mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-04-09 17:44:14 +00:00
Code Issues1 Projects Releases Wiki Activity

make some more things compile again

Browse source
This commit is contained in:
Wim Taymans 2011-10-27 19:00:52 +02:00
parent 4b6a226263
commit 9c14280b1d
13 changed files with 24 additions and 25 deletions

2
gst/audioparsers/gstaacparse.c
View file

@ -172,7 +172,7 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
if (sink_caps)
src_caps = gst_caps_copy (sink_caps);
else
src_caps = gst_caps_new_simple ("audio/mpeg", NULL);
src_caps = gst_caps_new_empty_simple ("audio/mpeg");
gst_caps_set_simple (src_caps, "framed", G_TYPE_BOOLEAN, TRUE,
"mpegversion", G_TYPE_INT, aacparse->mpegversion, NULL);

8
gst/avi/gstavidemux.c
View file

@ -139,17 +139,17 @@ gst_avi_demux_class_init (GstAviDemuxClass * klass)
gstelement_class->get_index = GST_DEBUG_FUNCPTR (gst_avi_demux_get_index);
audcaps = gst_riff_create_audio_template_caps ();
gst_caps_append (audcaps, gst_caps_new_simple ("audio/x-avi-unknown", NULL));
gst_caps_append (audcaps, gst_caps_new_empty_simple ("audio/x-avi-unknown"));
audiosrctempl = gst_pad_template_new ("audio_%02d",
GST_PAD_SRC, GST_PAD_SOMETIMES, audcaps);
vidcaps = gst_riff_create_video_template_caps ();
gst_caps_append (vidcaps, gst_riff_create_iavs_template_caps ());
gst_caps_append (vidcaps, gst_caps_new_simple ("video/x-avi-unknown", NULL));
gst_caps_append (vidcaps, gst_caps_new_empty_simple ("video/x-avi-unknown"));
videosrctempl = gst_pad_template_new ("video_%02d",
GST_PAD_SRC, GST_PAD_SOMETIMES, vidcaps);
subcaps = gst_caps_new_simple ("application/x-subtitle-avi", NULL);
subcaps = gst_caps_new_empty_simple ("application/x-subtitle-avi");
subsrctempl = gst_pad_template_new ("subtitle_%02d",
GST_PAD_SRC, GST_PAD_SOMETIMES, subcaps);
gst_element_class_add_pad_template (gstelement_class, audiosrctempl);
@ -2221,7 +2221,7 @@ gst_avi_demux_parse_stream (GstAviDemux * avi, GstBuffer * buf)
case GST_RIFF_FCC_txts:{
padname = g_strdup_printf ("subtitle_%02d", avi->num_t_streams);
templ = gst_element_class_get_pad_template (klass, "subtitle_%02d");
caps = gst_caps_new_simple ("application/x-subtitle-avi", NULL);
caps = gst_caps_new_empty_simple ("application/x-subtitle-avi");
tag_name = NULL;
avi->num_t_streams++;
break;

16
gst/flv/gstflvdemux.c
View file

@ -659,7 +659,7 @@ gst_flv_demux_audio_negotiate (GstFlvDemux * demux, guint32 codec_tag,
case 4:
case 5:
case 6:
caps = gst_caps_new_simple ("audio/x-nellymoser", NULL);
caps = gst_caps_new_empty_simple ("audio/x-nellymoser");
break;
case 10:
{
@ -693,13 +693,13 @@ gst_flv_demux_audio_negotiate (GstFlvDemux * demux, guint32 codec_tag,
break;
}
case 7:
caps = gst_caps_new_simple ("audio/x-alaw", NULL);
caps = gst_caps_new_empty_simple ("audio/x-alaw");
break;
case 8:
caps = gst_caps_new_simple ("audio/x-mulaw", NULL);
caps = gst_caps_new_empty_simple ("audio/x-mulaw");
break;
case 11:
caps = gst_caps_new_simple ("audio/x-speex", NULL);
caps = gst_caps_new_empty_simple ("audio/x-speex");
break;
default:
GST_WARNING_OBJECT (demux, "unsupported audio codec tag %u", codec_tag);
@ -1075,16 +1075,16 @@ gst_flv_demux_video_negotiate (GstFlvDemux * demux, guint32 codec_tag)
/* Generate caps for that pad */
switch (codec_tag) {
case 2:
caps = gst_caps_new_simple ("video/x-flash-video", NULL);
caps = gst_caps_new_empty_simple ("video/x-flash-video");
break;
case 3:
caps = gst_caps_new_simple ("video/x-flash-screen", NULL);
caps = gst_caps_new_empty_simple ("video/x-flash-screen");
break;
case 4:
caps = gst_caps_new_simple ("video/x-vp6-flash", NULL);
caps = gst_caps_new_empty_simple ("video/x-vp6-flash");
break;
case 5:
caps = gst_caps_new_simple ("video/x-vp6-alpha", NULL);
caps = gst_caps_new_empty_simple ("video/x-vp6-alpha");
break;
case 7:
caps =

