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C++ Packet::GetDataSize方法代码示例

本文整理汇总了C++中Packet::GetDataSize方法的典型用法代码示例。如果您正苦于以下问题:C++ Packet::GetDataSize方法的具体用法?C++ Packet::GetDataSize怎么用?C++ Packet::GetDataSize使用的例子?那么, 这里精选的方法代码示例或许可以为您提供帮助。您也可以进一步了解该方法所在Packet的用法示例。


在下文中一共展示了Packet::GetDataSize方法的7个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于系统推荐出更棒的C++代码示例。

示例1: lock

void Room::RoomImpl::BroadcastRoomInformation() {
    Packet packet;
    packet << static_cast<u8>(IdRoomInformation);
    packet << room_information.name;
    packet << room_information.member_slots;
    packet << room_information.uid;
    packet << room_information.port;
    packet << room_information.preferred_game;

    packet << static_cast<u32>(members.size());
    {
        std::lock_guard<std::mutex> lock(member_mutex);
        for (const auto& member : members) {
            packet << member.nickname;
            packet << member.mac_address;
            packet << member.game_info.name;
            packet << member.game_info.id;
        }
    }

    ENetPacket* enet_packet =
        enet_packet_create(packet.GetData(), packet.GetDataSize(), ENET_PACKET_FLAG_RELIABLE);
    enet_host_broadcast(server, 0, enet_packet);
    enet_host_flush(server);
}
开发者ID:namkazt,项目名称:citra,代码行数:25,代码来源:room.cpp

示例2: JoinAudioBuffers

void CAudioPin::JoinAudioBuffers(Packet* pBuffer, CDeMultiplexer* pDemuxer)
{
  if (pBuffer->pmt)
  {
    // Currently only uncompressed PCM audio is supported
    if (pBuffer->pmt->subtype == MEDIASUBTYPE_PCM)
    {
      //LogDebug("aud: Joinig Audio Buffers");
      WAVEFORMATEXTENSIBLE* wfe = (WAVEFORMATEXTENSIBLE*)pBuffer->pmt->pbFormat;
      WAVEFORMATEX* wf = (WAVEFORMATEX*)wfe;

      // Assuming all packets in the stream are the same size
      int packetSize = pBuffer->GetDataSize();

      int maxDurationInBytes = wf->nAvgBytesPerSec / 10; // max 100 ms buffer

      while (true)
      {
        if ((MAX_BUFFER_SIZE - pBuffer->GetDataSize() >= packetSize ) && 
            (maxDurationInBytes >= pBuffer->GetDataSize() + packetSize))
        {
          Packet* buf = pDemuxer->GetAudio(pBuffer->nPlaylist,pBuffer->nClipNumber);
          if (buf)
          {
            byte* data = buf->GetData();
            // Skip LPCM header when copying the next buffer
            pBuffer->SetCount(pBuffer->GetDataSize() + buf->GetDataSize() - LPCM_HEADER_SIZE);
            memcpy(pBuffer->GetData()+pBuffer->GetDataSize() - (buf->GetDataSize() - LPCM_HEADER_SIZE), &data[LPCM_HEADER_SIZE], buf->GetDataSize() - LPCM_HEADER_SIZE);
            delete buf;
          }
          else
          {
            // No new buffer was available in the demuxer
            break;
          }
        }
        else
        {
          // buffer limit reached
          break;
        }
      }
    }
  }
}
开发者ID:chefkoch,项目名称:MediaPortal-1,代码行数:45,代码来源:AudioPin.cpp

示例3:

void Room::RoomImpl::SendJoinSuccess(ENetPeer* client, MacAddress mac_address) {
    Packet packet;
    packet << static_cast<u8>(IdJoinSuccess);
    packet << mac_address;
    ENetPacket* enet_packet =
        enet_packet_create(packet.GetData(), packet.GetDataSize(), ENET_PACKET_FLAG_RELIABLE);
    enet_peer_send(client, 0, enet_packet);
    enet_host_flush(server);
}
开发者ID:namkazt,项目名称:citra,代码行数:9,代码来源:room.cpp

示例4: Send

////////////////////////////////////////////////////////////
/// Send a packet of data to the host (must be connected first)
////////////////////////////////////////////////////////////
Socket::Status SocketTCP::Send(Packet& PacketToSend)
{
    // Let the packet do custom stuff before sending it
    PacketToSend.OnSend();

    // First send the packet size
    Uint32 PacketSize = htonl(PacketToSend.GetDataSize());
    Send(reinterpret_cast<const char*>(&PacketSize), sizeof(PacketSize));

    // Send the packet data
    if (PacketSize > 0)
    {
        return Send(PacketToSend.GetData(), PacketToSend.GetDataSize());
    }
    else
    {
        return Socket::Done;
    }
}
开发者ID:fu7mu4,项目名称:entonetics,代码行数:22,代码来源:SocketTCP.cpp

