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sendrecv/gst: Add no-op audio/video converters

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This reduces the chance that someone will try to change the
audio/video source elements and get an error because they don't know
about the conversion elements. They will be no-ops in the usual case.

Closes https://github.com/centricular/gstwebrtc-demos/issues/8
This commit is contained in:
Nirbheek Chauhan 2018-04-01 01:15:16 +05:30
parent 7c5fbf1aca
commit 47bfa3cc27
1 changed files with 4 additions and 2 deletions
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6
webrtc/sendrecv/gst/webrtc-sendrecv.c
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@ -117,6 +117,8 @@ handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name,
g_assert_nonnull (sink);
if (g_strcmp0 (convert_name, "audioconvert") == 0) {
/* Might also need to resample, so add it just in case.
* Will be a no-op if it's not required. */
resample = gst_element_factory_make ("audioresample", NULL);
g_assert_nonnull (resample);
gst_bin_add_many (GST_BIN (pipe), q, conv, resample, sink, NULL);
@ -279,9 +281,9 @@ start_pipeline (void)
pipe1 =
gst_parse_launch ("webrtcbin name=sendrecv " STUN_SERVER
"videotestsrc pattern=ball ! queue ! vp8enc deadline=1 ! rtpvp8pay ! "
"videotestsrc pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! "
"queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
"audiotestsrc wave=red-noise ! queue ! opusenc ! rtpopuspay ! "
"audiotestsrc wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ",
&error);