C++ ELOG_DEBUG函数代码示例

 void WebRtcConnection::writeSsrc(char* buf, int len, unsigned int ssrc) {
   ELOG_DEBUG("LEN %d", len);
   RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
   RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
   //if it is RTCP we check it it is a compound packet
   if (chead->isRtcp()) {      
     char* movingBuf = buf;
     int rtcpLength = 0;
     int totalLength = 0;
     do{
       movingBuf+=rtcpLength;
       RtcpHeader *chead= reinterpret_cast<RtcpHeader*>(movingBuf);
       rtcpLength= (ntohs(chead->length)+1)*4;      
       totalLength+= rtcpLength;
       ELOG_DEBUG("Is RTCP, prev SSRC %u, new %u, len %d ", chead->getSSRC(), ssrc, rtcpLength);
       chead->ssrc=htonl(ssrc);
       if (chead->packettype == RTCP_PS_Feedback_PT){
         FirHeader *thefir = reinterpret_cast<FirHeader*>(movingBuf);
         if (thefir->fmt == 4){ // It is a FIR Packet, we generate it
           this->sendPLI();
         }
       }
     } while(totalLength<len);
   } else {
     head->setSSRC(ssrc);
   }
 }

bool NiceConnection::setRemoteCandidates(
    std::vector<CandidateInfo> &candidates) {

    ELOG_DEBUG("Setting remote candidates %d", candidates.size());


    for (unsigned int compId = 1; compId <= iceComponents_; compId++) {

        GSList* candList = NULL;

        for (unsigned int it = 0; it < candidates.size(); it++) {
            NiceCandidateType nice_cand_type;
            CandidateInfo cinfo = candidates[it];
            if (cinfo.mediaType != this->mediaType
                    || this->transportName->compare(cinfo.transProtocol)
                    || cinfo.componentId != compId)
                continue;

            switch (cinfo.hostType) {
            case HOST:
                nice_cand_type = NICE_CANDIDATE_TYPE_HOST;
                break;
            case SRLFX:
                nice_cand_type = NICE_CANDIDATE_TYPE_SERVER_REFLEXIVE;
                break;
            case PRFLX:
                nice_cand_type = NICE_CANDIDATE_TYPE_PEER_REFLEXIVE;
                break;
            case RELAY:
                nice_cand_type = NICE_CANDIDATE_TYPE_RELAYED;
                break;
            default:
                nice_cand_type = NICE_CANDIDATE_TYPE_HOST;
                break;
            }

            NiceCandidate* thecandidate = nice_candidate_new(nice_cand_type);
            NiceAddress* naddr = nice_address_new();
            nice_address_set_from_string(naddr, cinfo.hostAddress.c_str());
            nice_address_set_port(naddr, cinfo.hostPort);
            thecandidate->addr = *naddr;
            sprintf(thecandidate->foundation, "%s", cinfo.foundation.c_str());

            thecandidate->username = strdup(cinfo.username.c_str());
            thecandidate->password = strdup(cinfo.password.c_str());
            thecandidate->stream_id = (guint) 1;
            thecandidate->component_id = cinfo.componentId;
            thecandidate->priority = cinfo.priority;
            thecandidate->transport = NICE_CANDIDATE_TRANSPORT_UDP;
            candList = g_slist_append(candList, thecandidate);
            ELOG_DEBUG("New Candidate SET %s %d", cinfo.hostAddress.c_str(), cinfo.hostPort);

        }
        nice_agent_set_remote_candidates(agent_, (guint) 1, compId, candList);
    }

    ELOG_DEBUG("Candidates SET");
    this->updateIceState(NICE_CANDIDATES_RECEIVED);
    return true;
}

