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On Fri, Jul 3, 2009 at 10:10 PM, <[hidden email]> wrote:
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Today's Topics:
1. low framerate with udp (Mehta, Neel)
2. Re: analogue of gst-launch with caps (Tim-Philipp M?ller)
3. Re: analogue of gst-launch with caps (Tim-Philipp M?ller)
4. Re: Need help with first gstreamer program. (Ognen Bezanov)
5. A question regarding to dynamic pipeline change (Zheng, Huan)
6. Re: Save a stream to different file every hour (Juraj Kubelka)
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Message: 1
Date: Fri, 3 Jul 2009 16:52:29 +0530
From: "Mehta, Neel" <[hidden email]>
Subject: [gst-devel] low framerate with udp
To: Discussion of the development of GStreamer
<[hidden email]>
Message-ID:
<[hidden email]>
Content-Type: text/plain; charset="us-ascii"
Hi
I am currently streaming MPEG4 video via UDP (resolution 720P/30 fps). Currently using the Live555, and RTSP, the video can be played at 30 fps. Using Gstreamer, I am able to get only 21 fps via UDP (for VGA, I do get 30 fps). I am using gstreamer for transmission only. The encoding process is same. I am using appsrc to get the encoded data from application buffer. I am attaching the code that I am using currently for transmission using Gstreamer. Please help me out to increase the framerate.
The pipeline is basically: appsrc is-live=true name=source caps=" video/mpeg, mpegversion=(int)4, framerate=(fraction)1001/30000, width=(int)1280, height=(int)720" ! rtpmp4vpay send-config=TRUE ! udpsink host=172.24.136.232 port=5000
Note: I am using UDP instead of rtsp, as I am unable to write the code completely between appsrc and gst-rtsp-server. I have posted on this forum for help titled "appsrc and gst-rtsp-server" on 29th June, Monday, but I haven't received a help yet. Someone please help me out with this too.
Please fell free to ask for more information.
Thanks in advance.
Regards,
Neel.
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Message: 2
Date: Fri, 03 Jul 2009 12:30:24 +0100
From: Tim-Philipp M?ller <[hidden email]>
Subject: Re: [gst-devel] analogue of gst-launch with caps
To: [hidden email]
Message-ID: <1246620624.5472.2.camel@zingle>
Content-Type: text/plain; charset="UTF-8"
On Thu, 2009-07-02 at 17:52 +0530, Mehta, Neel wrote:
> Can anyone suggest a substitute of gst_parse_launch (string, NULL), so
> that I have the negotiated caps as the part of the output.
>
> i.e. I want an analogue of gst-launch-0.10 ?v.
A quick look at gst-launch.c reveals that -v boils down to:
g_signal_connect (pipeline, "deep-notify",
G_CALLBACK (gst_object_default_deep_notify), NULL);
Cheers
-Tim
------------------------------
Message: 3
Date: Fri, 03 Jul 2009 12:31:34 +0100
From: Tim-Philipp M?ller <[hidden email]>
Subject: Re: [gst-devel] analogue of gst-launch with caps
To: [hidden email]
Message-ID: <1246620694.5472.3.camel@zingle>
Content-Type: text/plain; charset="UTF-8"
On Thu, 2009-07-02 at 17:52 +0530, Mehta, Neel wrote:
> Can anyone suggest a substitute of gst_parse_launch (string, NULL), so
> that I have the negotiated caps as the part of the output.
>
> i.e. I want an analogue of gst-launch-0.10 ?v.
A quick look at gst-launch.c reveals that -v boils down to:
g_signal_connect (pipeline, "deep-notify",
G_CALLBACK (gst_object_default_deep_notify), NULL);
Cheers
-Tim
------------------------------
Message: 4
Date: Fri, 03 Jul 2009 13:00:29 +0100
From: Ognen Bezanov <[hidden email]>
Subject: Re: [gst-devel] Need help with first gstreamer program.
To: [hidden email]
Message-ID: <1246622429.9434.17.camel@PC133>
Content-Type: text/plain
No-one has any idea?
