Yes，
It's normally to get merged tcp pockets. I use tcpserversrc to
receive RTP stream, and add a mark 0x000001 at RTP's header.
So I extract RTP pockets with the mark from tcp stream.
在2010-05-17 19:10:00，[hidden email] 写道：
>Send gstreamer-devel mailing list submissions to
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>or, via email, send a message with subject or body 'help' to
> [hidden email]
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>When replying, please edit your Subject line so it is more specific
>than "Re: Contents of gstreamer-devel digest..."
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>
>Today's Topics:
>
> 1. Re: tcpserversrc restarting (Antoine Martin) (Antoine Martin)
> 2. Re: RTSP Seek and DVB subtitles (Marc Leeman)
> 3. Caps problem when writing encoder/decoder (Hunterwood)
>
>
>----------------------------------------------------------------------
>
>Message: 1
>Date: Mon, 17 May 2010 11:25:14 +0700
>From: Antoine Martin <[hidden email]>
>Subject: Re: [gst-devel] tcpserversrc restarting (Antoine Martin)
>To: Discussion of the development of GStreamer
> <[hidden email]>
>Message-ID: <[hidden email]>
>Content-Type: text/plain; charset=UTF-8
>
>libing195 wrote:
>> I have improved the tcpserversrc to a retentive service to receive tcp data continually.
>Thanks for that, the only difficulty for me is that my current code is
>pure python, and I dont fancy shipping my own gst modules for all the
>platforms I want to support..
>
>Have you looked at getting this merged upstream?
>
>Cheers
>Antoine
>
>
>>
>> this is my code:
>>
>> struct _GstTCPServerSrc {
>> GstPushSrc element;
>>
>> //../Add some gboolean vars
>> gboolean recvError; default = FALSE;
>> gboolean bReconnected; default=TRUE;
>> }
>>
>> Then in gst_tcp_server_src_create():
>>
>> static GstFlowReturn
>> gst_tcp_server_src_create(GstPushSrc * psrc,
>> GstBuffer ** outbuf) {
>> GstTCPServerSrc *src;
>> GstFlowReturn ret = GST_FLOW_OK;
>> /* static gint indexPocket = 0;
>> int i = 0;*/
>>
>> src = GST_TCP_SERVER_SRC (psrc);
>>
>> if (!GST_OBJECT_FLAG_IS_SET (src, GST_TCP_SERVER_SRC_OPEN))
>> goto wrong_state;
>>
>> restart:
>>
>> //Libing begin
>> if (src->client_sock_fd.fd >= 0) {
>> if(src->recvError){
>> // g_print("**********Client connection has closed fd=%d clientnum=%d**************\n",
>> // src->client_sock_fd.fd, src->clientnum);
>>
>> gst_poll_remove_fd(src->fdset, &src->client_sock_fd);
>> gst_poll_fd_ctl_read(src->fdset, &src->server_sock_fd, TRUE);
>> src->client_sock_fd.fd = -1;
>> src->bReconnected = TRUE;
>> }else{
>> /* if we have a client, wait for read */
>> gst_poll_fd_ctl_read(src->fdset, &src->server_sock_fd, FALSE);
>> gst_poll_fd_ctl_read(src->fdset, &src->client_sock_fd, TRUE);
>> }
>>
>> } else {
>> /* else wait on server socket for connections */
>> gst_poll_fd_ctl_read(src->fdset, &src->server_sock_fd, TRUE);
>> }
>> //Libing end
>>
>> // g_print("Poll will wait!!!!~~~~~~\n");
>>
>> /* no action (0) is an error too in our case */
>> if ((ret = gst_poll_wait(src->fdset, GST_CLOCK_TIME_NONE)) <= 0) {
>> if (ret == -1 && errno == EBUSY)
>> goto select_cancelled;
>> else
>> goto select_error;
>> }
>>
>> /* if we have no client socket we can accept one now */
>> if (src->client_sock_fd.fd < 0) {
>> if (gst_poll_fd_can_read(src->fdset, &src->server_sock_fd)) {
