diff options
| author | Luiz Augusto von Dentz <luiz.dentz@openbossa.org> | 2008-01-23 19:17:33 +0000 |
|---|---|---|
| committer | Luiz Augusto von Dentz <luiz.dentz@openbossa.org> | 2008-01-23 19:17:33 +0000 |
| commit | 1f9f22f64d5e7c66c6e2d20452f6e5918d0020fb (patch) | |
| tree | 3400e5b2a3e4c3fa482de52843ca5cf6c38aee65 /audio/gstrtpsbcpay.c | |
| parent | e8e68d294cd662970de6082473e30ecd6ee8dd61 (diff) | |
Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps.
Diffstat (limited to 'audio/gstrtpsbcpay.c')
| -rw-r--r-- | audio/gstrtpsbcpay.c | 6 |
1 files changed, 4 insertions, 2 deletions
diff --git a/audio/gstrtpsbcpay.c b/audio/gstrtpsbcpay.c index 68aa28a9..fdb42d18 100644 --- a/audio/gstrtpsbcpay.c +++ b/audio/gstrtpsbcpay.c @@ -193,10 +193,10 @@ static GstFlowReturn gst_rtp_sbc_pay_flush_buffers(GstRtpSBCPay *sbcpay) memcpy(payload_data + RTP_SBC_PAYLOAD_HEADER_SIZE, data, max_payload); g_free(data); - /* FIXME - timestamp it! */ + GST_BUFFER_TIMESTAMP(outbuf) = sbcpay->timestamp; GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", max_payload); - return gst_basertppayload_push (GST_BASE_RTP_PAYLOAD(sbcpay), outbuf); + return gst_basertppayload_push(GST_BASE_RTP_PAYLOAD(sbcpay), outbuf); } static GstFlowReturn gst_rtp_sbc_pay_handle_buffer(GstBaseRTPPayload *payload, @@ -208,6 +208,7 @@ static GstFlowReturn gst_rtp_sbc_pay_handle_buffer(GstBaseRTPPayload *payload, sbcpay = GST_RTP_SBC_PAY(payload); gst_adapter_push(sbcpay->adapter, gst_buffer_copy(buffer)); + sbcpay->timestamp = GST_BUFFER_TIMESTAMP(buffer); available = gst_adapter_available(sbcpay->adapter); if (available + RTP_SBC_HEADER_TOTAL >= GST_BASE_RTP_PAYLOAD_MTU(sbcpay) || @@ -325,6 +326,7 @@ static void gst_rtp_sbc_pay_init(GstRtpSBCPay *self, GstRtpSBCPayClass *klass) { self->adapter = gst_adapter_new(); self->frame_length = 0; + self->timestamp = 0; self->min_frames = DEFAULT_MIN_FRAMES; } |