static GstFlowReturn
gst_rtp_sbc_pay_flush_buffers (GstRtpSBCPay * sbcpay)
{
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
guint available;
guint max_payload;
GstBuffer *outbuf, *paybuf;
guint8 *payload_data;
guint frame_count;
guint payload_length;
struct rtp_payload *payload;
if (sbcpay->frame_length == 0) {
GST_ERROR_OBJECT (sbcpay, "Frame length is 0");
return GST_FLOW_ERROR;
}
available = gst_adapter_available (sbcpay->adapter);
max_payload =
gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU (sbcpay) -
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
max_payload = MIN (max_payload, available);
frame_count = max_payload / sbcpay->frame_length;
payload_length = frame_count * sbcpay->frame_length;
if (payload_length == 0) /* Nothing to send */
return GST_FLOW_OK;
outbuf = gst_rtp_buffer_new_allocate (RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
/* get payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_BASE_PAYLOAD_PT (sbcpay));
/* write header and copy data into payload */
payload_data = gst_rtp_buffer_get_payload (&rtp);
payload = (struct rtp_payload *) payload_data;
memset (payload, 0, sizeof (struct rtp_payload));
payload->frame_count = frame_count;
gst_rtp_buffer_unmap (&rtp);
paybuf = gst_adapter_take_buffer_fast (sbcpay->adapter, payload_length);
gst_rtp_copy_meta (GST_ELEMENT_CAST (sbcpay), outbuf, paybuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
outbuf = gst_buffer_append (outbuf, paybuf);
/* FIXME: what about duration? */
GST_BUFFER_PTS (outbuf) = sbcpay->timestamp;
GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", payload_length);
return gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (sbcpay), outbuf);
}
static GstFlowReturn
gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
{
guint avail, mtu;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
avail = gst_adapter_available (rtpmp2tpay->adapter);
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp2tpay);
while (avail > 0 && (ret == GST_FLOW_OK)) {
guint towrite;
guint payload_len;
guint packet_len;
GstBuffer *paybuf;
/* this will be the total length of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, mtu);
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
payload_len -= payload_len % 188;
/* need whole packets */
if (!payload_len)
break;
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
/* get payload */
paybuf = gst_adapter_take_buffer_fast (rtpmp2tpay->adapter, payload_len);
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmp2tpay), outbuf, paybuf, 0);
outbuf = gst_buffer_append (outbuf, paybuf);
avail -= payload_len;
GST_BUFFER_PTS (outbuf) = rtpmp2tpay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %u",
(guint) gst_buffer_get_size (outbuf));
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp2tpay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
{
GstBuffer *outbuf, *payload_buf;
GstFlowReturn ret;
guint avail;
GstRTPBuffer rtp = { NULL };
avail = gst_adapter_available (pay->adapter);
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
GST_BUFFER_PTS (outbuf) = pay->timestamp;
GST_BUFFER_DURATION (outbuf) = pay->duration;
/* copy G723 data as payload */
payload_buf = gst_adapter_take_buffer_fast (pay->adapter, avail);
pay->timestamp = GST_CLOCK_TIME_NONE;
pay->duration = 0;
/* set discont and marker */
if (pay->discont) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
gst_rtp_buffer_set_marker (&rtp, TRUE);
pay->discont = FALSE;
}
gst_rtp_buffer_unmap (&rtp);
gst_rtp_copy_meta (GST_ELEMENT_CAST (pay), outbuf, payload_buf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
outbuf = gst_buffer_append (outbuf, payload_buf);
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (pay), outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
guint16 frag_offset;
GstBufferList *list;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. In the case the
* adapter has more than one MTU, we need to split the MPA data
* over multiple packets. The frag_offset in each packet header
* needs to be updated with the position in the MPA frame. */
avail = gst_adapter_available (rtpmpapay->adapter);