>> if ((src->client_sock_fd.fd = accept(src->server_sock_fd.fd,
>> (struct sockaddr *) &src->client_sin,
>> &src->client_sin_len)) == -1)
>> goto accept_error;
>>
>> gst_poll_add_fd(src->fdset, &src->client_sock_fd);
>> src->recvError = FALSE;
>>
>> /* g_print("**********Client has connected fd=%d clientnum=%d**************\n",
>> src->client_sock_fd.fd, src->clientnum);*/
>> }
>> /* and restart now to poll the socket. */
>> goto restart;
>> }
>>
>> GST_LOG_OBJECT(src, "asked for a buffer");
>>
>> switch (src->protocol) {
>> case GST_TCP_PROTOCOL_NONE:{
>> if((ret = gst_tcp_server_src_read_buffer(src, src->client_sock_fd.fd,
>> src->fdset, outbuf)) == GST_FLOW_OK){
>> return ret;
>> }else{
>> g_print("____________Receive Error!\n");
>> close(src->client_sock_fd.fd);
>> src->recvError = TRUE;
>> goto restart;
>> }
>> break;
>> }
>> case GST_TCP_PROTOCOL_GDP:{
>> if (!src->caps_received) {
>> GstCaps *caps;
>> gchar *string;
>>
>> ret = gst_tcp_gdp_read_caps(GST_ELEMENT (src),
>> src->client_sock_fd.fd, src->fdset, &caps);
>>
>> if (ret == GST_FLOW_WRONG_STATE)
>> goto gdp_cancelled;
>>
>> if (ret != GST_FLOW_OK)
>> goto gdp_caps_read_error;
>>
>> src->caps_received = TRUE;
>> string = gst_caps_to_string(caps);
>> GST_DEBUG_OBJECT(src, "Received caps through GDP: %s", string);
>> g_free(string);
>>
>> gst_pad_set_caps(GST_BASE_SRC_PAD (psrc), caps);
>> }
>>
>> ret = gst_tcp_gdp_read_buffer(GST_ELEMENT (src),
>> src->client_sock_fd.fd, src->fdset, outbuf);
>>
>> if (ret == GST_FLOW_OK)
>> gst_buffer_set_caps(*outbuf, GST_PAD_CAPS (GST_BASE_SRC_PAD (src)));
>>
>> break;
>> }
>> default:
>> /* need to assert as buf == NULL */
>> g_assert("Unhandled protocol type");
>> break;
>> }
>>
>> if (ret == GST_FLOW_OK) {
>> GST_LOG_OBJECT (src,
>> "Returning buffer from _get of size %d, ts %"
>> GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT
>> ", offset %" G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
>> GST_BUFFER_SIZE (*outbuf),
>> GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (*outbuf)),
>> GST_TIME_ARGS (GST_BUFFER_DURATION (*outbuf)),
>> GST_BUFFER_OFFSET (*outbuf), GST_BUFFER_OFFSET_END (*outbuf));
>> }
>>
>> return ret;
>>
>> wrong_state:
>> {
>> GST_DEBUG_OBJECT (src, "connection to closed, cannot read data");
>> return GST_FLOW_WRONG_STATE;
>> }
>> select_error:
>> {
>> GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
>> ("Select error: %s", g_strerror (errno)));
>> return GST_FLOW_ERROR;
>> }
>> select_cancelled:
>> {
>> GST_DEBUG_OBJECT (src, "select canceled");
>> return GST_FLOW_WRONG_STATE;
>> }
>> accept_error:
>> {
>> GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
>> ("Could not accept client on server socket: %s", g_strerror (errno)));
>> return GST_FLOW_ERROR;
>> }
>> gdp_cancelled:
>> {
>> GST_DEBUG_OBJECT (src, "reading gdp canceled");
>> return GST_FLOW_WRONG_STATE;
>> }
>> gdp_caps_read_error:
>> {
>> /* if we did not get canceled, report an error */
>> if (ret != GST_FLOW_WRONG_STATE) {
>> GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
>> ("Could not read caps through GDP"));
>> }
>> return ret;
>> }
>> }
>>
>>
>> GstFlowReturn
>> gst_tcp_server_src_read_buffer (GstTCPServerSrc* src, int socket,
>> GstPoll * fdset, GstBuffer ** buf)
>>
>>
>>
>> --
>>
>> Bercy Li
>> +8615954811012
>> [hidden email]
>>
>>
>>
>>
>> ?2010-05-15 14:44:11?[hidden email] ???
