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rtpsbcpay: Fixes gstreamer caps and code cleanup.

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Luiz Augusto von Dentz 2008-01-30 14:21:43 +00:00 committed by Tim-Philipp Müller
parent a4f9624261
commit 687400ecf4
1 changed files with 9 additions and 3 deletions
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12
gst/rtp/gstrtpsbcpay.c
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@ -79,7 +79,7 @@ GST_ELEMENT_DETAILS ("RTP packet payloader",
static GstStaticPadTemplate gst_rtp_sbc_pay_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-sbc, " /* FIXME remove those caps? */
GST_STATIC_CAPS ("audio/x-sbc, "
"rate = (int) { 16000, 32000, 44100, 48000 }, "
"channels = (int) [ 1, 2 ], "
"mode = (string) { mono, dual, stereo, joint }, "
@ -91,7 +91,11 @@ static GstStaticPadTemplate gst_rtp_sbc_pay_sink_factory =
static GstStaticPadTemplate gst_rtp_sbc_pay_src_factory =
GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp") /* FIXME put things here */
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\","
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) { 16000, 32000, 44100, 48000 },"
"encoding-name = (string) \"SBC\"")
);
static void gst_rtp_sbc_pay_set_property (GObject * object, guint prop_id,
@ -150,7 +154,7 @@ gst_rtp_sbc_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
sbcpay->frame_length = frame_len;
gst_basertppayload_set_options (payload, "audio", FALSE, "SBC", rate);
gst_basertppayload_set_options (payload, "audio", TRUE, "SBC", rate);
GST_DEBUG_OBJECT (payload, "calculated frame length: %d ", frame_len);
@ -210,6 +214,8 @@ gst_rtp_sbc_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
GstRtpSBCPay *sbcpay;
guint available;
/* FIXME check for negotiation */
sbcpay = GST_RTP_SBC_PAY (payload);
gst_adapter_push (sbcpay->adapter, gst_buffer_copy (buffer));