ret = GST_FLOW_OK;
list =
gst_buffer_list_new_sized (avail / (GST_RTP_BASE_PAYLOAD_MTU (rtpmpapay) -
RTP_HEADER_LEN) + 1);
frag_offset = 0;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
GstRTPBuffer rtp = { NULL };
GstBuffer *paybuf;
/* this will be the total length of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, GST_RTP_BASE_PAYLOAD_MTU (rtpmpapay));
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
payload_len -= 4;
gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_PAYLOAD_MPA);
/*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | MBZ | Frag_offset |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
payload = gst_rtp_buffer_get_payload (&rtp);
payload[0] = 0;
payload[1] = 0;
payload[2] = frag_offset >> 8;
payload[3] = frag_offset & 0xff;
avail -= payload_len;
frag_offset += payload_len;
if (avail == 0)
gst_rtp_buffer_set_marker (&rtp, TRUE);
gst_rtp_buffer_unmap (&rtp);
paybuf = gst_adapter_take_buffer_fast (rtpmpapay->adapter, payload_len);
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmpapay), outbuf, paybuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
outbuf = gst_buffer_append (outbuf, paybuf);
GST_BUFFER_PTS (outbuf) = rtpmpapay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
gst_buffer_list_add (list, outbuf);
}
ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmpapay), list);
return ret;
}
/**
* gst_rtp_base_audio_payload_flush:
* @baseaudiopayload: a #GstRTPBasePayload
* @payload_len: length of payload
* @timestamp: a #GstClockTime
*
* Create an RTP buffer and store @payload_len bytes of the adapter as the
* payload. Set the timestamp on the new buffer to @timestamp before pushing
* the buffer downstream.
*
* If @payload_len is -1, all pending bytes will be flushed. If @timestamp is
* -1, the timestamp will be calculated automatically.
*
* Returns: a #GstFlowReturn
*/
GstFlowReturn
gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
guint payload_len, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf;
GstFlowReturn ret;
GstAdapter *adapter;
guint64 distance;
priv = baseaudiopayload->priv;
adapter = priv->adapter;
basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
if (payload_len == -1)
payload_len = gst_adapter_available (adapter);
/* nothing to do, just return */
if (payload_len == 0)
return GST_FLOW_OK;
if (timestamp == -1) {
/* calculate the timestamp */
timestamp = gst_adapter_prev_pts (adapter, &distance);
GST_LOG_OBJECT (baseaudiopayload,
"last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
GST_TIME_ARGS (timestamp), distance);
if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) {
/* convert the number of bytes since the last timestamp to time and add to
* the last seen timestamp */
timestamp += priv->bytes_to_time (baseaudiopayload, distance);
}
}
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
if (priv->buffer_list && gst_adapter_available_fast (adapter) >= payload_len) {
GstBuffer *buffer;
/* we can quickly take a buffer out of the adapter without having to copy
* anything. */
buffer = gst_adapter_take_buffer (adapter, payload_len);
ret =
gst_rtp_base_audio_payload_push_buffer (baseaudiopayload, buffer,
timestamp);
} else {
GstBuffer *paybuf;
CopyMetaData data;
/* create buffer to hold the payload */
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
paybuf = gst_adapter_take_buffer_fast (adapter, payload_len);
data.pay = baseaudiopayload;
data.outbuf = outbuf;
gst_buffer_foreach_meta (paybuf, foreach_metadata, &data);
outbuf = gst_buffer_append (outbuf, paybuf);
/* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_h263p_pay_flush (GstRtpH263PPay * rtph263ppay)
{
guint avail;
GstBufferList *list = NULL;
GstBuffer *outbuf = NULL;
GstFlowReturn ret;
gboolean fragmented = FALSE;
avail = gst_adapter_available (rtph263ppay->adapter);
if (avail == 0)
return GST_FLOW_OK;
fragmented = FALSE;
/* This algorithm assumes the H263/+/++ encoder sends complete frames in each
* buffer */
/* With Fragmentation Mode at GST_FRAGMENTATION_MODE_NORMAL:
* This algorithm implements the Follow-on packets method for packetization.
* This assumes low packet loss network.