>>> Send gstreamer-devel mailing list submissions to
>>> [hidden email]
>>>
>>> To subscribe or unsubscribe via the World Wide Web, visit
>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>> or, via email, send a message with subject or body 'help' to
>>> [hidden email]
>>>
>>> You can reach the person managing the list at
>>> [hidden email]
>>>
>>> When replying, please edit your Subject line so it is more specific
>>> than "Re: Contents of gstreamer-devel digest..."
>>>
>>>
>>> Today's Topics:
>>>
>>> 1. tcpserversrc restarting (Antoine Martin)
>>> 2. Re: GStreamer Conference 2010 (wl2776)
>>> 3. Re: No EOS message at the end of file. (wl2776)
>>> 4. Re: RTSP Seek and DVB subtitles (Alexander Olekhnovich)
>>> 5. Framestepping backwards in MPEG2 files (wl2776)
>>> 6. Pre-releases! gst-plugins-good 0.10.22.2, -ugly 0.10.14.2,
>>> -bad 0.10.18.2 (Tim-Philipp M?ller)
>>> 7. Memory profiling and hunting memory leaks (Loc Nguyen)
>>> 8. Re: Black-and-white output (Marco Ballesio)
>>>
>>>
>>> ----------------------------------------------------------------------
>>>
>>> Message: 1
>>> Date: Fri, 14 May 2010 18:20:28 +0700
>>> From: Antoine Martin <[hidden email]>
>>> Subject: [gst-devel] tcpserversrc restarting
>>> To: "[hidden email]"
>>> <[hidden email]>
>>> Message-ID: <[hidden email]>
>>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>>>
>>> Hi,
>>>
>>> I have some code largely based on this tcpserversrc example:
>>> http://www.jejik.com/articles/2007/01/streaming_audio_over_tcp_with_python-gstreamer/
>>>
>>> I want to ensure that it keeps working after the first client
>>> disconnects, which is not the case by default.
>>> At the moment, the tcp socket shows as "listening" but any further data
>>> sent will not be processed.
>>>
>>> I've tried unlink()ing the decoder when receiving EOS and re-adding a
>>> new one, but it complained that that there was an existing one with the
>>> same name. How do I free the resources to start again? How about even
>>> freeing the socket?
>>>
>>> I would rather not switch to using RTP... But if I have to I will, how
>>> does gstrtpbin deal with multiple clients connecting?
>>>
>>> Thanks
>>> Antoine
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> Message: 2
>>> Date: Fri, 14 May 2010 04:38:05 -0700 (PDT)
>>> From: wl2776 <[hidden email]>
>>> Subject: Re: [gst-devel] GStreamer Conference 2010
>>> To: [hidden email]
>>> Message-ID: <[hidden email]>
>>> Content-Type: text/plain; charset=us-ascii
>>>
>>>
>>> Great to hear this.
>>>
>>> What's the registration fee?
>>> --
>>> View this message in context: http://gstreamer-devel.966125.n4.nabble.com/GStreamer-Conference-2010-tp2123970p2216448.html
>>> Sent from the GStreamer-devel mailing list archive at Nabble.com.
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> Message: 3
>>> Date: Fri, 14 May 2010 05:34:17 -0700 (PDT)
>>> From: wl2776 <[hidden email]>
>>> Subject: Re: [gst-devel] No EOS message at the end of file.
>>> To: [hidden email]
>>> Message-ID: <[hidden email]>
>>> Content-Type: text/plain; charset=us-ascii
>>>
>>>
>>>
>>> Wim Taymans wrote:
>>>> On Fri, 2010-05-14 at 11:42 +0100, Giles Atkinson wrote:
>>>>> > My problem is that the EOS message doesn't appear on playbin2's bus,
>>>>> after
>>>>> > some manipulations with reverse and forward playback. But the playbin2
>>>>> is
>>>>> > surely at the end of file, as reported by _query_position().
>>>> EOS should always be sent in the PLAYING state when the pipeline is EOS,
>>>> if not, please file a bug with an example or a way to reproduce the
>>>> strange behaviour that you are seeing.
>>>>
>>> Hmm... The problem is gone now. After rebuild.
>>> --
>>> View this message in context: http://gstreamer-devel.966125.n4.nabble.com/No-EOS-message-at-the-end-of-file-tp2216227p2216515.html
>>> Sent from the GStreamer-devel mailing list archive at Nabble.com.
