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 Implementation
int 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;
    }
  }
}

void
RemoteSourceStreamInfo::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;
    }
  }
}

nsresult
PeerConnectionMedia::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.
 */
MediaConduitErrorCode
WebrtcVideoConduit::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.........

nsresult
PeerConnectionMedia::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.........

MediaConduitErrorCode
WebrtcVideoConduit::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 protected]): need some way to set not offerer later
  // Looks like a bug in the NrIceCtx API.
  mIceCtx = NrIceCtx::Create("PC:" + mParentName,
                             true, // Offerer
                             mParent->GetAllowIceLoopback(),
                             ice_tcp,
                             mParent->GetAllowIceLinkLocal(),
                             default_address_only,
                             policy);
  if(!mIceCtx) {
    CSFLogError(logTag, "%s: Failed to create Ice Context", __FUNCTION__);
    return NS_ERROR_FAILURE;
  }

  if (NS_FAILED(rv = mIceCtx->SetStunServers(stun_servers))) {
    CSFLogError(logTag, "%s: Failed to set stun servers", __FUNCTION__);
    return rv;
  }
  // Give us a way to globally turn off TURN support
#if !defined(MOZILLA_EXTERNAL_LINKAGE)
  bool disabled = Preferences::GetBool("media.peerconnection.turn.disable", false);
#else
  bool disabled = false;
#endif
  if (!disabled) {
//.........这里部分代码省略.........

nsresult PeerConnectionMedia::Init(const std::vector<NrIceStunServer>& stun_servers)
{
  // TODO([email protected]): need some way to set not offerer later
  // Looks like a bug in the NrIceCtx API.
  mIceCtx = NrIceCtx::Create("PC:" + mParent->GetHandle(), true);
  if(!mIceCtx) {
    CSFLogError(logTag, "%s: Failed to create Ice Context", __FUNCTION__);
    return NS_ERROR_FAILURE;
  }
  nsresult rv;
  if (NS_FAILED(rv = mIceCtx->SetStunServers(stun_servers))) {
    CSFLogError(logTag, "%s: Failed to set stun servers", __FUNCTION__);
    return rv;
  }
  if (NS_FAILED(rv = mDNSResolver->Init())) {
    CSFLogError(logTag, "%s: Failed to initialize dns resolver", __FUNCTION__);
    return rv;
  }
  if (NS_FAILED(rv = mIceCtx->SetResolver(mDNSResolver->AllocateResolver()))) {
    CSFLogError(logTag, "%s: Failed to get dns resolver", __FUNCTION__);
    return rv;
  }
  mIceCtx->SignalGatheringCompleted.connect(this,
                                            &PeerConnectionMedia::IceGatheringCompleted);
  mIceCtx->SignalCompleted.connect(this,
                                   &PeerConnectionMedia::IceCompleted);

  // Create three streams to start with.
  // One each for audio, video and DataChannel
  // TODO: this will be re-visited
  RefPtr<NrIceMediaStream> audioStream = mIceCtx->CreateStream("stream1", 2);
  RefPtr<NrIceMediaStream> videoStream = mIceCtx->CreateStream("stream2", 2);
  RefPtr<NrIceMediaStream> dcStream = mIceCtx->CreateStream("stream3", 2);

  if (!audioStream) {
    CSFLogError(logTag, "%s: audio stream is NULL", __FUNCTION__);
    return NS_ERROR_FAILURE;
  } else {
    mIceStreams.push_back(audioStream);
  }

  if (!videoStream) {
    CSFLogError(logTag, "%s: video stream is NULL", __FUNCTION__);
    return NS_ERROR_FAILURE;
  } else {
    mIceStreams.push_back(videoStream);
  }

  if (!dcStream) {
    CSFLogError(logTag, "%s: datachannel stream is NULL", __FUNCTION__);
    return NS_ERROR_FAILURE;
  } else {
    mIceStreams.push_back(dcStream);
  }

  // TODO([email protected]): This is not connected to the PCCimpl.
  // Will need to do that later.
  for (std::size_t i=0; i<mIceStreams.size(); i++) {
    mIceStreams[i]->SignalReady.connect(this, &PeerConnectionMedia::IceStreamReady);
  }

  // Start gathering
  nsresult res;
  mIceCtx->thread()->Dispatch(WrapRunnableRet(
    mIceCtx, &NrIceCtx::StartGathering, &res), NS_DISPATCH_SYNC
  );

  if (NS_FAILED(res)) {
    CSFLogError(logTag, "%s: StartGathering failed: %u",
      __FUNCTION__, static_cast<uint32_t>(res));
    return res;
  }

  return NS_OK;
}

RefPtr<SrtpFlow> SrtpFlow::Create(int cipher_suite,
                                           bool inbound,
                                           const void *key,
                                           size_t key_len) {
  nsresult res = Init();
  if (!NS_SUCCEEDED(res))
    return nullptr;