* With Fragmentation Mode at GST_FRAGMENTATION_MODE_SYNC:
* This algorithm separates large frames at synchronisation points (Segments)
* (See RFC 4629 section 6). It would be interesting to have a property such as network
* quality to select between both packetization methods */
/* TODO Add VRC supprt (See RFC 4629 section 5.2) */
while (avail > 0) {
guint towrite;
guint8 *payload;
gint header_len;
guint next_gop = 0;
gboolean found_gob = FALSE;
GstRTPBuffer rtp = { NULL };
GstBuffer *payload_buf;
if (rtph263ppay->fragmentation_mode == GST_FRAGMENTATION_MODE_SYNC) {
/* start after 1st gop possible */
/* Check if we have a gob or eos , eossbs */
/* FIXME EOS and EOSSBS packets should never contain any gobs and vice-versa */
next_gop =
gst_adapter_masked_scan_uint32 (rtph263ppay->adapter, 0xffff8000,
0x00008000, 0, avail);
if (next_gop == 0) {
GST_DEBUG_OBJECT (rtph263ppay, " Found GOB header");
found_gob = TRUE;
}
/* Find next and cut the packet accordingly */
/* TODO we should get as many gobs as possible until MTU is reached, this
* code seems to just get one GOB per packet */
if (next_gop == 0 && avail > 3)
next_gop =
gst_adapter_masked_scan_uint32 (rtph263ppay->adapter, 0xffff8000,
0x00008000, 3, avail - 3);
GST_DEBUG_OBJECT (rtph263ppay, " Next GOB Detected at : %d", next_gop);
if (next_gop == -1)
next_gop = 0;
}
/* for picture start frames (non-fragmented), we need to remove the first
* two 0x00 bytes and set P=1 */
if (!fragmented || found_gob) {
gst_adapter_flush (rtph263ppay->adapter, 2);
avail -= 2;
}
header_len = 2;
towrite = MIN (avail, gst_rtp_buffer_calc_payload_len
(GST_RTP_BASE_PAYLOAD_MTU (rtph263ppay) - header_len, 0, 0));
if (next_gop > 0)
towrite = MIN (next_gop, towrite);
outbuf = gst_rtp_buffer_new_allocate (header_len, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* last fragment gets the marker bit set */
gst_rtp_buffer_set_marker (&rtp, avail > towrite ? 0 : 1);
payload = gst_rtp_buffer_get_payload (&rtp);
/* 0 1
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | RR |P|V| PLEN |PEBIT|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
/* if fragmented or gop header , write p bit =1 */
payload[0] = (fragmented && !found_gob) ? 0x00 : 0x04;
payload[1] = 0;
GST_BUFFER_PTS (outbuf) = rtph263ppay->first_timestamp;
GST_BUFFER_DURATION (outbuf) = rtph263ppay->first_duration;
gst_rtp_buffer_unmap (&rtp);
payload_buf = gst_adapter_take_buffer_fast (rtph263ppay->adapter, towrite);
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtph263ppay), outbuf, payload_buf,
g_quark_from_static_string (GST_META_TAG_VIDEO_STR));
outbuf = gst_buffer_append (outbuf, payload_buf);
avail -= towrite;
/* If more data is available and this is our first iteration,
* we create a buffer list and remember that we're fragmented.
*
* If we're fragmented already, add buffers to the previously
* created buffer list.
*
* Otherwise fragmented will be FALSE and we just push the single output
* buffer, and no list is allocated.