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> Message: 4
>>> Date: Fri, 14 May 2010 15:23:16 +0300
>>> From: Alexander Olekhnovich <[hidden email]>
>>> Subject: Re: [gst-devel] RTSP Seek and DVB subtitles
>>> To: Marc Leeman <[hidden email]>, Discussion of the
>>> development of GStreamer <[hidden email]>
>>> Message-ID:
>>> <[hidden email]>
>>> Content-Type: text/plain; charset="iso-8859-1"
>>>
>>> Hi Marc,
>>>
>>> I think ppl are interested :) At least I would really like to have a look at
>>> that.
>>>
>>> On Thu, May 13, 2010 at 7:24 PM, Marc Leeman <[hidden email]> wrote:
>>>
>>>>> Regarding dvb subtitles, there is a little bit of work going on. One
>>>>> with gst-teletext to grab subtitles from teletext and another with
>>>>> image subtitles.
>>>> I've got a working implementation that I am willing to share if ppl are
>>>> interested.
>>>>
>>>> You can use pango to get an approximate of a full TT page or just get
>>>> the subs (page) in text format.
>>>>
>>>> Tested on a number of DVB-S streams with good result.
>>>>
>>>> --
>>>> greetz, marc
>>>> After an instrument has been assembled, extra components will be found
>>>> on the bench.
>>>> crichton 2.6.26 #1 PREEMPT Tue Jul 29 21:17:59 CDT 2008 GNU/Linux
>>>>
>>>> -----BEGIN PGP SIGNATURE-----
>>>> Version: GnuPG v1.4.6 (GNU/Linux)
>>>>
>>>> iD8DBQFL7CfOUQpj09NWLeERAoeFAKDPbRQw/hGwVPZXD7/ll3NZdjEp8QCZAVsN
>>>> 7CCwL8z16lUE6OgsbktpAWU=
>>>> =lYBv
>>>> -----END PGP SIGNATURE-----
>>>>
>>>>
>>>> ------------------------------------------------------------------------------
>>>>
>>>>
>>>> _______________________________________________
>>>> gstreamer-devel mailing list
>>>> [hidden email]
>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>>>
>>>>
>>>
>>> --
>>> Thank you,
>>> Alexander Olekhnovich
>>> -------------- next part --------------
>>> An HTML attachment was scrubbed...
>>>
>>> ------------------------------
>>>
>>> Message: 5
>>> Date: Fri, 14 May 2010 08:03:24 -0700 (PDT)
>>> From: wl2776 <[hidden email]>
>>> Subject: [gst-devel] Framestepping backwards in MPEG2 files
>>> To: [hidden email]
>>> Message-ID: <[hidden email]>
>>> Content-Type: text/plain; charset=us-ascii
>>>
>>>
>>> Is it complete?
>>> My pipeline doesn't step one frame backwards.
>>>
>>> m_player is an instance of the playbin2. It has loaded the MPEG2 Program
>>> Stream.
>>> Then, the pipeline was paused before calling step_left
>>>
>>> void gst_player::step_left(void)
>>> {GstFormat fmt=GST_FORMAT_TIME;
>>>
>>> if(m_player){
>>> if(m_current_position>=40*GST_MSECOND){
>>> gboolean rb=gst_element_seek(m_player,-1.0,GST_FORMAT_TIME,
>>>
>>> (GstSeekFlags)(GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE),
>>>
>>> GST_SEEK_TYPE_SET,m_current_position,GST_SEEK_TYPE_SET,m_stream_duration);
>>> GST_DEBUG("seek backwards: %d",rb);
>>> GstEvent *event = gst_event_new_step (GST_FORMAT_BUFFERS, 1, 1.0,
>>> TRUE, FALSE);
>>> rb=gst_element_send_event (m_player, event);
>>> GST_DEBUG("send step event: %d",rb);
>>> }
>>> }
>>> }
>>>
>>>
>>> Both _seek and _send_event(_new_step) return TRUE.
>>> However, I don't see any changes in picture on the screen.
>>> After several calls to step_left() I get the EOS.
>>> What am I doing wrong?
>>> --
>>> View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Framestepping-backwards-in-MPEG2-files-tp2216718p2216718.html
>>> Sent from the GStreamer-devel mailing list archive at Nabble.com.
