Fix gtreamer payloader sending fragmented frames.

1 files changed, 10 insertions, 6 deletions

diff --git a/audio/gstrtpsbcpay.c b/audio/gstrtpsbcpay.c
index fdb42d18..25bf70eb 100644
--- a/audio/gstrtpsbcpay.c
+++ b/audio/gstrtpsbcpay.c
@@ -162,6 +162,8 @@ static GstFlowReturn gst_rtp_sbc_pay_flush_buffers(GstRtpSBCPay *sbcpay)
 	GstBuffer* outbuf;
 	guint8 *payload_data;
 	guint8 *data;
+	guint frame_count;
+	guint payload_length;
 	struct rtp_payload *payload;
 
 	if (sbcpay->frame_length == 0) {
@@ -176,25 +178,27 @@ static GstFlowReturn gst_rtp_sbc_pay_flush_buffers(GstRtpSBCPay *sbcpay)
 		0, 0);
 
 	max_payload = MIN(max_payload, available);
+	frame_count = max_payload / sbcpay->frame_length;
+	payload_length = frame_count * sbcpay->frame_length;
 
-	outbuf = gst_rtp_buffer_new_allocate(max_payload +
+	outbuf = gst_rtp_buffer_new_allocate(payload_length +
 			RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
 
 	gst_rtp_buffer_set_payload_type(outbuf,
 			GST_BASE_RTP_PAYLOAD_PT(sbcpay));
 
-	data = gst_adapter_take(sbcpay->adapter, max_payload);
 	payload_data = gst_rtp_buffer_get_payload(outbuf);
-
 	payload = (struct rtp_payload*) payload_data;
 	memset(payload, 0, sizeof(struct rtp_payload));
-	payload->frame_count = max_payload / sbcpay->frame_length;
+	payload->frame_count = frame_count;
 
-	memcpy(payload_data + RTP_SBC_PAYLOAD_HEADER_SIZE, data, max_payload);
+	data = gst_adapter_take(sbcpay->adapter, payload_length);
+	memcpy(payload_data + RTP_SBC_PAYLOAD_HEADER_SIZE, data,
+			payload_length);
 	g_free(data);
 
 	GST_BUFFER_TIMESTAMP(outbuf) = sbcpay->timestamp;
-	GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", max_payload);
+	GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", payload_length);
 
 	return gst_basertppayload_push(GST_BASE_RTP_PAYLOAD(sbcpay), outbuf);
 }