Problem with scaletempo

[padam]

Hi,
So i try to test demo of scaletempo and implement that in my app but in both cases i get this error:

CRITICAL**: file ..\..\..\Source\gstreamer\gst\gstevent.c:
line 609: assertion 'start <= stop' failed

CRITICAL**: file ..\..\..\Source\gstreamer\gst\gstpad.c:
line 4840: assertion 'event != NULL' failed

Year ago @victorvictor wrote about this CLICK

Is there a simple solution for that? I do not get this error when i watch movie, but when i listen .mp3 i do.


EDIT:

I see that i get errors when i try change rate of movie:


gst_element_query_position(GST_ELEMENT(Player.pipeline), &fmt, &pos);
        GST_TIME_ARGS (pos);
                       
        gst_element_seek (Player.pipeline, Player.rate, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH,
                         GST_SEEK_TYPE_SET, pos,
                         GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE);


This is caused by getting ghostpad:

Player.sink = gst_element_factory_make ("autovideosink", "sink");

ghostpad = gst_element_get_pad (filter, "sink");
        gst_element_add_pad (Player.audioSink, gst_ghost_pad_new ("sink", ghostpad));
        gst_object_unref (ghostpad);


When i play music i can't get it, so i get this errors.
So it seems that scaletempo demo didn't work with audio files.

Does anyone know how to fix that?

[Stefan Sauer]

Hi,

> Hi,
> So i try to test demo of scaletempo and implement that in my app but in both
> cases i get this error:
>
> CRITICAL**: file ..\..\..\Source\gstreamer\gst\gstevent.c:
> line 609: assertion 'start <= stop' failed
>
> CRITICAL**: file ..\..\..\Source\gstreamer\gst\gstpad.c:
> line 4840: assertion 'event != NULL' failed
>
> Year ago @victorvictor wrote about this
> http://gstreamer-devel.966125.n4.nabble.com/Scaletempo-and-it-quot-rate-quot-property-td3630851.html
> CLICK
>
> Is there a simple solution for that? I do not get this error when i watch
> movie, but when i listen .mp3 i do.

What is the whole pipeline. Ideally file a bug and attach a small
standalone test example (a small command line only c-example) that shows
the bug. A patch that adds a unit test is even better :)

Stefan
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[padam]

Stefan, i have edited my first post.

And here you can see how i build pipeline

Player.loop = g_main_loop_new(NULL, FALSE);
        Player.pipeline = gst_element_factory_make("playbin2", "player");

GstCaps *caps;
        GstElement *filter;
        GstElement *format, *resample, *asink;
        GstPad *ghostpad;
        Player.scaletempo = gst_element_factory_make("scaletempo", "scaletempo");
        Player.sink = gst_element_factory_make ("autovideosink", "sink");
       
       
        caps = gst_caps_new_simple ("video/x-raw-rgb",
                           "width", G_TYPE_INT, 384,
                           "height", G_TYPE_INT, 288,
                           "framerate", GST_TYPE_FRACTION, 25, 1,
                           "bpp", G_TYPE_INT, 16,
                           "depth", G_TYPE_INT, 16,
                           "endianness", G_TYPE_INT, G_BYTE_ORDER,
                           NULL);

        gst_caps_unref(caps);

        filter = Player.scaletempo;
        g_object_set(G_OBJECT(Player.pipeline), "video-sink", Player.sink, (void*)NULL);

        Player.audioSink = gst_bin_new("audioline");
        gst_bin_add(GST_BIN(Player.audioSink), Player.scaletempo);
        MAKE_ELEMENT(Player.audioSink, format, "audioconvert",  "format");
        MAKE_ELEMENT(Player.audioSink, resample, "audioresample", "resample");
        MAKE_ELEMENT(Player.audioSink, asink, "autoaudiosink", "audio_sink");
        LINK_ELEMENTS(Player.scaletempo, format);
        LINK_ELEMENTS(format, resample);
        LINK_ELEMENTS(resample, asink);
       
        ghostpad = gst_element_get_pad (filter, "sink");
        gst_element_add_pad (Player.audioSink, gst_ghost_pad_new ("sink", ghostpad));
        gst_object_unref (ghostpad);

        g_object_set (G_OBJECT (Player.pipeline), "audio-sink", Player.audioSink, (void*)NULL);
        MAKE_ELEMENT(NULL, Player.scalerate_line, "autoaudiosink", "scaling_audio_sink");
        g_object_ref (Player.audioSink);
        g_object_ref (Player.scalerate_line);

[Stefan Sauer]

On 04/17/2012 01:05 PM, padam wrote:
> Stefan, i have edited my first post.
>
> And here you can see how i build pipeline

This works for me:
gst-launch playbin2 uri=file://$HOME/Music/1.mp3 audio-sink="scaletempo
! audioconvert ! audioresample ! autoaudiosink"

the code below is not complete and thus not enough to reproduce. If you
like some help then please follow my advice and post a complete code
snippet, that can be run.

Stefan

> Player.loop = g_main_loop_new(NULL, FALSE);
> Player.pipeline = gst_element_factory_make("playbin2", "player");
>
> GstCaps *caps;
> GstElement *filter;
> GstElement *format, *resample, *asink;
> GstPad *ghostpad;
> Player.scaletempo = gst_element_factory_make("scaletempo", "scaletempo");
> Player.sink = gst_element_factory_make ("autovideosink", "sink");
>
>
> caps = gst_caps_new_simple ("video/x-raw-rgb",
>   "width", G_TYPE_INT, 384,
>   "height", G_TYPE_INT, 288,
>   "framerate", GST_TYPE_FRACTION, 25, 1,
>   "bpp", G_TYPE_INT, 16,
>   "depth", G_TYPE_INT, 16,
>   "endianness", G_TYPE_INT, G_BYTE_ORDER,
>   NULL);
>
> gst_caps_unref(caps);
>
> filter = Player.scaletempo;
> g_object_set(G_OBJECT(Player.pipeline), "video-sink", Player.sink,
> (void*)NULL);
>
> Player.audioSink = gst_bin_new("audioline");
> gst_bin_add(GST_BIN(Player.audioSink), Player.scaletempo);
> MAKE_ELEMENT(Player.audioSink, format, "audioconvert",  "format");
> MAKE_ELEMENT(Player.audioSink, resample, "audioresample", "resample");
> MAKE_ELEMENT(Player.audioSink, asink, "autoaudiosink", "audio_sink");
> LINK_ELEMENTS(Player.scaletempo, format);
> LINK_ELEMENTS(format, resample);
> LINK_ELEMENTS(resample, asink);
>
> ghostpad = gst_element_get_pad (filter, "sink");
> gst_element_add_pad (Player.audioSink, gst_ghost_pad_new ("sink",
> ghostpad));
> gst_object_unref (ghostpad);
>
> g_object_set (G_OBJECT (Player.pipeline), "audio-sink", Player.audioSink,
> (void*)NULL);
> MAKE_ELEMENT(NULL, Player.scalerate_line, "autoaudiosink",
> "scaling_audio_sink");
> g_object_ref (Player.audioSink);
> g_object_ref (Player.scalerate_line);
>
> --
> View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Problem-with-scaletempo-tp4561625p4564397.html
> Sent from the GStreamer-devel mailing list archive at Nabble.com.
> _______________________________________________
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[padam]

You're right. So here you are http://wklej.org/id/734348/