*/
if (avail && !fragmented) {
fragmented = TRUE;
list = gst_buffer_list_new ();
gst_buffer_list_add (list, outbuf);
} else if (fragmented) {
gst_buffer_list_add (list, outbuf);
}
}
if (fragmented) {
ret =
gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtph263ppay),
list);
} else {
ret =
gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtph263ppay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay)
{
guint avail, total;
GstBuffer *outbuf;
GstFlowReturn ret;
guint mtu;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. In the case the
* adapter has more than one MTU, we need to fragment the MPEG data
* over multiple packets. */
total = avail = gst_adapter_available (rtpmp4gpay->adapter);
ret = GST_FLOW_OK;
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4gpay);
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
GstRTPBuffer rtp = { NULL };
GstBuffer *paybuf;
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
/* fill one MTU or all available bytes, we need 4 spare bytes for
* the AU header. */
towrite = MIN (packet_len, mtu - 4);
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
GST_DEBUG_OBJECT (rtpmp4gpay,
"avail %d, towrite %d, packet_len %d, payload_len %d", avail, towrite,
packet_len, payload_len);
/* create buffer to hold the payload, also make room for the 4 header bytes. */
outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* copy payload */
payload = gst_rtp_buffer_get_payload (&rtp);
/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
* |AU-headers-length|AU-header|AU-header| |AU-header|padding|
* | | (1) | (2) | | (n) | bits |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
*/
/* AU-headers-length, we only have 1 AU-header */
payload[0] = 0x00;
payload[1] = 0x10; /* we use 16 bits for the header */
/* +---------------------------------------+
* | AU-size |
* +---------------------------------------+
* | AU-Index / AU-Index-delta |
* +---------------------------------------+
* | CTS-flag |
* +---------------------------------------+
* | CTS-delta |
* +---------------------------------------+
* | DTS-flag |
* +---------------------------------------+
* | DTS-delta |
* +---------------------------------------+
* | RAP-flag |
* +---------------------------------------+
* | Stream-state |
* +---------------------------------------+
*/
/* The AU-header, no CTS, DTS, RAP, Stream-state
*
* AU-size is always the total size of the AU, not the fragmented size
*/
payload[2] = (total & 0x1fe0) >> 5;
payload[3] = (total & 0x1f) << 3; /* we use 13 bits for the size, 3 bits index */
/* marker only if the packet is complete */
gst_rtp_buffer_set_marker (&rtp, avail <= payload_len);
gst_rtp_buffer_unmap (&rtp);
paybuf = gst_adapter_take_buffer_fast (rtpmp4gpay->adapter, payload_len);
outbuf = gst_buffer_append (outbuf, paybuf);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_timestamp;
GST_BUFFER_DURATION (outbuf) = rtpmp4gpay->first_duration;
if (rtpmp4gpay->frame_len) {
GST_BUFFER_OFFSET (outbuf) = rtpmp4gpay->offset;
rtpmp4gpay->offset += rtpmp4gpay->frame_len;
}
if (rtpmp4gpay->discont) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
/* Only the first outputted buffer has the DISCONT flag */
rtpmp4gpay->discont = FALSE;
}
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp4gpay), outbuf);
avail -= payload_len;
}
return ret;
}
static GstFlowReturn
gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay)
{
GstBuffer *outbuf;
GstFlowReturn ret;
guint avail;
guint8 *payload;
avail = gst_adapter_available (rtpmpvpay->adapter);
ret = GST_FLOW_OK;
while (avail > 0) {
guint towrite;
guint packet_len;
guint payload_len;
GstRTPBuffer rtp = { NULL };
GstBuffer *paybuf;
packet_len = gst_rtp_buffer_calc_packet_len (avail, 4, 0);
towrite = MIN (packet_len, GST_RTP_BASE_PAYLOAD_MTU (rtpmpvpay));
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 4, 0);
outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
payload = gst_rtp_buffer_get_payload (&rtp);
/* enable MPEG Video-specific header
*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | MBZ |T| TR | |N|S|B|E| P | | BFC | | FFC |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* AN FBV FFV
*/
/* fill in the MPEG Video-specific header
* data is set to 0x0 here
*/
memset (payload, 0x0, 4);
avail -= payload_len;
gst_rtp_buffer_set_marker (&rtp, avail == 0);