示例5: lock

void Room::RoomImpl::SendModBanListResponse(ENetPeer* client) {
    Packet packet;
    packet << static_cast<u8>(IdModBanListResponse);
    {
        std::lock_guard lock(ban_list_mutex);
        packet << username_ban_list;
        packet << ip_ban_list;
    }

    ENetPacket* enet_packet =
        enet_packet_create(packet.GetData(), packet.GetDataSize(), ENET_PACKET_FLAG_RELIABLE);
    enet_peer_send(client, 0, enet_packet);
    enet_host_flush(server);
}
开发者ID:citra-emu,项目名称:citra,代码行数:14,代码来源:room.cpp

示例6: cAutoLock

// Get a packet from the beginning of the list
Packet *CPacketQueue::Get()
{
  CAutoLock cAutoLock(this);

  if (m_queue.size() == 0) {
    return NULL;
  }
  Packet *pPacket = m_queue.front();
  m_queue.pop_front();

  if (pPacket)
    m_dataSize -= pPacket->GetDataSize();

  return pPacket;
}
开发者ID:JERUKA9,项目名称:LAVFilters,代码行数:16,代码来源:PacketQueue.cpp

示例7: FillBuffer


//.........这里部分代码省略.........

                return ERROR_NO_DATA;
              }
            } // comparemediatypes
          }
        } // lock ends

        m_rtTitleDuration = buffer->rtTitleDuration;

        if (checkPlaybackState)
        {
          buffer->nNewSegment = 0;
          m_pCachedBuffer = buffer;

          CheckPlaybackState();

          LogDebug("vid: cached push  %6.3f clip: %d playlist: %d", m_pCachedBuffer->rtStart / 10000000.0, m_pCachedBuffer->nClipNumber, m_pCachedBuffer->nPlaylist);

          return ERROR_NO_DATA;
        }

        bool hasTimestamp = buffer->rtStart != Packet::INVALID_TIME;
        REFERENCE_TIME rtCorrectedStartTime = 0;
        REFERENCE_TIME rtCorrectedStopTime = 0;

        if (hasTimestamp)
        {
          if (m_bZeroTimeStream)
          {
            m_rtStreamTimeOffset = buffer->rtStart - buffer->rtClipStartTime;
            m_bZeroTimeStream = false;
          }

          if (m_bDiscontinuity || buffer->bDiscontinuity)
          {
            LogDebug("vid: set discontinuity");
            pSample->SetDiscontinuity(true);
            pSample->SetMediaType(buffer->pmt);
            m_bDiscontinuity = false;
          }

          rtCorrectedStartTime = buffer->rtStart - m_rtStreamTimeOffset;
          rtCorrectedStopTime = buffer->rtStop - m_rtStreamTimeOffset;

          pSample->SetTime(&rtCorrectedStartTime, &rtCorrectedStopTime);

          if (m_bInitDuration)
          {
            m_pFilter->SetTitleDuration(m_rtTitleDuration);
            m_pFilter->ResetPlaybackOffset(buffer->rtPlaylistTime - rtCorrectedStartTime);
            m_bInitDuration = false;
          }

          m_pFilter->OnPlaybackPositionChange();
        }
        else // Buffer has no timestamp
          pSample->SetTime(NULL, NULL);

        pSample->SetSyncPoint(buffer->bSyncPoint);

        {
          CAutoLock lock(&m_csDeliver);

          if (!m_bFlushing)
          {
            BYTE* pSampleBuffer;
            pSample->SetActualDataLength(buffer->GetDataSize());
            pSample->GetPointer(&pSampleBuffer);
            memcpy(pSampleBuffer, buffer->GetData(), buffer->GetDataSize());

            m_bFirstSample = false;

#ifdef LOG_VIDEO_PIN_SAMPLES
            LogDebug("vid: %6.3f corr %6.3f playlist time %6.3f clip: %d playlist: %d size: %d", buffer->rtStart / 10000000.0, rtCorrectedStartTime / 10000000.0, 
              buffer->rtPlaylistTime / 10000000.0, buffer->nClipNumber, buffer->nPlaylist, buffer->GetCount());
#endif
          }
          else
          {
            LogDebug("vid: dropped sample as flush is active!");
            return ERROR_NO_DATA;
          }
        }

        //static int iFrameNumber = 0;
        //LogMediaSample(pSample, iFrameNumber++);

        delete buffer;
      }
    } while (!buffer);
    return NOERROR;
  }

  catch(...)
  {
    LogDebug("vid: FillBuffer exception");
  }

  return S_OK;
}
开发者ID:BMOTech,项目名称:MediaPortal-1,代码行数:101,代码来源:VideoPin.cpp


注:本文中的Packet::GetDataSize方法示例由纯净天空整理自Github/MSDocs等开源代码及文档管理平台,相关代码片段筛选自各路编程大神贡献的开源项目,源码版权归原作者所有,传播和使用请参考对应项目的License;未经允许,请勿转载。