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }  nsCOMPtr<nsIScriptSecurityManager> secMan(      do_GetService(NS_SCRIPTSECURITYMANAGER_CONTRACTID, &rv));  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to get IOService: %d",        __FUNCTION__, (int)rv);    CSFLogError(logTag, "%s: Failed to get securityManager: %d", __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }  nsCOMPtr<nsIPrincipal> systemPrincipal;  rv = secMan->GetSystemPrincipal(getter_AddRefs(systemPrincipal));  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to get systemPrincipal: %d", __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }  nsCOMPtr<nsIChannel> channel;  rv = NS_NewChannel(getter_AddRefs(channel),                     fakeHttpsLocation,                     systemPrincipal,                     nsILoadInfo::SEC_ALLOW_CROSS_ORIGIN_DATA_IS_NULL,                     nsIContentPolicy::TYPE_OTHER);  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to get channel from URI: %d",                __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }  RefPtr<ProtocolProxyQueryHandler> handler = new ProtocolProxyQueryHandler(this);  rv = pps->AsyncResolve(channel,                         nsIProtocolProxyService::RESOLVE_PREFER_HTTPS_PROXY |                         nsIProtocolProxyService::RESOLVE_ALWAYS_TUNNEL,                         handler, getter_AddRefs(mProxyRequest));  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to resolve protocol proxy: %d", __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }  return NS_OK;}

/** * Peforms intialization of the MANDATORY components of the Video Engine */MediaConduitErrorCode WebrtcVideoConduit::Init(){  CSFLogDebug(logTag,  "%s ", __FUNCTION__);  if( !(mVideoEngine = webrtc::VideoEngine::Create()) )  {    CSFLogError(logTag, "%s Unable to create video engine ", __FUNCTION__);     return kMediaConduitSessionNotInited;  }#if 0  // TRACING  mVideoEngine->SetTraceFilter(webrtc::kTraceAll);  mVideoEngine->SetTraceFile( "Vievideotrace.out" );#endif  if( !(mPtrViEBase = ViEBase::GetInterface(mVideoEngine)))  {    CSFLogError(logTag, "%s Unable to create video engine ", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if( !(mPtrViECapture = ViECapture::GetInterface(mVideoEngine)))  {    CSFLogError(logTag, "%s Unable to create video engine ", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if( !(mPtrViECodec = ViECodec::GetInterface(mVideoEngine)))  {    CSFLogError(logTag, "%s Unable to create video engine ", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if( !(mPtrViENetwork = ViENetwork::GetInterface(mVideoEngine)))  {    CSFLogError(logTag, "%s Unable to create video engine ", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if( !(mPtrViERender = ViERender::GetInterface(mVideoEngine)))  {    CSFLogError(logTag, "%s Unable to create video engine ", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  CSFLogDebug(logTag, "%sEngine Created: Init'ng the interfaces ",__FUNCTION__);  if(mPtrViEBase->Init() == -1)  {    CSFLogError(logTag, " %s Video Engine Init Failed %d ",__FUNCTION__,                                               mPtrViEBase->LastError());    return kMediaConduitSessionNotInited;  }  if(mPtrViEBase->CreateChannel(mChannel) == -1)  {    CSFLogError(logTag, " %s Channel creation Failed %d ",__FUNCTION__,                                               mPtrViEBase->LastError());    return kMediaConduitChannelError;  }  if(mPtrViENetwork->RegisterSendTransport(mChannel, *this) == -1)  {    CSFLogError(logTag,  "%s ViENetwork Failed %d ", __FUNCTION__,                                          mPtrViEBase->LastError());    return kMediaConduitTransportRegistrationFail;  }  mPtrExtCapture = 0;  if(mPtrViECapture->AllocateExternalCaptureDevice(mCapId,                                                   mPtrExtCapture) == -1)  {    CSFLogError(logTag, "%s Unable to Allocate capture module: %d ",                               __FUNCTION__, mPtrViEBase->LastError());    return kMediaConduitCaptureError;  }  if(mPtrViECapture->ConnectCaptureDevice(mCapId,mChannel) == -1)  {    CSFLogError(logTag, "%s Unable to Connect capture module: %d ",                               __FUNCTION__,mPtrViEBase->LastError());    return kMediaConduitCaptureError;  }  if(mPtrViERender->AddRenderer(mChannel,                                webrtc::kVideoI420,                                (webrtc::ExternalRenderer*) this) == -1)  {    CSFLogError(logTag, "%s Failed to added external renderer ", __FUNCTION__);    return kMediaConduitInvalidRenderer;  }//.........这里部分代码省略.........

