/**
* 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 */
payload = gst_rtp_buffer_get_payload (outbuf);
memcpy (payload, data, payload_len);
/* set metadata */
gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
{
guint avail;
guint8 *payload;
GstFlowReturn ret;
GstBuffer *outbuf;
avail = gst_adapter_available (rtpmp2tpay->adapter);
if (avail == 0)
return GST_FLOW_OK;
outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
/* get payload */
payload = gst_rtp_buffer_get_payload (outbuf);
/* copy stuff from adapter to payload */
gst_adapter_copy (rtpmp2tpay->adapter, payload, 0, avail);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp2tpay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp2tpay), outbuf);
/* flush the adapter content */
gst_adapter_flush (rtpmp2tpay->adapter, avail);
return ret;
}
static GstFlowReturn
gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
{
GstBuffer *outbuf;
GstFlowReturn ret;
guint8 *payload;
guint avail;
avail = gst_adapter_available (pay->adapter);
outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
payload = gst_rtp_buffer_get_payload (outbuf);
GST_BUFFER_TIMESTAMP (outbuf) = pay->timestamp;
GST_BUFFER_DURATION (outbuf) = pay->duration;
/* copy G723 data as payload */
gst_adapter_copy (pay->adapter, payload, 0, avail);
/* flush bytes from adapter */
gst_adapter_flush (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 (outbuf, TRUE);
pay->discont = FALSE;
}
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (pay), outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay, GstClockTime timestamp,
GstClockTime duration)
{
GstBuffer *outbuf;
GstFlowReturn ret;
guint avail;
guint8 *payload;
gint packet_size;
gint payload_size;
avail = gst_adapter_available (rtpmpvpay->adapter);
packet_size = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);
/* check for the maximum size of the rtp buffer */
if (packet_size > GST_BASE_RTP_PAYLOAD_MTU (rtpmpvpay)) {
payload_size =
GST_BASE_RTP_PAYLOAD_MTU (rtpmpvpay) -
gst_rtp_buffer_calc_packet_len (4, 0, 0);
} else {
payload_size = avail;
}
outbuf = gst_rtp_buffer_new_allocate (4 + payload_size, 0, 0);
/* 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
*/
payload = gst_rtp_buffer_get_payload (outbuf);
/* fill in the MPEG Video-specific header */
memset (payload, 0x0, 4);
/* copy stuff from adapter to payload */
gst_adapter_copy (rtpmpvpay->adapter, payload + 4, 0, payload_size);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpvpay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmpvpay->duration;
GST_DEBUG_OBJECT (rtpmpvpay, "pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpvpay), outbuf);
gst_adapter_flush (rtpmpvpay->adapter, payload_size);
/* update the timestamp and duration */
rtpmpvpay->first_ts = timestamp;
rtpmpvpay->duration = duration;
/* check if there is enough data for another rtp buffer */
avail = gst_adapter_available (rtpmpvpay->adapter);
packet_size = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);
if (packet_size >= GST_BASE_RTP_PAYLOAD_MTU (rtpmpvpay) && ret == GST_FLOW_OK) {
GST_DEBUG_OBJECT (rtpmpvpay, "Have enough data for another rtp packet");
ret = gst_rtp_mpv_pay_flush (rtpmpvpay, timestamp, duration);
}
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;
packet_len = gst_rtp_buffer_calc_packet_len (avail, 4, 0);
towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmpvpay));
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 4, 0);
outbuf = gst_rtp_buffer_new_allocate (payload_len, 4, 0);
payload = gst_rtp_buffer_get_payload (outbuf);
/* 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);
gst_adapter_copy (rtpmpvpay->adapter, payload + 4, 0, payload_len);
gst_adapter_flush (rtpmpvpay->adapter, payload_len);
avail -= payload_len;
gst_rtp_buffer_set_marker (outbuf, avail == 0);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpvpay->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpvpay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay)
{
GstFlowReturn ret;
GstBuffer *buf, *outbuf;
guint8 *payload, *spayload;
guint payload_len;
GstClockTime duration;
payload_len = rtpceltpay->bytes + rtpceltpay->sbytes;
duration = rtpceltpay->qduration;
GST_DEBUG_OBJECT (rtpceltpay, "flushing out %u, duration %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (rtpceltpay->qduration));
/* get a big enough packet for the sizes + payloads */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
GST_BUFFER_DURATION (outbuf) = duration;
/* point to the payload for size headers and data */
spayload = gst_rtp_buffer_get_payload (outbuf);
payload = spayload + rtpceltpay->sbytes;
while ((buf = g_queue_pop_head (rtpceltpay->queue))) {
guint size;
/* copy first timestamp to output */
if (GST_BUFFER_TIMESTAMP (outbuf) == -1)
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
/* write the size to the header */
size = GST_BUFFER_SIZE (buf);
while (size > 0xff) {
*spayload++ = 0xff;
size -= 0xff;
}
*spayload++ = size;
size = GST_BUFFER_SIZE (buf);
/* copy payload */
