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audiortppay: move function around

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This commit is contained in:
Wim Taymans 2009-09-02 13:13:54 +02:00
parent 5808041f44
commit c1ae0a2003
1 changed files with 43 additions and 43 deletions
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86
gst-libs/gst/rtp/gstbasertpaudiopayload.c
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@ -292,6 +292,49 @@ gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
/**
* gst_base_rtp_audio_payload_push:
* @baseaudiopayload: a #GstBaseRTPPayload
* @data: data to set as payload
* @payload_len: length of payload
* @timestamp: a #GstClockTime
*
* Create an RTP buffer and store @payload_len bytes of @data as the
* payload. Set the timestamp on the new buffer to @timestamp before pushing
* the buffer downstream.
*
* Returns: a #GstFlowReturn
*
* Since: 0.10.13
*/
GstFlowReturn
gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len, GstClockTime timestamp)
{
GstBaseRTPPayload *basepayload;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
gst_rtp_buffer_set_payload_type (outbuf, basepayload->pt);
payload = gst_rtp_buffer_get_payload (outbuf);
memcpy (payload, data, payload_len);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
/**
* gst_base_rtp_audio_payload_flush:
* @baseaudiopayload: a #GstBaseRTPPayload
@ -542,49 +585,6 @@ config_error:
}
}
/**
* gst_base_rtp_audio_payload_push:
* @baseaudiopayload: a #GstBaseRTPPayload
* @data: data to set as payload
* @payload_len: length of payload
* @timestamp: a #GstClockTime
*
* Create an RTP buffer and store @payload_len bytes of @data as the
* payload. Set the timestamp on the new buffer to @timestamp before pushing
* the buffer downstream.
*
* Returns: a #GstFlowReturn
*
* Since: 0.10.13
*/
GstFlowReturn
gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len, GstClockTime timestamp)
{
GstBaseRTPPayload *basepayload;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
gst_rtp_buffer_set_payload_type (outbuf, basepayload->pt);
payload = gst_rtp_buffer_get_payload (outbuf);
memcpy (payload, data, payload_len);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
static GstStateChangeReturn
gst_base_rtp_payload_audio_change_state (GstElement * element,
GstStateChange transition)