/* * WebRTCAudioConduit Implementation */MediaConduitErrorCode WebrtcAudioConduit::Init(){  CSFLogDebug(logTag,  "%s this=%p", __FUNCTION__, this);#ifdef MOZ_WIDGET_ANDROID    jobject context = jsjni_GetGlobalContextRef();    // get the JVM    JavaVM *jvm = jsjni_GetVM();    JNIEnv* jenv = jsjni_GetJNIForThread();    if (webrtc::VoiceEngine::SetAndroidObjects(jvm, jenv, (void*)context) != 0) {      CSFLogError(logTag, "%s Unable to set Android objects", __FUNCTION__);      return kMediaConduitSessionNotInited;    }#endif  // Per WebRTC APIs below function calls return nullptr on failure  if(!(mVoiceEngine = webrtc::VoiceEngine::Create()))  {    CSFLogError(logTag, "%s Unable to create voice engine", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  EnableWebRtcLog();  if(!(mPtrVoEBase = VoEBase::GetInterface(mVoiceEngine)))  {    CSFLogError(logTag, "%s Unable to initialize VoEBase", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if(!(mPtrVoENetwork = VoENetwork::GetInterface(mVoiceEngine)))  {    CSFLogError(logTag, "%s Unable to initialize VoENetwork", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if(!(mPtrVoECodec = VoECodec::GetInterface(mVoiceEngine)))  {    CSFLogError(logTag, "%s Unable to initialize VoEBCodec", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if(!(mPtrVoEProcessing = VoEAudioProcessing::GetInterface(mVoiceEngine)))  {    CSFLogError(logTag, "%s Unable to initialize VoEProcessing", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if(!(mPtrVoEXmedia = VoEExternalMedia::GetInterface(mVoiceEngine)))  {    CSFLogError(logTag, "%s Unable to initialize VoEExternalMedia", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if(!(mPtrVoERTP_RTCP = VoERTP_RTCP::GetInterface(mVoiceEngine)))  {    CSFLogError(logTag, "%s Unable to initialize VoERTP_RTCP", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if(!(mPtrVoEVideoSync = VoEVideoSync::GetInterface(mVoiceEngine)))  {    CSFLogError(logTag, "%s Unable to initialize VoEVideoSync", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if (!(mPtrRTP = webrtc::VoERTP_RTCP::GetInterface(mVoiceEngine)))  {    CSFLogError(logTag, "%s Unable to get audio RTP/RTCP interface ",                __FUNCTION__);    return kMediaConduitSessionNotInited;  }  // init the engine with our audio device layer  if(mPtrVoEBase->Init() == -1)  {    CSFLogError(logTag, "%s VoiceEngine Base Not Initialized", __FUNCTION__);    return kMediaConduitSessionNotInited;  }  if( (mChannel = mPtrVoEBase->CreateChannel()) == -1)  {    CSFLogError(logTag, "%s VoiceEngine Channel creation failed",__FUNCTION__);    return kMediaConduitChannelError;  }  CSFLogDebug(logTag, "%s Channel Created %d ",__FUNCTION__, mChannel);  if(mPtrVoENetwork->RegisterExternalTransport(mChannel, *this) == -1)  {    CSFLogError(logTag, "%s VoiceEngine, External Transport Failed",__FUNCTION__);    return kMediaConduitTransportRegistrationFail;  }  if(mPtrVoEXmedia->SetExternalRecordingStatus(true) == -1)  {    CSFLogError(logTag, "%s SetExternalRecordingStatus Failed %d",__FUNCTION__,                mPtrVoEBase->LastError());//.........这里部分代码省略.........