memcpy (payload, GST_BUFFER_DATA (buf), size);
payload += size;
gst_buffer_unref (buf);
}
/* we consumed it all */
rtpceltpay->bytes = 0;
rtpceltpay->sbytes = 0;
rtpceltpay->qduration = 0;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpceltpay), outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_gsm_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRTPGSMPay *rtpgsmpay;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp, duration;
GstFlowReturn ret;
rtpgsmpay = GST_RTP_GSM_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
/* FIXME, only one GSM frame per RTP packet for now */
payload_len = size;
/* FIXME, just error out for now */
if (payload_len > GST_BASE_RTP_PAYLOAD_MTU (rtpgsmpay)) {
GST_ELEMENT_ERROR (rtpgsmpay, STREAM, ENCODE, (NULL),
("payload_len %u > mtu %u", payload_len,
GST_BASE_RTP_PAYLOAD_MTU (rtpgsmpay)));
return GST_FLOW_ERROR;
}
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy timestamp and duration */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = duration;
/* get payload */
payload = gst_rtp_buffer_get_payload (outbuf);
data = GST_BUFFER_DATA (buffer);
/* copy data in payload */
memcpy (&payload[0], data, size);
gst_buffer_unref (buffer);
GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_mp4v_pay_flush (GstRtpMP4VPay * rtpmp4vpay)
{
guint avail;
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 MP4V data
* over multiple packets. */
avail = gst_adapter_available (rtpmp4vpay->adapter);
ret = GST_FLOW_OK;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
/* 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 */
towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmp4vpay));
/* 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 (payload_len, 0, 0);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
gst_adapter_copy (rtpmp4vpay->adapter, payload, 0, payload_len);
gst_adapter_flush (rtpmp4vpay->adapter, payload_len);
avail -= payload_len;
gst_rtp_buffer_set_marker (outbuf, avail == 0);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4vpay->first_timestamp;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), outbuf);
}
return ret;
}
static GstFlowReturn gst_rtp_sbc_pay_flush_buffers(GstRtpSBCPay *sbcpay)
{
guint available;
guint max_payload;
GstBuffer *outbuf;
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_BASE_RTP_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(payload_length +
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
gst_rtp_buffer_set_payload_type(outbuf,
GST_BASE_RTP_PAYLOAD_PT(sbcpay));
payload_data = gst_rtp_buffer_get_payload(outbuf);
payload = (struct rtp_payload *) payload_data;
memset(payload, 0, sizeof(struct rtp_payload));
payload->frame_count = frame_count;
gst_adapter_copy(sbcpay->adapter, payload_data +
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, payload_length);
gst_adapter_flush(sbcpay->adapter, payload_length);
GST_BUFFER_TIMESTAMP(outbuf) = sbcpay->timestamp;
GST_DEBUG_OBJECT(sbcpay, "Pushing %d bytes", payload_length);
return gst_basertppayload_push(GST_BASE_RTP_PAYLOAD(sbcpay), outbuf);
}
static GstFlowReturn
gst_rtp_vraw_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
{
GstRtpVRawPay *rtpvrawpay;
GstFlowReturn ret = GST_FLOW_OK;
guint line, offset;
guint8 *data, *yp, *up, *vp;
guint ystride, uvstride;
guint size, pgroup;
guint mtu;
guint width, height;
gint field;
rtpvrawpay = GST_RTP_VRAW_PAY (payload);
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
GST_LOG_OBJECT (rtpvrawpay, "new frame of %u bytes", size);
/* get pointer and strides of the planes */
yp = data + rtpvrawpay->yp;
up = data + rtpvrawpay->up;
vp = data + rtpvrawpay->vp;
ystride = rtpvrawpay->ystride;
uvstride = rtpvrawpay->uvstride;
mtu = GST_BASE_RTP_PAYLOAD_MTU (payload);
/* amount of bytes for one pixel */
pgroup = rtpvrawpay->pgroup;
width = rtpvrawpay->width;
height = rtpvrawpay->height;
/* start with line 0, offset 0 */
for (field = 0; field < 1 + rtpvrawpay->interlaced; field++) {
line = field;
offset = 0;
/* write all lines */
while (line < height) {
guint left;
GstBuffer *out;
guint8 *outdata, *headers;
gboolean next_line;
guint length, cont, pixels;
/* get the max allowed payload length size, we try to fill the complete MTU */
left = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
out = gst_rtp_buffer_new_allocate (left, 0, 0);
if (field == 0) {
GST_BUFFER_TIMESTAMP (out) = GST_BUFFER_TIMESTAMP (buffer);
} else {
GST_BUFFER_TIMESTAMP (out) = GST_BUFFER_TIMESTAMP (buffer) +
GST_BUFFER_DURATION (buffer) / 2;
}
outdata = gst_rtp_buffer_get_payload (out);
GST_LOG_OBJECT (rtpvrawpay, "created buffer of size %u for MTU %u", left,
mtu);
/*
* 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
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Extended Sequence Number | Length |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* |F| Line No |C| Offset |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Length |F| Line No |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* |C| Offset | .