------------------------------
Message: 5
Date: Wed, 1 Jul 2009 16:20:44 +0800
From: "Zheng, Huan" <[hidden email]>
Subject: [gst-devel] A question regarding to dynamic pipeline change
To: gstreamer-devel <[hidden email]>
Message-ID:
<[hidden email]>
Content-Type: text/plain; charset="us-ascii"
Hi, dear developers
If I dynamically replace one element inside the pipeline, will the negotiation of the *whole pipeline* be performed again when the pipeline restarted next time?
Thanks a lot for your answering!
Best Regards, Zheng, Huan(ZBT)
OTC/SSD/SSG
Intel Asia-Pacific Research & Developement Ltd
Tel: 021-6116 6435
Inet: 8821 6435
Cub: 3W035
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Message: 6
Date: Fri, 3 Jul 2009 18:40:21 +0200
From: Juraj Kubelka <[hidden email]>
Subject: Re: [gst-devel] Save a stream to different file every hour
To: Discussion of the development of GStreamer
<[hidden email]>
Message-ID:
<[hidden email]>
Content-Type: text/plain; charset="utf-8"
Hopefully I got it and my piece of code works :) So, if anyone is interested
in saving a stream to a new file every hour, download here
http://files.getdropbox.com/u/710942/audio-2009-07-03_18-32.tar.gz
Best regards,
Jura
On Fri, Jul 3, 2009 at 11:42 AM, Juraj Kubelka <[hidden email]
> wrote:
> Hi!
>
> As I have played with it, it seams it does not work properly all the time.
> Can you, please, check my source code in Python? There are TODOs which tag
> lines with possible wrong gstreamer handling (as I suppose). I will
> appreciate any suggestions.
>
> the file (pipelineholder.py): http://cesta.pastebin.com/m3b0c7ad
> other two files witch work together: http://cesta.pastebin.com/m46860962
> http://cesta.pastebin.com/m5afca033
> full sourcecode (3 files, run ./pipelinemanager.py, pipelineholder.py is
> the main one): http://files.getdropbox.com/u/710942/audio.tar.gz
>
> Thank you in advance.
> Jura
>
>
> On Tue, Jun 23, 2009 at 4:34 PM, Juraj Kubelka <
> [hidden email]> wrote:
>
>> On Thu, Jun 18, 2009 at 8:46 AM, Arnout Vandecappelle <[hidden email]>wrote:
>>
>>> On Wednesday 17 June 2009 10:29:43 Viraj Karandikar wrote:
>>> > And after every hour you will have to: pause the pipeline, create new
>>> > filesink, disconnect old filesink, connect new filesink and close old
>>> > filesink.
>>>
>>> Actually, no. Pausing the pipeline will not (immediately) stop the
>>> threads
>>> in the source and queues. In addition, once it's paused any new data
>>> coming
>>> in on the live source will be dropped until it plays again, which
>>> probably is
>>> not the intention.
>>>
>>> Instead, you have to use blocking. Below there's a piece of code that I
>>> use.
>>> See also docs/design/part-block.txt. There should be high-level API for
>>> that, but nobody has bothered up to now to write it.
>>>
>>> You need to replace the decoder and muxer as well, so the proper headers
>>> are
>>> created in the new file. To simplify all that I put it in a bin (which
>>> can
>>> be unreffed as a whole).
>>>
>>> > You can do above things in another independent thread which will
>>> > just wait till 1 hour is complete.
>>>
>>> Or you can use g_timeout_add_seconds (3600, ...) in your main loop.
>>>
>>> Regards,
>>> Arnout
>>>
>>> --
>>>
>>> static gboolean destroy_bin_cb (gpointer user_data)
>>> {
>>> GstElement *oldbin = user_data;
>>>
>>> log_message("Destroying old bin.\n");
>>> if (pipeline)
>>> {
>>> gst_element_set_state (oldbin, GST_STATE_NULL);
>>> gst_bin_remove (GST_BIN(pipeline), oldbin);
>>> }
>>> return FALSE;
>>> }
>>>
>>>
>>> static void replace_filesink_blocked_cb (GstPad *pad, gboolean blocked,
>>> gpointer user_data)
>>> {
>>> if (blocked)
>>> {
>>> GstElement *oldbin;
>>> GstPad *sinkpad;
>>>
>>> log_message("Blocked filesink queue.\n");
>>> g_static_rec_mutex_lock(&mutex);
>>> log_message("Locked.\n");
>>> oldbin = filesinkbin;
>>> sinkpad = gst_element_get_static_pad(oldbin, "sink");
>>> if (!sinkpad)
>>> {
>>> log_message("replace_filesink_blocked_cb: oldbin doesn't have
>>> sink
>>> pad.\n");
>>> goto fail;
>>> }
>>> gst_pad_unlink(queuesrcpad, sinkpad);
>>> log_message("Unlinked.\n");
>>>
>>> /* Finalize the old tail. */
>>> /* Sending EOS should be done from here (we're in the queue
>>> thread).