  ExternalOutput::~ExternalOutput(){
    ELOG_DEBUG("Destructor");
    ELOG_DEBUG("Closing Sink");
    delete in;
    in = NULL;
    
    
    if (context_!=NULL){
      if (writeheadres_>=0)
        av_write_trailer(context_);
      if (avio_close>=0)
        avio_close(context_->pb);
      avformat_free_context(context_);
      context_=NULL;
    }
    if (videoCodec_!=NULL){
      avcodec_close(videoCodecCtx_);
      videoCodec_=NULL;
    }
    if (audioCodec_!=NULL){
      avcodec_close(audioCodecCtx_);
      audioCodec_ = NULL;    
    }

    sending_ = false;
    cond_.notify_one();
    thread_.join();
    /* boost::unique_lock<boost::mutex> lock(queueMutex_); */
    ELOG_DEBUG("ExternalOutput closed Successfully");
  }

  bool OutputProcessor::initAudioCoder() {

    aCoder = avcodec_find_encoder(static_cast<AVCodecID>(mediaInfo.audioCodec.codec));
    if (!aCoder) {
      ELOG_DEBUG("Encoder de audio no encontrado");
      return false;
    }

    aCoderContext = avcodec_alloc_context3(aCoder);
    if (!aCoderContext) {
      ELOG_DEBUG("Error de memoria en coder de audio");
      return false;
    }

    aCoderContext->sample_fmt = AV_SAMPLE_FMT_S16;
    aCoderContext->bit_rate = mediaInfo.audioCodec.bitRate;
    aCoderContext->sample_rate = mediaInfo.audioCodec.sampleRate;
    aCoderContext->channels = 1;

    if (avcodec_open2(aCoderContext, aCoder, NULL) < 0) {
      ELOG_DEBUG("Error al abrir el coder de audio");
      return false;
    }

    audioCoder = 1;
    return true;
  }

void RRGenerationHandler::notifyUpdate() {
  if (initialized_) {
    return;
  }

  auto pipeline = getContext()->getPipelineShared();
  if (!pipeline) {
    return;
  }

  connection_ = pipeline->getService<WebRtcConnection>().get();
  if (!connection_) {
    return;
  }

  uint32_t video_ssrc = connection_->getVideoSourceSSRC();
  if (video_ssrc != 0) {
    auto video_packets = std::make_shared<RRPackets>();
    video_packets->ssrc = video_ssrc;
    video_packets->type = VIDEO_PACKET;
    rr_info_map_[video_ssrc] = video_packets;
    ELOG_DEBUG("%s, message: Initialized video, ssrc: %u", connection_->toLog(), video_ssrc);
    initialized_ = true;
  }
  uint32_t audio_ssrc = connection_->getAudioSourceSSRC();
  if (audio_ssrc != 0) {
    auto audio_packets = std::make_shared<RRPackets>();
    audio_packets->ssrc = audio_ssrc;
    audio_packets->type = AUDIO_PACKET;
    rr_info_map_[audio_ssrc] = audio_packets;
    initialized_ = true;
    ELOG_DEBUG("%s, message: Initialized audio, ssrc: %u", connection_->toLog(), audio_ssrc);
  }
}

  int OutputProcessor::init(const MediaInfo& info, RTPDataReceiver* rtpReceiver) {
    this->mediaInfo = info;
    this->rtpReceiver_ = rtpReceiver;

    encodedBuffer_ = (unsigned char*) malloc(UNPACKAGED_BUFFER_SIZE);
    packagedBuffer_ = (unsigned char*) malloc(PACKAGED_BUFFER_SIZE);
    rtpBuffer_ = (unsigned char*) malloc(PACKAGED_BUFFER_SIZE);
    if(info.processorType == PACKAGE_ONLY){
      this->initVideoPackager();
      this->initAudioPackager();
      return 0;
    }
    if (mediaInfo.hasVideo) {
      this->mediaInfo.videoCodec.codec = VIDEO_CODEC_VP8;
      if (vCoder.initEncoder(mediaInfo.videoCodec)) {
        ELOG_DEBUG("Error initing encoder");
      }
      this->initVideoPackager();
    }
    if (mediaInfo.hasAudio) {

