How to set capability for the src's of rtpmp4gpay?
Psychesnet Hsieh
psychesnet at gmail.com
Tue Aug 4 06:02:30 UTC 2020
Hi all,
Does anyone can provide me a sample code which can stream out h264/aac at
same time when RTSP client do a request? Thanks.
Psychesnet Hsieh <psychesnet at gmail.com> 於 2020年8月3日 週一 下午2:53寫道:
> Hi all,
>
> I'm testing gst-rtsp-server library, I call test_appsrc.cpp and change the
> input as appsrc(video), anything looks work good.
>
> And I try to add AAC stream. My appsrc can output complete AAC frame at
> each callback. So, I add
> 1. factory
>
>
> * gst_rtsp_media_factory_set_launch (factory, "( appsrc
> name=dummy_h264 ! rtph264pay name=pay0 pt=96 " " appsrc
> name=dummy_aac ! rtpmp4gpay name=pay1 pt=97 )");*
> 2. when connect is coming
>
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> * /* get our vsrc, we named it 'dummy' with the name property */
> asrc = gst_bin_get_by_name_recurse_up (GST_BIN (element), "dummy_aac");
> /* this instructs asrc that we will be dealing with timed buffer */
> gst_util_set_object_arg (G_OBJECT (asrc), "format", "time"); /*
> configure the caps of the video */ g_object_set (G_OBJECT (asrc),
> "caps", gst_caps_new_simple ("audio/mpeg",
> "format", G_TYPE_STRING, "AAC-LC", "layout", G_TYPE_STRING,
> "interleaved", "rate", G_TYPE_INT, 16000,
> "channels", G_TYPE_INT, 1, NULL), NULL); /* install the
> callback that will be called when a buffer is needed */ g_signal_connect
> (asrc, "need-data", (GCallback) need_aac_data, ctx); gst_object_unref
> (asrc);*
>
> At 2, the capability is for appsrc(audio).
> I double check the capability of rtpmp4gpay, like
>
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> * SRC template: 'src' Availability: Always Capabilities:
> application/x-rtp media: { (string)video, (string)audio,
> (string)application } payload: [ 96, 127 ]
> clock-rate: [ 1, 2147483647 ] encoding-name: MPEG4-GENERIC
> streamtype: { (string)4, (string)5 } mode: {
> (string)generic, (string)CELP-cbr, (string)CELP-vbr, (string)AAC-lbr,
> (string)AAC-hbr }*
>
> How do I set up the src's of rtpmp4gpay? Does anyone can give me a tip?
> Thanks.
>
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