예제 #1
0
/***********************
* GetNextPacket
*	Obtiene el siguiente paquete para enviar
************************/
int H263Encoder1996::GetNextPacket(BYTE *out,DWORD &len)
{
	//Get info
	MediaFrame::RtpPacketizationInfo& info = frame->GetRtpPacketizationInfo();

	//Check
	if (num>info.size())
		//Exit
		return Error("No mor rtp packets!!");
	
	//Get packet
	MediaFrame::RtpPacketization* rtp = info[num++];

	//Make sure it is enought length
	if (rtp->GetTotalLength()>len)
	{
		//Nullify
		len = 0;
		//Send it
		return Error("H263 MB too big!!! [%d,%d]\n",rtp->GetTotalLength(),len);
	}
	
	//Copy prefic
	memcpy(out,rtp->GetPrefixData(),rtp->GetPrefixLen());
	//Copy data
	memcpy(out+rtp->GetPrefixLen(),frame->GetData()+rtp->GetPos(),rtp->GetSize());
	//Set len
	len = rtp->GetPrefixLen()+rtp->GetSize();

	//It is last??
	return (num<info.size());
}
예제 #2
0
int RTPMultiplexerSmoother::SmoothFrame(const MediaFrame* frame,DWORD duration)
{
	//Check
	if (!frame || !frame->HasRtpPacketizationInfo())
		//Error
		return Error("Frame do not has packetization info");

	//Get info
	const MediaFrame::RtpPacketizationInfo& info = frame->GetRtpPacketizationInfo();

	DWORD codec = 0;
	BYTE *frameData = NULL;
	DWORD frameSize = 0;

	//Depending on the type
	switch(frame->GetType())
	{
		case MediaFrame::Audio:
		{
			//get audio frame
			AudioFrame * audio = (AudioFrame*)frame;
			//Get codec
			codec = audio->GetCodec();
			//Get data
			frameData = audio->GetData();
			//Get size
			frameSize = audio->GetLength();
		}
			break;
		case MediaFrame::Video:
		{
			//get Video frame
			VideoFrame * video = (VideoFrame*)frame;
			//Get codec
			codec = video->GetCodec();
			//Get data
			frameData = video->GetData();
			//Get size
			frameSize = video->GetLength();
		}
			break;
		default:
			return Error("No smoother for frame");
	}

	DWORD frameLength = 0;
	//Calculate total length
	for (int i=0;i<info.size();i++)
		//Get total length
		frameLength += info[i]->GetTotalLength();

	//Calculate bitrate for frame
	DWORD current = 0;
	
	//For each one
	for (int i=0;i<info.size();i++)
	{
		//Get packet
		MediaFrame::RtpPacketization* rtp = info[i];

		//Create rtp packet
		RTPPacketSched *packet = new RTPPacketSched(frame->GetType(),codec);

		//Make sure it is enought length
		if (rtp->GetPrefixLen()+rtp->GetSize()>packet->GetMaxMediaLength())
			//Error
			continue;
		
		//Get pointer to media data
		BYTE* out = packet->GetMediaData();
		//Copy prefic
		memcpy(out,rtp->GetPrefixData(),rtp->GetPrefixLen());
		//Copy data
		memcpy(out+rtp->GetPrefixLen(),frameData+rtp->GetPos(),rtp->GetSize());
		//Set length
		DWORD len = rtp->GetPrefixLen()+rtp->GetSize();
		//Set length
		packet->SetMediaLength(len);
		switch(packet->GetMedia())
		{
			case MediaFrame::Video:
				//Set timestamp
				packet->SetTimestamp(frame->GetTimeStamp()*90);
				break;
			case MediaFrame::Audio:
				//Set timestamp
				packet->SetTimestamp(frame->GetTimeStamp()*8);
				break;
			default:
				//Set timestamp
				packet->SetTimestamp(frame->GetTimeStamp());
		}
		//Check
		if (i+1==info.size())
			//last
			packet->SetMark(true);
		else
			//No last
			packet->SetMark(false);
		//Calculate partial lenght
		current += len;
		//Calculate sending time offset from first frame
		packet->SetSendingTime(current*duration/frameLength);
		//Append it
		queue.Add(packet);
	}

	return 1;
}