fakesrc for audio stream.

[Tiago Katcipis]

Hi,


On Wed, Oct 24, 2012 at 6:26 PM, rasnaut <rasnaut at gmail.com> wrote:

> Hi to All!
> I working with gstreamer-1.0 and try use fakesrc for work with audiostream.
> For start trying I copy only 0 in buffer (I used MapInfo structure). So, my
> code:
>

Have you tried to use audiotestsrc instead of fakesrc ? (you will not need
the audiorate element)

http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-audiotestsrc.html

Best regards,
Tiago Katcipis


>
> main()
> {
>     gst_init (NULL,NULL);
>     loop = g_main_loop_new (NULL, FALSE);
>
>     /* setup pipeline */
>     pipeline = gst_pipeline_new ("pipeline");
>     g_assert(pipeline);
>     fakesrc = gst_element_factory_make ("fakesrc", "source");
>     g_assert(fakesrc);
>     flt = gst_element_factory_make ("capsfilter", "flt");
>     g_assert(flt);
>     rate = gst_element_factory_make ("audiorate", "rate");
>     g_assert(rate);
>     conv = gst_element_factory_make ("audioconvert", "conv");
>     g_assert(conv);
>     audiosink = gst_element_factory_make ("alsasink", "asink");
>     g_assert(videosink);
>
>     /* setup */
>     g_object_set (G_OBJECT (flt), "caps",
>                  gst_caps_new_simple ("audio/x-raw",
>                  "format",G_TYPE_STRING,"S16LE",
>                  "rate", G_TYPE_INT,16000,
>                   "channels", G_TYPE_INT, 1,
>                    NULL), NULL);
>     gst_bin_add_many (GST_BIN (pipeline), fakesrc, flt,rate, conv,
> audiosink, NULL);
>     if(!gst_element_link_many (fakesrc, flt, rate, conv, audiosink, NULL))
>     {
>         g_error("Link Error\n");
>     }
>
>     /* setup fake source */
>     g_object_set (G_OBJECT (fakesrc),
>           "signal-handoffs", TRUE,
>           "sizemax", 16000,
>           "sizetype", 2, NULL);
>     g_signal_connect (fakesrc, "handoff", G_CALLBACK (cb_handoff), NULL);
>
>     /* play */
>     gst_element_set_state (pipeline, GST_STATE_PLAYING);
>     g_main_loop_run (loop);
>
>     /* clean up */
>     gst_element_set_state (pipeline, GST_STATE_NULL);
>     gst_object_unref (GST_OBJECT (pipeline));
>     g_main_loop_unref (loop);
> }
>
> So, when I start my programm apear next error:
> GStreamer-CRITICAL **: gst_segment_to_running_time: assertion
> `segment->format == format' failed
>
> When I changed alsasink on fakesink, all work. Maybe it's because my
> audiocard is working with another details:
> format = SL32_LE
> rate = 48000
> channels = 2
>
> I tryed change my details:
>     g_object_set (G_OBJECT (flt), "caps",
>                  gst_caps_new_simple ("audio/x-raw",
>                  "format",G_TYPE_STRING,"S32LE",
>                  "rate", G_TYPE_INT,48000,
>                   "channels", G_TYPE_INT, 2,
>                    NULL), NULL);
>
>     g_object_set (G_OBJECT (fakesrc),
>           "signal-handoffs", TRUE,
>           "sizemax", 48000*2,
>           "sizetype", 8, NULL);
>
>  but error still stayed.
> Anybody know, what I do wrong?
>
>
>
> --
> View this message in context:
> http://gstreamer-devel.966125.n4.nabble.com/fakesrc-for-audio-stream-tp4656701.html
> Sent from the GStreamer-devel mailing list archive at Nabble.com.
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