>>> */
>>> gst_pad_send_event(sinkpad, gst_event_new_eos());
>>> log_message("Sent event.\n");
>>>
>>> log_message("Checked sinks.\n");
>>> /* Setting state should be done from the main thread. Do it before
>>> the
>>> pipeline has been set to PLAYING - implicitly, because that is
>>> also
>>> done in an idle callback, but one that is started after this
>>> one.
>>> */
>>> g_idle_add(destroy_bin_cb, oldbin);
>>>
>>> create_filesink_tail();
>>> log_message("Created filesink.\n");
>>> g_static_rec_mutex_unlock(&mutex);
>>> log_message("Unlocked mutex.\n");
>>>
>>> /* And unblock again. */
>>> gst_pad_set_blocked_async(queuesrcpad, FALSE,
>>> replace_filesink_blocked_cb, NULL);
>>>
>>> log_message("Done replacing filesink.\n");
>>> }
>>> else
>>> {
>>> log_message("Unblocked filesink queue.\n");
>>> }
>>> return;
>>>
>>> fail:
>>> pipeline_terminate_async();
>>> g_static_rec_mutex_unlock(&mutex);
>>> }
>>>
>>> static void replace_filesink()
>>> {
>>> if (queuesrcpad == NULL)
>>> log_message("replace_filesink while no queuesrcpad yet, waiting
>>> for it
>>> to appear.\n");
>>> else
>>> gst_pad_set_blocked_async(queuesrcpad, TRUE,
>>> replace_filesink_blocked_cb, NULL);
>>> }
>>>
>>>
>> Hi,
>>
>> thank you a lot for your help. I did it in Python and it seams it works.
>> Source code is on pastebin: http://pastebin.com/f3a80ed09
>> (pipelineholder.py <http://pastebin.com/f3a80ed09%28pipelineholder.py>)
>> and http://pastebin.com/f4ee27d6d (pipelinemanager). Any comments are
>> appreciated.
>>
>> Best regards,
>> Jura
>>
>
>
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hi I recorded the audio and video with out any encoding and decoding.Video is coming from the camera as mpeg4 and audio in the format of aac.My pipeline looks like this, gst-launch -e rtspsrc location="rtsp:// user:carinov123@10.0.0.104/live.sdp" name=rtsp ! rtpmp4vdepay ! mpeg4videoparse ! qtmux name=qt ! filesink location=a.mp4 rtsp. ! rtpmp4gdepay ! aacparse ! qt.
But when i tried to play it with mplayer,i got the video only, the out put of "mplayer a.mp4 " is, libavformat file format detected. [lavf] Audio stream found, -aid 0 [lavf] Video stream found, -vid 1
VIDEO: [mp4v] 320x240 24bpp 30.333 fps 0.0 kbps ( 0.0 kbyte/s) ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) ========================================================================== ========================================================================== Opening audio decoder: [faad] AAC (MPEG2/4 Advanced Audio Coding)
FAAD: compressed input bitrate missing, assuming 128kbit/s! AUDIO: 32000 Hz, 2 ch, s16le, 128.0 kbit/12.50% (ratio: 16000->128000) Selected audio codec: [faad] afm: faad (FAAD AAC (MPEG-2/MPEG-4 Audio)) ==========================================================================
[AO OSS] audio_setup: Can't open audio device /dev/dsp: Device or resource busy AO: [alsa] 32000Hz 2ch s16le (2 bytes per sample) Starting playback... FAAD: Failed to decode frame: Maximum number of scalefactor bands exceeded
-- Sreerenj B http://sreerenj.livejournal.com[hidden email]mob: +91 9995377714
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