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> Message: 6
>>> Date: Fri, 14 May 2010 20:49:47 +0100
>>> From: Tim-Philipp M?ller <[hidden email]>
>>> Subject: [gst-devel] Pre-releases! gst-plugins-good 0.10.22.2, -ugly
>>> 0.10.14.2, -bad 0.10.18.2
>>> To: [hidden email]
>>> Message-ID: <1273866587.26703.6.camel@zingle>
>>> Content-Type: text/plain; charset="UTF-8"
>>>
>>> Hi,
>>>
>>> Below the latest pre-releases for gst-plugins-good, -ugly and -bad.
>>>
>>> Please test them thoroughly and file blocker bugs for all regressions or
>>> other major issues you find at http://gstreamer.freedesktop.org/bugs/
>>>
>>> Packagers please note some plugins/elements have moved from -bad to
>>> -good (imagefreeze plugin, oss4 plugin, capsfilter element).
>>>
>>> md5sums and links:
>>>
>>> 794e8d737657c60b2d6f4d44475d8b59 gst-plugins-good-0.10.22.2.tar.gz
>>> 59403cd259529dee36ad09435898d80f gst-plugins-good-0.10.22.2.tar.bz2
>>>
>>> http://gstreamer.freedesktop.org/src/gst-plugins-good/pre/gst-plugins-good-0.10.22.2.tar.gz
>>> http://gstreamer.freedesktop.org/src/gst-plugins-good/pre/gst-plugins-good-0.10.22.2.tar.bz2
>>>
>>> bafd26e74b2bacecb59fd6c938888ed0 gst-plugins-ugly-0.10.14.2.tar.gz
>>> 41bf784355cce044c0a7072c20fa053c gst-plugins-ugly-0.10.14.2.tar.bz2
>>>
>>> http://gstreamer.freedesktop.org/src/gst-plugins-ugly/pre/gst-plugins-ugly-0.10.14.2.tar.gz
>>> http://gstreamer.freedesktop.org/src/gst-plugins-ugly/pre/gst-plugins-ugly-0.10.14.2.tar.bz2
>>>
>>> 2f5f14c58c50e1b476fb2a31af6270c4 gst-plugins-bad-0.10.18.2.tar.gz
>>> fe4fde65ed036c927427a158f0165298 gst-plugins-bad-0.10.18.2.tar.bz2
>>>
>>> http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.18.2.tar.gz
>>> http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.18.2.tar.bz2
>>>
>>> Cheers
>>> -Tim
>>>
>>>
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> Message: 7
>>> Date: Fri, 14 May 2010 16:19:22 -0700
>>> From: Loc Nguyen <[hidden email]>
>>> Subject: [gst-devel] Memory profiling and hunting memory leaks
>>> To: [hidden email]
>>> Message-ID: <[hidden email]>
>>> Content-Type: text/plain; charset=UTF-8; format=flowed
>>>
>>> Hey, I sent an email a few days ago but was never sent to the mailing
>>> list. I'm trying to hunt down some memory leaks in gstreamer on
>>> Windows. Can anyone advise on how core gstreamer devs are doing this?
>>> Any internal APIs that maybe useful for me to try?
>>>
>>> -Loc
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> Message: 8
>>> Date: Sat, 15 May 2010 09:44:04 +0300
>>> From: Marco Ballesio <[hidden email]>
>>> Subject: Re: [gst-devel] Black-and-white output
>>> To: Discussion of the development of GStreamer
>>> <[hidden email]>
>>> Message-ID:
>>> <[hidden email]>
>>> Content-Type: text/plain; charset="iso-8859-1"
>>>
>>> Hi,
>>>
>>> On Wed, May 12, 2010 at 7:11 PM, Louis-Simon Houde <[hidden email]>wrote:
>>>
>>>> Hello,
>>>>
>>>> The question might sound completely silly for you but does gstreamer needs
>>>> a video card to generate video output on command line with its gst-launch
>>>> command ?
>>>>
>>> it shouldn't matter as long as you're not rendering the output on the card
>>> itself.. can you please post the gst-launch command you're using?
>>>
>>> Regards
>>>
>>>
>>>> I'm asking this question because we use gst-launch to generate video
>>>> output. On one of the server, with identical command, the output is
>>>> black-and-white. One of the server doesn't have any video card because it is
>>>> hosted on Amazon EC2.
>>>>
>>>> Ubuntu versions are different also so it might be another cue. But it is
>>>> the same gstreamer version on both servers.
