[gst-devel] converting a gst-launch pipeline into source code problem

[Felipe Contreras]

On 10/18/07, Stefan Kost <ensonic at hora-obscura.de> wrote:
> Hi,
>
> Quoting Terry Leung <terry83 at gmail.com>:
>
> > Hi all,
> >
> > I got a problem when i convert a pipeline into source code
> > The pipeline is
> >
> > gst-launch -m ffmux_3gp name=mux ! filesink
> > location=/home/video/receive.3gp \
> > { udpsrc num-buffers=500 port=8100 name=audioudp
> > caps="application/x-rtp, media=(string)audio, payload=(int)98,
> > clock-rate=(int)8000, encoding-name=(string)AMR,
> > encoding-params=(string)1, octet-align=(string)1" ! rtpamrdepay
> > queue-delay=0 ! queue } ! mux.audio_0
> >
> You don't need the '{' that was gstreamer-0.8.
>
> >
> > The code is shown below:
> >
> >         m_pPipe = gst_pipeline_new (
> > (string("VideoPipe").append(sId)).c_str());
> >
>
>   m_pPipe = gst_pipeline_new ("VideoPipe");
> is just fine.
>
> >
> >         m_pMux = gst_element_factory_make ("ffmux_3gp",
> > (string("ffmux_3gp").append(sId)).c_str());
> >         m_pFileSink = gst_element_factory_make
> > ("filesink",(string("filesink").append(sId)).c_str() );
> >
> >         m_pAudioUdpSrc= gst_element_factory_make ("udpsrc",
> > (string("audioudpsrc").append(sId)).c_str());
> >         m_pAudioJitter= gst_element_factory_make
> > ("gstrtpjitterbuffer", (string("audiojitter").append(sId)).c_str());
> >         m_pAudioDepay= gst_element_factory_make ("rtpamrdepay",
> > (string("audiodepay").append(sId)).c_str());
> >         m_pAudioQueue= gst_element_factory_make ("queue",
> > (string("audioqueue").append(sId)).c_str());
> >
> >         gst_bin_add_many (GST_BIN (m_pPipe),
> >         m_pMux , m_pFileSink ,
> >         m_pAudioUdpSrc , m_pAudioJitter , m_pAudioDepay ,
> > m_pAudioQueue ,NULL);
> >
> > GstCaps* pAudioUdpSrcCaps = gst_caps_new_simple ("application/x-rtp",
> >         "media",G_TYPE_STRING,"audio",
> >         "payload",G_TYPE_INT,98,
> >         "clock-rate", G_TYPE_INT, 8000,
> >         "encoding-name", G_TYPE_STRING, "AMR",
> >         "encoding-params",G_TYPE_STRING,"1",
> >         "octet-align",G_TYPE_STRING,"1",
> >         NULL);
> >         nResult =
> > gst_element_link_filtered(m_pAudioUdpSrc,m_pAudioJitter,pAudioUdpSrcCaps);
> >         gst_caps_unref(pAudioUdpSrcCaps);
> >
> >         nResult = gst_element_link (m_pMux,m_pFileSink);
> >         nResult = nResult &&
> > gst_element_link_many(m_pAudioJitter,m_pAudioDepay,m_pAudioQueue,m_pMux,NULL);
> >
> >
> > I am not sure if the pipeline is the same as the one in gst-launch
> > I've tried my best to do the same thing as what it is in the gst-launch
> > (As i've also write another program to stream 3gp clips to a specific
> > ip and port, and i can do it without problem, i think there may be
> > some tricks for this case?)
> >
> The biggest difference is that demuxers create pads on the fly. So you
> would register a signal handler for "pad-added" of gstElement and then
> connect.

This keeps coming again and again.

Wouldn't it make sense to allow things like:

gst_element_link_pads (demuxer, "video_%02d", decoder, "sink");

-- 
Felipe Contreras