/** * Note: Setting the send-codec on the Video Engine will restart the encoder, * sets up new SSRC and reset RTP_RTCP module with the new codec setting. * * Note: this is called from MainThread, and the codec settings are read on * videoframe delivery threads (i.e in SendVideoFrame().  With * renegotiation/reconfiguration, this now needs a lock!  Alternatively * changes could be queued until the next frame is delivered using an * Atomic pointer and swaps. */MediaConduitErrorCodeWebrtcVideoConduit::ConfigureSendMediaCodec(const VideoCodecConfig* codecConfig){  CSFLogDebug(logTag,  "%s for %s", __FUNCTION__, codecConfig ? codecConfig->mName.c_str() : "<null>");  bool codecFound = false;  MediaConduitErrorCode condError = kMediaConduitNoError;  int error = 0; //webrtc engine errors  webrtc::VideoCodec  video_codec;  std::string payloadName;  memset(&video_codec, 0, sizeof(video_codec));  {    //validate basic params    if((condError = ValidateCodecConfig(codecConfig,true)) != kMediaConduitNoError)    {      return condError;    }  }  condError = StopTransmitting();  if (condError != kMediaConduitNoError) {    return condError;  }  if (mExternalSendCodec &&      codecConfig->mType == mExternalSendCodec->mType) {    CSFLogError(logTag, "%s Configuring External H264 Send Codec", __FUNCTION__);    // width/height will be overridden on the first frame    video_codec.width = 320;    video_codec.height = 240;#ifdef MOZ_WEBRTC_OMX    if (codecConfig->mType == webrtc::kVideoCodecH264) {      video_codec.resolution_divisor = 16;    } else {      video_codec.resolution_divisor = 1; // We could try using it to handle odd resolutions    }#else    video_codec.resolution_divisor = 1; // We could try using it to handle odd resolutions#endif    video_codec.qpMax = 56;    video_codec.numberOfSimulcastStreams = 1;    video_codec.mode = webrtc::kRealtimeVideo;    codecFound = true;  } else {    // we should be good here to set the new codec.    for(int idx=0; idx < mPtrViECodec->NumberOfCodecs(); idx++)    {      if(0 == mPtrViECodec->GetCodec(idx, video_codec))      {        payloadName = video_codec.plName;        if(codecConfig->mName.compare(payloadName) == 0)        {          // Note: side-effect of this is that video_codec is filled in          // by GetCodec()          codecFound = true;          break;        }      }    }//for  }  if(codecFound == false)  {    CSFLogError(logTag, "%s Codec Mismatch ", __FUNCTION__);    return kMediaConduitInvalidSendCodec;  }  // Note: only for overriding parameters from GetCodec()!  CodecConfigToWebRTCCodec(codecConfig, video_codec);  if(mPtrViECodec->SetSendCodec(mChannel, video_codec) == -1)  {    error = mPtrViEBase->LastError();    if(error == kViECodecInvalidCodec)    {      CSFLogError(logTag, "%s Invalid Send Codec", __FUNCTION__);      return kMediaConduitInvalidSendCodec;    }    CSFLogError(logTag, "%s SetSendCodec Failed %d ", __FUNCTION__,                mPtrViEBase->LastError());    return kMediaConduitUnknownError;  }  if (!mVideoCodecStat) {    mVideoCodecStat = new VideoCodecStatistics(mChannel, mPtrViECodec);  }  mVideoCodecStat->Register(true);//.........这里部分代码省略.........

nsresultPeerConnectionMedia::UpdateTransportFlow(    size_t aLevel,    bool aIsRtcp,    const JsepTransport& aTransport){  if (aIsRtcp && aTransport.mComponents < 2) {    RemoveTransportFlow(aLevel, aIsRtcp);    return NS_OK;  }  if (!aIsRtcp && !aTransport.mComponents) {    RemoveTransportFlow(aLevel, aIsRtcp);    return NS_OK;  }  nsresult rv;  RefPtr<TransportFlow> flow = GetTransportFlow(aLevel, aIsRtcp);  if (flow) {    if (IsIceRestarting()) {      CSFLogInfo(LOGTAG, "Flow[%s]: detected ICE restart - level: %u rtcp: %d",                 flow->id().c_str(), (unsigned)aLevel, aIsRtcp);      RefPtr<PeerConnectionMedia> pcMedia(this);      rv = GetSTSThread()->Dispatch(          WrapRunnableNM(AddNewIceStreamForRestart_s,                         pcMedia, flow, aLevel, aIsRtcp),          NS_DISPATCH_NORMAL);      if (NS_FAILED(rv)) {        CSFLogError(LOGTAG, "Failed to dispatch AddNewIceStreamForRestart_s");        return rv;      }    }    return NS_OK;  }  std::ostringstream osId;  osId << mParentHandle << ":" << aLevel << "," << (aIsRtcp ? "rtcp" : "rtp");  flow = new TransportFlow(osId.str());  // The media streams are made on STS so we need to defer setup.  