      ELOG_DEBUG("Init AUDIO processor");
      mediaInfo.audioCodec.codec = AUDIO_CODEC_PCM_U8;
      mediaInfo.audioCodec.sampleRate= 44100;
      mediaInfo.audioCodec.bitRate = 64000;
      encodedAudioBuffer_ = (unsigned char*) malloc(UNPACKAGED_BUFFER_SIZE);
      packagedAudioBuffer_ = (unsigned char*) malloc(UNPACKAGED_BUFFER_SIZE);
      this->initAudioCoder();
      this->initAudioPackager();

    }

    return 0;
  }

  void WebRtcConnection::onTransportData(char* buf, int len, Transport *transport) {
    boost::mutex::scoped_lock lock(writeMutex_);
    if (audioSink_ == NULL && videoSink_ == NULL && fbSink_==NULL)
      return;
    int length = len;
    rtcpheader *chead = reinterpret_cast<rtcpheader*> (buf);
    if (chead->packettype == RTCP_Receiver_PT || chead->packettype == RTCP_PS_Feedback_PT || chead->packettype == RTCP_RTP_Feedback_PT){
      if (fbSink_ != NULL) {
        fbSink_->deliverFeedback(buf,length);
      }
    } else {
      // RTP or RTCP Sender Report
      if (bundle_) {

        // Check incoming SSRC
        rtpheader *head = reinterpret_cast<rtpheader*> (buf);
        rtcpheader *chead = reinterpret_cast<rtcpheader*> (buf);
        unsigned int recvSSRC = ntohl(head->ssrc);

        if (chead->packettype == RTCP_Sender_PT) { //Sender Report
          ELOG_DEBUG ("RTP Sender Report %d length %d ", chead->packettype, ntohs(chead->length));
          recvSSRC = ntohl(chead->ssrc);
        }

        // Deliver data
        if (recvSSRC==this->getVideoSourceSSRC() || recvSSRC==this->getVideoSinkSSRC()) {
          videoSink_->deliverVideoData(buf, length);
        } else if (recvSSRC==this->getAudioSourceSSRC() || recvSSRC==this->getAudioSinkSSRC()) {
          audioSink_->deliverAudioData(buf, length);
        } else {
          ELOG_DEBUG("Unknown SSRC %u, localVideo %u, remoteVideo %u, ignoring", recvSSRC, this->getVideoSourceSSRC(), this->getVideoSinkSSRC());
        }
      } else if (transport->mediaType == AUDIO_TYPE) {
        if (audioSink_ != NULL) {
          rtpheader *head = (rtpheader*) buf;
          // Firefox does not send SSRC in SDP
          if (this->getAudioSourceSSRC() == 0) {
            ELOG_DEBUG("Audio Source SSRC is %u", ntohl(head->ssrc));
            this->setAudioSourceSSRC(ntohl(head->ssrc));
            this->updateState(TRANSPORT_READY, transport);
          }
          head->ssrc = htonl(this->getAudioSinkSSRC());
          audioSink_->deliverAudioData(buf, length);
        }
      } else if (transport->mediaType == VIDEO_TYPE) {
        if (videoSink_ != NULL) {
          rtpheader *head = (rtpheader*) buf;
          // Firefox does not send SSRC in SDP
          if (this->getVideoSourceSSRC() == 0) {
            ELOG_DEBUG("Video Source SSRC is %u", ntohl(head->ssrc));
            this->setVideoSourceSSRC(ntohl(head->ssrc));
            this->updateState(TRANSPORT_READY, transport);
          }

          head->ssrc = htonl(this->getVideoSinkSSRC());
          videoSink_->deliverVideoData(buf, length);
        }
      }
    }
  }