>>>>
>>>> Thanks
>>>>
>>>> ------------------------------
>>>> 10 000 $ de magasinage avec Hotmail. Inscrivez-vous!<http://go.microsoft.com/?linkid=9729716>
>>>>
>>>>
>>>> ------------------------------------------------------------------------------
>>>>
>>>>
>>>> _______________________________________________
>>>> gstreamer-devel mailing list
>>>> [hidden email]
>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>>>
>>>>
>>> -------------- next part --------------
>>> An HTML attachment was scrubbed...
>>>
>>> ------------------------------
>>>
>>> ------------------------------------------------------------------------------
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> _______________________________________________
>>> gstreamer-devel mailing list
>>> [hidden email]
>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>>
>>>
>>> End of gstreamer-devel Digest, Vol 48, Issue 23
>>> ***********************************************
>>>
>>> ------------------------------------------------------------------------
>>>
>>> ------------------------------------------------------------------------------
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> gstreamer-devel mailing list
>>> [hidden email]
>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>
>
>
>------------------------------
>
>Message: 2
>Date: Mon, 17 May 2010 11:04:53 +0200
>From: Marc Leeman <[hidden email]>
>Subject: Re: [gst-devel] RTSP Seek and DVB subtitles
>To: Alexander Olekhnovich <[hidden email]>
>Cc: Discussion of the development of GStreamer
> <[hidden email]>
>Message-ID: <[hidden email]>
>Content-Type: text/plain; charset="iso-8859-15"
>
>> I think ppl are interested :) At least I would really like to have a look at
>> that.
>
>I have tested it with Western European languages (Italian, French,
>German, English, Dutch); but not with Eastern (or others).
>
>Submitted it to gstreamer:
>https://bugzilla.gnome.org/show_bug.cgi?id=618850
>
>Using the pango=True parameter allows you to get an approx of the TT
>page on TV (there are some issues with the gfx chars in TT that have no
>counterpart in UTF8).
>
>--
> greetz, marc
>This is clearly another case of too many mad scientists, and not enough
>hunchbacks.
>crichton 2.6.26 #1 PREEMPT Tue Jul 29 21:17:59 CDT 2008 GNU/Linux
>-------------- next part --------------
>A non-text attachment was scrubbed...
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>
>------------------------------
>
>Message: 3
>Date: Mon, 17 May 2010 04:09:52 -0700 (PDT)
>From: Hunterwood <[hidden email]>
>Subject: [gst-devel] Caps problem when writing encoder/decoder
>To: [hidden email]
>Message-ID:
> <[hidden email]>
>Content-Type: text/plain; charset="us-ascii"
>
>
>Hi,
>
>I'm new to GStreamer and try to write a encoder and a decoder for a compressed audio format. But I have problems to get the Caps-negotiation to work.
>I base the code on the gst-template and have basically been looking at the alawenc/alawdec as my example.
>
>Everything works fine if I use "ANY" or "audio/x-raw-int" in the source of the encoder and sink of the decoder:
>static GstStaticPadTemplate gtest_enc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
> GST_PAD_SRC,
> GST_PAD_ALWAYS,
> GST_STATIC_CAPS ("ANY")
>static GstStaticPadTemplate gtest_dec_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
> GST_PAD_SINK,
> GST_PAD_ALWAYS,
> GST_STATIC_CAPS ("ANY")
>
>But when I change the MIME-type to something else, like for example "audio/mytesttype", I get the error "WARNING: erroneous pipeline: could not link gtestenc0 to gtestdec0".
>I'm using the pipeline:
>gst-launch-0.10.exe audiotestsrc ! gtestenc ! gtestdec ! audioresample ! autoaudiosink
>
>I get the same error if I only change one of the two caps:es and keep the other one as ANY.
>What have I missed? Do I need to register my new type in some way, or what could be the problem?
>
>Markus
>
>--
>View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Caps-problem-when-writing-encoder-decoder-tp2219436p2219436.html
>Sent from the GStreamer-devel mailing list archive at Nabble.com.
>-------------- next part --------------
>An HTML attachment was scrubbed...
>
>------------------------------
>
>------------------------------------------------------------------------------
>
>
>
>------------------------------
>
>_______________________________________________
>gstreamer-devel mailing list
>[hidden email]
>https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>
>End of gstreamer-devel Digest, Vol 48, Issue 27
>***********************************************
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