auto ice = MakeUnique<TransportLayerIce>();  auto dtls = MakeUnique<TransportLayerDtls>();  dtls->SetRole(aTransport.mDtls->GetRole() ==                        JsepDtlsTransport::kJsepDtlsClient                    ? TransportLayerDtls::CLIENT                    : TransportLayerDtls::SERVER);  RefPtr<DtlsIdentity> pcid = mParent->Identity();  if (!pcid) {    CSFLogError(LOGTAG, "Failed to get DTLS identity.");    return NS_ERROR_FAILURE;  }  dtls->SetIdentity(pcid);  const SdpFingerprintAttributeList& fingerprints =      aTransport.mDtls->GetFingerprints();  for (const auto& fingerprint : fingerprints.mFingerprints) {    std::ostringstream ss;    ss << fingerprint.hashFunc;    rv = dtls->SetVerificationDigest(ss.str(), &fingerprint.fingerprint[0],                                     fingerprint.fingerprint.size());    if (NS_FAILED(rv)) {      CSFLogError(LOGTAG, "Could not set fingerprint");      return rv;    }  }  std::vector<uint16_t> srtpCiphers;  srtpCiphers.push_back(SRTP_AES128_CM_HMAC_SHA1_80);  srtpCiphers.push_back(SRTP_AES128_CM_HMAC_SHA1_32);  rv = dtls->SetSrtpCiphers(srtpCiphers);  if (NS_FAILED(rv)) {    CSFLogError(LOGTAG, "Couldn't set SRTP ciphers");    return rv;  }  // Always permits negotiation of the confidential mode.  // Only allow non-confidential (which is an allowed default),  // if we aren't confidential.  std::set<std::string> alpn;  std::string alpnDefault = "";  alpn.insert("c-webrtc");  if (!mParent->PrivacyRequested()) {    alpnDefault = "webrtc";    alpn.insert(alpnDefault);  }  rv = dtls->SetAlpn(alpn, alpnDefault);  if (NS_FAILED(rv)) {    CSFLogError(LOGTAG, "Couldn't set ALPN");    return rv;  }  nsAutoPtr<PtrVector<TransportLayer> > layers(new PtrVector<TransportLayer>);  layers->values.push_back(ice.release());  layers->values.push_back(dtls.release());//.........这里部分代码省略.........

MediaConduitErrorCodeWebrtcVideoConduit::ConfigureRecvMediaCodecs(    const std::vector<VideoCodecConfig* >& codecConfigList){  CSFLogDebug(logTag,  "%s ", __FUNCTION__);  MediaConduitErrorCode condError = kMediaConduitNoError;  int error = 0; //webrtc engine errors  bool success = false;  std::string  payloadName;  // are we receiving already? If so, stop receiving and playout  // since we can't apply new recv codec when the engine is playing.  if(mEngineReceiving)  {    CSFLogDebug(logTag, "%s Engine Already Receiving . Attemping to Stop ", __FUNCTION__);    if(mPtrViEBase->StopReceive(mChannel) == -1)    {      error = mPtrViEBase->LastError();      if(error == kViEBaseUnknownError)      {        CSFLogDebug(logTag, "%s StopReceive() Success ", __FUNCTION__);        mEngineReceiving = false;      } else {        CSFLogError(logTag, "%s StopReceive() Failed %d ", __FUNCTION__,                    mPtrViEBase->LastError());        return kMediaConduitUnknownError;      }    }  }  mEngineReceiving = false;  if(codecConfigList.empty())  {    CSFLogError(logTag, "%s Zero number of codecs to configure", __FUNCTION__);    return kMediaConduitMalformedArgument;  }  webrtc::ViEKeyFrameRequestMethod kf_request = webrtc::kViEKeyFrameRequestNone;  bool use_nack_basic = false;  //Try Applying the codecs in the list  // we treat as success if atleast one codec was applied and reception was  // started successfully.  for(std::vector<VideoCodecConfig*>::size_type i=0;i < codecConfigList.size();i++)  {    //if the codec param is invalid or diplicate, return error    if((condError = ValidateCodecConfig(codecConfigList[i],false)) != kMediaConduitNoError)    {      return condError;    }    // Check for the keyframe request type: PLI is preferred    // over FIR, and FIR is preferred over none.    if (codecConfigList[i]->RtcpFbIsSet(SDP_RTCP_FB_NACK_PLI))    {      kf_request = webrtc::kViEKeyFrameRequestPliRtcp;    } else if(kf_request == webrtc::kViEKeyFrameRequestNone &&              codecConfigList[i]->RtcpFbIsSet(SDP_RTCP_FB_CCM_FIR))    {      kf_request = webrtc::kViEKeyFrameRequestFirRtcp;    }    // Check whether NACK is requested    if(codecConfigList[i]->RtcpFbIsSet(SDP_RTCP_FB_NACK_BASIC))    {      use_nack_basic = true;    }    webrtc::VideoCodec  video_codec;    mEngineReceiving = false;    memset(&video_codec, 0, sizeof(webrtc::VideoCodec));    //Retrieve pre-populated codec structure for our codec.    