bool NicerConnection::setRemoteCandidates(const std::vector<CandidateInfo> &candidates, bool is_bundle) {
  std::vector<CandidateInfo> cands(candidates);
  auto remote_candidates_promise = std::make_shared<std::promise<void>>();
  nr_ice_peer_ctx *peer = peer_;
  nr_ice_media_stream *stream = stream_;
  std::shared_ptr<NicerInterface> nicer = nicer_;
  async([cands, is_bundle, nicer, peer, stream, this, remote_candidates_promise] {
    ELOG_DEBUG("%s message: adding remote candidates (%ld)", toLog(), cands.size());
    for (const CandidateInfo &cand : cands) {
      std::string sdp = cand.sdp;
      std::size_t pos = sdp.find(",");
      std::string candidate = sdp.substr(0, pos);
      ELOG_DEBUG("%s message: New remote ICE candidate (%s)", toLog(), candidate.c_str());
      UINT4 r = nicer->IcePeerContextParseTrickleCandidate(peer, stream, const_cast<char *>(candidate.c_str()));
      if (r && r != R_ALREADY) {
        ELOG_WARN("%s message: Couldn't add remote ICE candidate (%s) (%d)", toLog(), candidate.c_str(), r);
      }
    }
    remote_candidates_promise->set_value();
  });
  std::future<void> remote_candidates_future = remote_candidates_promise->get_future();
  std::future_status status = remote_candidates_future.wait_for(std::chrono::seconds(1));
  if (status == std::future_status::timeout) {
    ELOG_WARN("%s message: Could not set remote candidates", toLog());
    return false;
  }
  return true;
}

ExternalOutput::~ExternalOutput(){
    ELOG_DEBUG("Destructing");

    // Stop our thread so we can safely nuke libav stuff and close our
    // our file.
    recording_ = false;
    cond_.notify_one();
    thread_.join();

    if (audio_stream_ != NULL && video_stream_ != NULL && context_ != NULL){
        av_write_trailer(context_);
    }

    if (video_stream_ && video_stream_->codec != NULL){
        avcodec_close(video_stream_->codec);
    }

    if (audio_stream_ && audio_stream_->codec != NULL){
        avcodec_close(audio_stream_->codec);
    }

    if (context_ != NULL){
        avio_close(context_->pb);
        avformat_free_context(context_);
        context_ = NULL;
    }

    ELOG_DEBUG("Closed Successfully");
}

bool MediaStream::setRemoteSdp(std::shared_ptr<SdpInfo> sdp) {
  ELOG_DEBUG("%s message: setting remote SDP", toLog());
  remote_sdp_ = sdp;
  if (remote_sdp_->videoBandwidth != 0) {
    ELOG_DEBUG("%s message: Setting remote BW, maxVideoBW: %u", toLog(), remote_sdp_->videoBandwidth);
    this->rtcp_processor_->setMaxVideoBW(remote_sdp_->videoBandwidth*1000);
  }

  if (pipeline_initialized_) {
    pipeline_->notifyUpdate();
    return true;
  }


  bundle_ = remote_sdp_->isBundle;

  setVideoSourceSSRCList(remote_sdp_->video_ssrc_list);
  setAudioSourceSSRC(remote_sdp_->audio_ssrc);

  audio_enabled_ = remote_sdp_->hasAudio;
  video_enabled_ = remote_sdp_->hasVideo;

  rtcp_processor_->addSourceSsrc(getAudioSourceSSRC());
  std::for_each(video_source_ssrc_list_.begin(), video_source_ssrc_list_.end(), [this] (uint32_t new_ssrc){
      rtcp_processor_->addSourceSsrc(new_ssrc);
  });

  initializePipeline();