gst_rtp_buffer_unmap (&rtp);
paybuf = gst_adapter_take_buffer_fast (rtpmpvpay->adapter, payload_len);
outbuf = gst_buffer_append (outbuf, paybuf);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpvpay->first_ts;
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmpvpay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay)
{
GstFlowReturn ret;
guint avail;
GstBufferList *list;
GstBuffer *outbuf;
guint8 *payload;
avail = gst_adapter_available (rtpmpvpay->adapter);
ret = GST_FLOW_OK;
list =
gst_buffer_list_new_sized (avail / (GST_RTP_BASE_PAYLOAD_MTU (rtpmpvpay) -
RTP_HEADER_LEN) + 1);
while (avail > 0) {
guint towrite;
guint packet_len;
guint payload_len;
GstRTPBuffer rtp = { NULL };
GstBuffer *paybuf;
packet_len = gst_rtp_buffer_calc_packet_len (avail + 4, 0, 0);
towrite = MIN (packet_len, GST_RTP_BASE_PAYLOAD_MTU (rtpmpvpay));
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);
payload_len -= 4;
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
payload = gst_rtp_buffer_get_payload (&rtp);
/* enable MPEG Video-specific header
*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | MBZ |T| TR | |N|S|B|E| P | | BFC | | FFC |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* AN FBV FFV
*/
/* fill in the MPEG Video-specific header
* data is set to 0x0 here
*/
memset (payload, 0x0, 4);
avail -= payload_len;
gst_rtp_buffer_set_marker (&rtp, avail == 0);
gst_rtp_buffer_unmap (&rtp);
paybuf = gst_adapter_take_buffer_fast (rtpmpvpay->adapter, payload_len);
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmpvpay), outbuf, paybuf,
g_quark_from_static_string (GST_META_TAG_VIDEO_STR));
outbuf = gst_buffer_append (outbuf, paybuf);
GST_BUFFER_PTS (outbuf) = rtpmpvpay->first_ts;
gst_buffer_list_add (list, outbuf);
}
ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmpvpay), list);
return ret;
}
static GstFlowReturn
gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay)
{
guint avail, FT, NF, mtu;
GstBuffer *outbuf;
GstFlowReturn ret;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. In the case the
* adapter has more than one MTU, we need to split the AC3 data
* over multiple packets. */
avail = gst_adapter_available (rtpac3pay->adapter);
ret = GST_FLOW_OK;
FT = 0;
/* number of frames */
NF = rtpac3pay->NF;
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpac3pay);
GST_LOG_OBJECT (rtpac3pay, "flushing %u bytes", avail);
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
GstRTPBuffer rtp = { NULL, };
GstBuffer *payload_buffer;
/* this will be the total length of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (2 + avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, mtu);
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (2, 0, 0);
if (FT == 0) {
/* check if it all fits */
if (towrite < packet_len) {
guint maxlen;
GST_LOG_OBJECT (rtpac3pay, "we need to fragment");
/* check if we will be able to put at least 5/8th of the total
* frame in this first frame. */
if ((avail * 5) / 8 >= (payload_len - 2))
FT = 1;
else
FT = 2;
/* check how many fragments we will need */
maxlen = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
NF = (avail + maxlen - 1) / maxlen;
}
} else if (FT != 3) {
/* remaining fragment */
FT = 3;
}
/*
* 0 1
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | MBZ | FT| NF |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*
* FT: 0: one or more complete frames
* 1: initial 5/8 fragment
* 2: initial fragment not 5/8
* 3: other fragment
* NF: amount of frames if FT = 0, else number of fragments.
*/
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
GST_LOG_OBJECT (rtpac3pay, "FT %u, NF %u", FT, NF);
payload = gst_rtp_buffer_get_payload (&rtp);
payload[0] = (FT & 3);
payload[1] = NF;
payload_len -= 2;
if (avail == payload_len)
gst_rtp_buffer_set_marker (&rtp, TRUE);
gst_rtp_buffer_unmap (&rtp);
payload_buffer =
gst_adapter_take_buffer_fast (rtpac3pay->adapter, payload_len);
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpac3pay), outbuf, payload_buffer,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
outbuf = gst_buffer_append (outbuf, payload_buffer);
avail -= payload_len;
GST_BUFFER_PTS (outbuf) = rtpac3pay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpac3pay->duration;
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpac3pay), outbuf);
}
return ret;
}
