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rtp: rename gst_rtp_sbc_pay_flush_buffers()
gst_rtp_sbc_pay_flush_buffers() is a misleading name. A better name would be gst_rtp_sbc_pay_drain_buffers(), because that's what it does, it drains any leftover queued data and pushes it downstream. "Flushing" in GStreamer typically means to throw away any queued data and not process/push it downstream. Signed-off-by: Michal Dzik <michal.dzik@streamunlimited.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/700>
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1 changed files with 5 additions and 5 deletions
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@ -173,7 +173,7 @@ gst_rtp_sbc_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
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}
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static GstFlowReturn
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gst_rtp_sbc_pay_flush_buffers (GstRtpSBCPay * sbcpay)
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gst_rtp_sbc_pay_drain_buffers (GstRtpSBCPay * sbcpay)
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{
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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guint available;
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@ -251,7 +251,7 @@ gst_rtp_sbc_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
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if (GST_BUFFER_IS_DISCONT (buffer)) {
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/* Try to flush whatever's left */
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gst_rtp_sbc_pay_flush_buffers (sbcpay);
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gst_rtp_sbc_pay_drain_buffers (sbcpay);
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/* Drop the rest */
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gst_adapter_flush (sbcpay->adapter,
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gst_adapter_available (sbcpay->adapter));
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@ -268,7 +268,7 @@ gst_rtp_sbc_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
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if (available + RTP_SBC_HEADER_TOTAL >=
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GST_RTP_BASE_PAYLOAD_MTU (sbcpay) ||
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(available > (sbcpay->min_frames * sbcpay->frame_length)))
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return gst_rtp_sbc_pay_flush_buffers (sbcpay);
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return gst_rtp_sbc_pay_drain_buffers (sbcpay);
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return GST_FLOW_OK;
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}
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@ -280,13 +280,13 @@ gst_rtp_sbc_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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gst_rtp_sbc_pay_flush_buffers (sbcpay);
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gst_rtp_sbc_pay_drain_buffers (sbcpay);
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break;
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case GST_EVENT_FLUSH_STOP:
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gst_adapter_clear (sbcpay->adapter);
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break;
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case GST_EVENT_SEGMENT:
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gst_rtp_sbc_pay_flush_buffers (sbcpay);
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gst_rtp_sbc_pay_drain_buffers (sbcpay);
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break;
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default:
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break;
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