  return true;
}

bool ExternalOutput::initContext() {
    if (context_->oformat->video_codec != AV_CODEC_ID_NONE &&
            context_->oformat->audio_codec != AV_CODEC_ID_NONE &&
            video_stream_ == NULL &&
            audio_stream_ == NULL) {
        AVCodec* videoCodec = avcodec_find_encoder(context_->oformat->video_codec);
        ELOG_DEBUG("Found Video Codec %s", videoCodec->name);
        if (videoCodec==NULL) {
            ELOG_ERROR("Could not find video codec");
            return false;
        }
        video_stream_ = avformat_new_stream (context_, videoCodec);
        video_stream_->id = 0;
        video_stream_->codec->codec_id = context_->oformat->video_codec;
        video_stream_->codec->width = 640;
        video_stream_->codec->height = 480;
        video_stream_->codec->time_base = (AVRational) {
            1,30
        };   // A decent guess here suffices; if processing the file with ffmpeg,
        // use -vsync 0 to force it not to duplicate frames.
        video_stream_->codec->pix_fmt = PIX_FMT_YUV420P;
        if (context_->oformat->flags & AVFMT_GLOBALHEADER) {
            video_stream_->codec->flags|=CODEC_FLAG_GLOBAL_HEADER;
        }
        context_->oformat->flags |= AVFMT_VARIABLE_FPS;

        AVCodec* audioCodec = avcodec_find_encoder(context_->oformat->audio_codec);
        if (audioCodec==NULL) {
            ELOG_ERROR("Could not find audio codec");
            return false;
        }
        ELOG_DEBUG("Found Audio Codec %s", audioCodec->name);
        audio_stream_ = avformat_new_stream (context_, audioCodec);
        audio_stream_->id = 1;
        audio_stream_->codec->codec_id = context_->oformat->audio_codec;
        audio_stream_->codec->sample_rate = context_->oformat->audio_codec == AV_CODEC_ID_PCM_MULAW ? 8000 : 48000; // TODO is it always 48 khz for opus?
        audio_stream_->codec->time_base = (AVRational) {
            1, audio_stream_->codec->sample_rate
        };
        audio_stream_->codec->channels = context_->oformat->audio_codec == AV_CODEC_ID_PCM_MULAW ? 1 : 2;   // TODO is it always two channels for opus?
        if (context_->oformat->flags & AVFMT_GLOBALHEADER) {
            audio_stream_->codec->flags|=CODEC_FLAG_GLOBAL_HEADER;
        }

        context_->streams[0] = video_stream_;
        context_->streams[1] = audio_stream_;
        if (avio_open(&context_->pb, context_->filename, AVIO_FLAG_WRITE) < 0) {
            ELOG_ERROR("Error opening output file");
            return false;
        }

        if (avformat_write_header(context_, NULL) < 0) {
            ELOG_ERROR("Error writing header");
            return false;
        }
        ELOG_DEBUG("avformat configured");
    }

    return true;
}

void DtlsTransport::onHandshakeCompleted(DtlsSocketContext* ctx, std::string clientKey, std::string serverKey, std::string srtp_profile)
{
    boost::mutex::scoped_lock lock(sessionMutex_);
    if (ctx == dtlsRtp_.get()) {
        ELOG_DEBUG("%s - Setting RTP srtp params", transport_name.c_str());
        srtp_.reset(new SrtpChannel());
        if (srtp_->setRtpParams((char*)clientKey.c_str(), (char*)serverKey.c_str())) {
            readyRtp_ = true;
        } else {
            updateTransportState(TRANSPORT_FAILED);
        }
        if (dtlsRtcp_ == NULL) {
            readyRtcp_ = true;
        }
    }
    if (ctx == dtlsRtcp_.get()) {
        ELOG_DEBUG("%s - Setting RTCP srtp params", transport_name.c_str());
        srtcp_.reset(new SrtpChannel());
        if (srtcp_->setRtpParams((char*)clientKey.c_str(), (char*)serverKey.c_str())) {
            readyRtcp_ = true;
        } else {
            updateTransportState(TRANSPORT_FAILED);
        }
    }
    ELOG_DEBUG("%s - Ready? %d %d", transport_name.c_str(), readyRtp_, readyRtcp_);
    if (readyRtp_ && readyRtcp_) {
        ELOG_DEBUG("%s - Ready!!!", transport_name.c_str());
        updateTransportState(TRANSPORT_READY);
    }
}

