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rtp: Update codes based on 1.18.4
[platform/upstream/gst-plugins-good.git]
/
gst
/
rtp
/
gstrtpopuspay.c
diff --git
a/gst/rtp/gstrtpopuspay.c
b/gst/rtp/gstrtpopuspay.c
index
ba54134
..
871a182
100644
(file)
--- a/
gst/rtp/gstrtpopuspay.c
+++ b/
gst/rtp/gstrtpopuspay.c
@@
-39,8
+39,7
@@
static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS
- ("audio/x-opus, channels = (int) [1, 2], channel-mapping-family = (int) 0")
+ GST_STATIC_CAPS ("audio/x-opus, channel-mapping-family = (int) 0")
);
static GstStaticPadTemplate gst_rtp_opus_pay_src_template =
);
static GstStaticPadTemplate gst_rtp_opus_pay_src_template =
@@
-181,7
+180,7
@@
gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
dts = GST_BUFFER_DTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
dts = GST_BUFFER_DTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
- outbuf = gst_rtp_b
uffer_new_allocate (
0, 0, 0);
+ outbuf = gst_rtp_b
ase_payload_allocate_output_buffer (basepayload,
0, 0, 0);
gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);