  RefPtr<SrtpFlow> flow = new SrtpFlow();

  if (!key) {
    CSFLogError(LOGTAG, "Null SRTP key specified");
    return nullptr;
  }

  if (key_len != SRTP_TOTAL_KEY_LENGTH) {
    CSFLogError(LOGTAG, "Invalid SRTP key length");
    return nullptr;
  }

  srtp_policy_t policy;
  memset(&policy, 0, sizeof(srtp_policy_t));

  // Note that we set the same cipher suite for RTP and RTCP
  // since any flow can only have one cipher suite with DTLS-SRTP
  switch (cipher_suite) {
    case SRTP_AES128_CM_HMAC_SHA1_80:
      CSFLogDebug(LOGTAG,
                  "Setting SRTP cipher suite SRTP_AES128_CM_HMAC_SHA1_80");
      srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
      srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
      break;
    case SRTP_AES128_CM_HMAC_SHA1_32:
      CSFLogDebug(LOGTAG,
                  "Setting SRTP cipher suite SRTP_AES128_CM_HMAC_SHA1_32");
      srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp);
      srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp); // 80-bit per RFC 5764
      break;                                                   // S 4.1.2.
    default:
      CSFLogError(LOGTAG, "Request to set unknown SRTP cipher suite");
      return nullptr;
  }
  // This key is copied into the srtp_t object, so we don't
  // need to keep it.
  policy.key = const_cast<unsigned char *>(
      static_cast<const unsigned char *>(key));
  policy.ssrc.type = inbound ? ssrc_any_inbound : ssrc_any_outbound;
  policy.ssrc.value = 0;
  policy.ekt = nullptr;
  policy.window_size = 1024;   // Use the Chrome value.  Needs to be revisited.  Default is 128
  policy.allow_repeat_tx = 1;  // Use Chrome value; needed for NACK mode to work
  policy.next = nullptr;

  // Now make the session
  srtp_err_status_t r = srtp_create(&flow->session_, &policy);
  if (r != srtp_err_status_ok) {
    CSFLogError(LOGTAG, "Error creating srtp session");
    return nullptr;
  }

  return flow;
}

MediaConduitErrorCode
WebrtcVideoConduit::SendVideoFrame(unsigned char* video_frame,
                                   unsigned int video_frame_length,
                                   unsigned short width,
                                   unsigned short height,
                                   VideoType video_type,
                                   uint64_t capture_time)
{

  CSFLogDebug(logTag,  "%s ", __FUNCTION__);

  //check for  the parameters sanity
  if(!video_frame || video_frame_length == 0 ||
     width == 0 || height == 0)
  {
    CSFLogError(logTag,  "%s Invalid Parameters ",__FUNCTION__);
    MOZ_ASSERT(PR_FALSE);
    return kMediaConduitMalformedArgument;
  }

  webrtc::RawVideoType type;
  switch (video_type) {
    case kVideoI420:
      type = webrtc::kVideoI420;
      break;
    case kVideoNV21:
      type = webrtc::kVideoNV21;
      break;
    default:
      CSFLogError(logTag,  "%s VideoType Invalid. Only 1420 and NV21 Supported",__FUNCTION__);
      MOZ_ASSERT(PR_FALSE);
      return kMediaConduitMalformedArgument;
  }
  //Transmission should be enabled before we insert any frames.
  if(!mEngineTransmitting)
  {
    CSFLogError(logTag, "%s Engine not transmitting ", __FUNCTION__);
    return kMediaConduitSessionNotInited;
  }

  // enforce even width/height (paranoia)
  MOZ_ASSERT(!(width & 1));
  MOZ_ASSERT(!(height & 1));

  if (!SelectSendResolution(width, height))
  {
    return kMediaConduitCaptureError;
  }

  //insert the frame to video engine in I420 format only
  if(mPtrExtCapture->IncomingFrame(video_frame,
                                   video_frame_length,
                                   width, height,
                                   type,
                                   (unsigned long long)capture_time) == -1)
  {
    CSFLogError(logTag,  "%s IncomingFrame Failed %d ", __FUNCTION__,
                                            mPtrViEBase->LastError());
    return kMediaConduitCaptureError;
  }

  CSFLogError(logTag, "%s Inserted A Frame", __FUNCTION__);
  return kMediaConduitNoError;
}