  int VideoDecoder::initDecoder (const VideoCodecInfo& info){
    ELOG_DEBUG("Init Decoder");
    vDecoder = avcodec_find_decoder(VideoCodecID2ffmpegDecoderID(info.codec));
    if (!vDecoder) {
      ELOG_DEBUG("Error getting video decoder");
      return -1;
    }

    vDecoderContext = avcodec_alloc_context3(vDecoder);
    if (!vDecoderContext) {
      ELOG_DEBUG("Error getting allocating decoder context");
      return -1;
    }

    vDecoderContext->width = info.width;
    vDecoderContext->height = info.height;

    if (avcodec_open2(vDecoderContext, vDecoder, NULL) < 0) {
      ELOG_DEBUG("Error opening video decoder");
      return -1;
    }

    dPicture = av_frame_alloc();
    if (!dPicture) {
      ELOG_DEBUG("Error allocating video frame");
      return -1;
    }

    return 0;
  }

  // memory is only valid for duration of callback; must be copied if queueing
  // is required
  DtlsSocketContext::DtlsSocketContext() {
      started = false;
      mSocket = NULL;
      receiver = NULL;
      DtlsSocketContext::Init();

      ELOG_DEBUG("Creating Dtls factory, Openssl v %s", OPENSSL_VERSION_TEXT);

      mContext = SSL_CTX_new(DTLSv1_method());
      assert(mContext);

      int r = SSL_CTX_use_certificate(mContext, mCert);
      assert(r == 1);

      r = SSL_CTX_use_PrivateKey(mContext, privkey);
      assert(r == 1);

      SSL_CTX_set_cipher_list(mContext, "ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH");

      SSL_CTX_set_info_callback(mContext, SSLInfoCallback);

      SSL_CTX_set_verify(mContext, SSL_VERIFY_PEER |SSL_VERIFY_FAIL_IF_NO_PEER_CERT,
        SSLVerifyCallback);

      // SSL_CTX_set_session_cache_mode(mContext, SSL_SESS_CACHE_OFF);
      // SSL_CTX_set_options(mContext, SSL_OP_NO_TICKET);
      // Set SRTP profiles
      r = SSL_CTX_set_tlsext_use_srtp(mContext, DefaultSrtpProfile);
      assert(r == 0);

      SSL_CTX_set_verify_depth(mContext, 2);
      SSL_CTX_set_read_ahead(mContext, 1);

      ELOG_DEBUG("DtlsSocketContext %p created", this);
    }

int AudioEncoder::initEncoder(const AudioCodecInfo& mediaInfo) {
  ELOG_DEBUG("Init audioEncoder begin");
  aCoder_ = avcodec_find_encoder(AudioCodecID2ffmpegDecoderID(mediaInfo.codec));
  if (!aCoder_) {
    ELOG_DEBUG("Audio Codec not found");
    return false;
  }

  aCoderContext_ = avcodec_alloc_context3(aCoder_);
  if (!aCoderContext_) {
    ELOG_DEBUG("Memory error allocating audio coder context");
    return false;
  }

  aCoderContext_->sample_fmt = AV_SAMPLE_FMT_FLT;
  // aCoderContext_->bit_rate = mediaInfo.bitRate;
  aCoderContext_->sample_rate = 8 /*mediaInfo.sampleRate*/;
  aCoderContext_->channels = 1;
  char errbuff[500];
  int res = avcodec_open2(aCoderContext_, aCoder_, NULL);
  if (res != 0) {
    av_strerror(res, reinterpret_cast<char*>(&errbuff), 500);
    ELOG_DEBUG("fail when opening input %s", errbuff);
    return -1;
  }
  ELOG_DEBUG("Init audioEncoder end");
  return true;
}