for(int idx=0; idx < mPtrViECodec->NumberOfCodecs(); idx++)    {      if(mPtrViECodec->GetCodec(idx, video_codec) == 0)      {        payloadName = video_codec.plName;        if(codecConfigList[i]->mName.compare(payloadName) == 0)        {          CodecConfigToWebRTCCodec(codecConfigList[i], video_codec);          if(mPtrViECodec->SetReceiveCodec(mChannel,video_codec) == -1)          {            CSFLogError(logTag, "%s Invalid Receive Codec %d ", __FUNCTION__,                        mPtrViEBase->LastError());          } else {            CSFLogError(logTag, "%s Successfully Set the codec %s", __FUNCTION__,                        codecConfigList[i]->mName.c_str());            if(CopyCodecToDB(codecConfigList[i]))            {              success = true;            } else {              CSFLogError(logTag,"%s Unable to updated Codec Database", __FUNCTION__);              return kMediaConduitUnknownError;            }          }          break; //we found a match        }      }//.........这里部分代码省略.........

nsresult PeerConnectionMedia::Init(const std::vector<NrIceStunServer>& stun_servers,                                   const std::vector<NrIceTurnServer>& turn_servers,                                   NrIceCtx::Policy policy){  nsresult rv;  nsCOMPtr<nsIProtocolProxyService> pps =    do_GetService(NS_PROTOCOLPROXYSERVICE_CONTRACTID, &rv);  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to get proxy service: %d", __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }  // We use the following URL to find the "default" proxy address for all HTTPS  // connections.  We will only attempt one HTTP(S) CONNECT per peer connection.  // "example.com" is guaranteed to be unallocated and should return the best default.  nsCOMPtr<nsIURI> fakeHttpsLocation;  rv = NS_NewURI(getter_AddRefs(fakeHttpsLocation), "https://example.com");  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to set URI: %d", __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }  nsCOMPtr<nsIScriptSecurityManager> secMan(      do_GetService(NS_SCRIPTSECURITYMANAGER_CONTRACTID, &rv));  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to get IOService: %d",        __FUNCTION__, (int)rv);    CSFLogError(logTag, "%s: Failed to get securityManager: %d", __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }  nsCOMPtr<nsIPrincipal> systemPrincipal;  rv = secMan->GetSystemPrincipal(getter_AddRefs(systemPrincipal));  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to get systemPrincipal: %d", __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }  nsCOMPtr<nsIChannel> channel;  rv = NS_NewChannel(getter_AddRefs(channel),                     fakeHttpsLocation,                     systemPrincipal,                     nsILoadInfo::SEC_ALLOW_CROSS_ORIGIN_DATA_IS_NULL,                     nsIContentPolicy::TYPE_OTHER);  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to get channel from URI: %d",                __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }  RefPtr<ProtocolProxyQueryHandler> handler = new ProtocolProxyQueryHandler(this);  rv = pps->AsyncResolve(channel,                         nsIProtocolProxyService::RESOLVE_PREFER_HTTPS_PROXY |                         nsIProtocolProxyService::RESOLVE_ALWAYS_TUNNEL,                         handler, getter_AddRefs(mProxyRequest));  if (NS_FAILED(rv)) {    CSFLogError(logTag, "%s: Failed to resolve protocol proxy: %d", __FUNCTION__, (int)rv);    return NS_ERROR_FAILURE;  }#if !defined(MOZILLA_EXTERNAL_LINKAGE)  bool ice_tcp = Preferences::GetBool("media.peerconnection.ice.tcp", false);  if (!XRE_IsParentProcess()) {    CSFLogError(logTag, "%s: ICE TCP not support on e10s", __FUNCTION__);    ice_tcp = false;  }  bool default_address_only = Preferences::GetBool(    "media.peerconnection.ice.default_address_only", false);#else  bool ice_tcp = false;  bool default_address_only = false;#endif  // TODO([email
C++ CSFML_CALL函数代码示例
C++ CRegString函数代码示例