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<div class="moz-cite-prefix">On 08/27/2012 03:03 PM, Matt Pekar
wrote:<br>
</div>
<blockquote
cite="mid:CANw9OncFtOoPKV92hAhp58xMERfDsDoG6U8+NfXMMWKhpr+9eA@mail.gmail.com"
type="cite">I set my live appsrc video elements like this:
<div><br>
</div>
<div>g_object_set(G_OBJECT(appsrc), "caps", src_caps, NULL);</div>
<div>
<div>g_object_set(G_OBJECT(appsrc), "is-live", TRUE, NULL);</div>
<div>
g_object_set(G_OBJECT(appsrc), "block", FALSE, NULL);</div>
<div>g_object_set(G_OBJECT(appsrc), "do-timestamp", TRUE, NULL);</div>
</div>
</blockquote>
FYI: a more efficient way would be to do it in one call:<br>
<div>g_object_set(G_OBJECT(appsrc), "caps", src_caps, "is-live",
TRUE, "block", FALSE,"do-timestamp", TRUE, NULL);</div>
<br>
Stefan<br>
<blockquote
cite="mid:CANw9OncFtOoPKV92hAhp58xMERfDsDoG6U8+NfXMMWKhpr+9eA@mail.gmail.com"
type="cite">
<div>
<div><br>
</div>
<div>I seem to recall other elements needing the timestamps.
Then I spawn a background thread that pushes at a constant
rate.</div>
<div><br>
</div>
<div>GstBuffer* buffer = gst_buffer_new_and_alloc(width * height
* 4);</div>
<div>
<div>GstFlowReturn flow_return =
gst_app_src_push_buffer(GST_APP_SRC(pd->appsrc), buffer);</div>
</div>
<br>
<div class="gmail_quote">
On Sun, Aug 26, 2012 at 10:22 PM, Soho Soho123 <span
dir="ltr"><<a moz-do-not-send="true"
href="mailto:soho123.2012@gmail.com" target="_blank">soho123.2012@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi All,<br>
<br>
Can someone give me example about how to set latency for
appsrc?<br>
I can find the API about latency setting,<br>
but I do not know how to use that:<br>
which value is suitable for a rtp receiver?<br>
Since I implement a rtp socket to receive rtp packet, it
can receive<br>
rtp packet, when I call gst_app_src_push_buffer to input
payload to<br>
appsrc element. It seems does not workable, the audio data
can not<br>
hear from alsa device.<br>
I would like to modify the property of Appsrc,<br>
like set latency, is-live,<br>
which value is suitable for a rtp receiver when I call API<br>
gst_app_src_set_latency?<br>
And How to set property "is-live"?<br>
<br>
Thanks!<br>
Best Regards,<br>
Soho<br>
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</blockquote>
</div>
<br>
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