3
gst/flv/gstflvmux.c
View file

@ -1172,8 +1172,7 @@ gst_flv_mux_write_header (GstFlvMux * mux)
gst_flv_mux_put_buffer_in_streamheader (&streamheader, audio_codec_data);
/* create the caps and put the streamheader in them */
caps = gst_caps_new_simple ("video/x-flv", NULL);
caps = gst_caps_make_writable (caps);
caps = gst_caps_new_empty_simple ("video/x-flv");
structure = gst_caps_get_structure (caps, 0);
gst_structure_set_value (structure, "streamheader", &streamheader);
g_value_unset (&streamheader);

2
gst/icydemux/gsticydemux.c
View file

@ -414,7 +414,7 @@ gst_icydemux_typefind_or_forward (GstICYDemux * icydemux, GstBuffer * buf)
if (G_UNLIKELY (icydemux->content_type)) {
if (!g_ascii_strcasecmp (icydemux->content_type, "video/nsv")) {
GST_DEBUG ("We have a NSV stream");
caps = gst_caps_new_simple ("video/x-nsv", NULL);
caps = gst_caps_new_empty_simple ("video/x-nsv");
} else {
GST_DEBUG ("Upstream Content-Type isn't supported");
g_free (icydemux->content_type);

4
gst/rtp/README
View file

@ -37,11 +37,11 @@ The following fields can or must (*) be specified in the structure:
The ssrc value currently in use. (default = the SSRC of the first RTP
packet)
clock-base: (uint) [0 - MAXINT]
timestamp-offset: (uint) [0 - MAXINT]
The RTP time representing time npt-start. (default = rtptime of first RTP
packet).
seqnum-base: (uint) [0 - MAXINT]
seqnum-offset: (uint) [0 - MAXINT]
The RTP sequence number representing the first rtp packet. When this
parameter is given, all sequence numbers below this seqnum should be
ignored. (default = seqnum of first RTP packet).

2
gst/rtp/gstrtpac3depay.c
View file

@ -100,7 +100,7 @@ gst_rtp_ac3_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("audio/ac3", NULL);
srccaps = gst_caps_new_empty_simple ("audio/ac3");
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);

2
gst/rtp/gstrtpceltdepay.c
View file

@ -171,7 +171,7 @@ gst_rtp_celt_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
GST_WRITE_UINT32_LE (ptr, 0); /* extra headers */
gst_buffer_unmap (buf, data, size);
srccaps = gst_caps_new_simple ("audio/x-celt", NULL);
srccaps = gst_caps_new_empty_simple ("audio/x-celt");
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);

2
gst/rtp/gstrtph264depay.c
View file

@ -299,7 +299,7 @@ gst_rtp_h264_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
clock_rate = 90000;
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("video/x-h264", NULL);
srccaps = gst_caps_new_empty_simple ("video/x-h264");
/* Base64 encoded, comma separated config NALs */
ps = gst_structure_get_string (structure, "sprop-parameter-sets");

2
gst/rtp/gstrtph264pay.c
View file

@ -358,7 +358,7 @@ gst_rtp_h264_pay_getcaps (GstBaseRTPPayload * payload, GstPad * pad,
}
any:
return gst_caps_new_simple ("video/x-h264", NULL);
return gst_caps_new_empty_simple ("video/x-h264");
}
/* take the currently configured SPS and PPS lists and set them on the caps as

2
gst/rtp/gstrtpspeexdepay.c
View file

@ -171,7 +171,7 @@ gst_rtp_speex_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
GST_WRITE_UINT32_LE (data, 0); /* reserved2 */
gst_buffer_unmap (buf, bdata, -1);
srccaps = gst_caps_new_simple ("audio/x-speex", NULL);
srccaps = gst_caps_new_empty_simple ("audio/x-speex");
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);

2
gst/rtp/gstrtptheoradepay.c
View file

@ -339,7 +339,7 @@ gst_rtp_theora_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
}
/* set caps on pad and on header */
srccaps = gst_caps_new_simple ("video/x-theora", NULL);
srccaps = gst_caps_new_empty_simple ("video/x-theora");
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);

2
gst/rtp/gstrtpvorbisdepay.c
View file

@ -377,7 +377,7 @@ gst_rtp_vorbis_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
depayload->clock_rate = clock_rate;
/* set caps on pad and on header */
srccaps = gst_caps_new_simple ("audio/x-vorbis", NULL);
srccaps = gst_caps_new_empty_simple ("audio/x-vorbis");
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);