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ .
* . .
* . Two (partial) lines of video data .
* . .
* +---------------------------------------------------------------+
*/
/* need 2 bytes for the extended sequence number */
*outdata++ = 0;
*outdata++ = 0;
left -= 2;
/* the headers start here */
headers = outdata;
/* while we can fit at least one header and one pixel */
while (left > (6 + pgroup)) {
/* we need a 6 bytes header */
left -= 6;
/* get how may bytes we need for the remaining pixels */
pixels = width - offset;
length = (pixels * pgroup) / rtpvrawpay->xinc;
if (left >= length) {
/* pixels and header fit completely, we will write them and skip to the
* next line. */
next_line = TRUE;
} else {
/* line does not fit completely, see how many pixels fit */
pixels = (left / pgroup) * rtpvrawpay->xinc;
length = (pixels * pgroup) / rtpvrawpay->xinc;
next_line = FALSE;
}
GST_LOG_OBJECT (rtpvrawpay, "filling %u bytes in %u pixels", length,
pixels);
left -= length;
/* write length */
*outdata++ = (length >> 8) & 0xff;
*outdata++ = length & 0xff;
/* write line no */
*outdata++ = ((line >> 8) & 0x7f) | ((field << 7) & 0x80);
*outdata++ = line & 0xff;
if (next_line) {
/* go to next line we do this here to make the check below easier */
line += rtpvrawpay->yinc;
}
/* calculate continuation marker */
cont = (left > (6 + pgroup) && line < height) ? 0x80 : 0x00;
/* write offset and continuation marker */
*outdata++ = ((offset >> 8) & 0x7f) | cont;
*outdata++ = offset & 0xff;
if (next_line) {
/* reset offset */
offset = 0;
GST_LOG_OBJECT (rtpvrawpay, "go to next line %u", line);
} else {
offset += pixels;
GST_LOG_OBJECT (rtpvrawpay, "next offset %u", offset);
}
if (!cont)
break;
}
GST_LOG_OBJECT (rtpvrawpay, "consumed %u bytes",
(guint) (outdata - headers));
/* second pass, read headers and write the data */
while (TRUE) {
guint offs, lin;
/* read length and cont */
length = (headers[0] << 8) | headers[1];
lin = ((headers[2] & 0x7f) << 8) | headers[3];
offs = ((headers[4] & 0x7f) << 8) | headers[5];
cont = headers[4] & 0x80;
pixels = length / pgroup;
headers += 6;
GST_LOG_OBJECT (payload,
"writing length %u, line %u, offset %u, cont %d", length, lin, offs,
cont);
switch (rtpvrawpay->sampling) {
case GST_VIDEO_FORMAT_RGB:
case GST_VIDEO_FORMAT_RGBA:
case GST_VIDEO_FORMAT_BGR:
case GST_VIDEO_FORMAT_BGRA:
case GST_VIDEO_FORMAT_UYVY:
offs /= rtpvrawpay->xinc;
memcpy (outdata, yp + (lin * ystride) + (offs * pgroup), length);
outdata += length;
break;
case GST_VIDEO_FORMAT_AYUV:
{
gint i;
guint8 *datap;
datap = yp + (lin * ystride) + (offs * 4);
for (i = 0; i < pixels; i++) {
*outdata++ = datap[2];
*outdata++ = datap[1];
*outdata++ = datap[3];
datap += 4;
}
break;
}
case GST_VIDEO_FORMAT_I420:
{
gint i;
guint uvoff;
guint8 *yd1p, *yd2p, *udp, *vdp;
yd1p = yp + (lin * ystride) + (offs);
yd2p = yd1p + ystride;
uvoff =
(lin / rtpvrawpay->yinc * uvstride) + (offs / rtpvrawpay->xinc);
udp = up + uvoff;
vdp = vp + uvoff;
for (i = 0; i < pixels; i++) {
*outdata++ = *yd1p++;
*outdata++ = *yd1p++;
*outdata++ = *yd2p++;
*outdata++ = *yd2p++;
*outdata++ = *udp++;
*outdata++ = *vdp++;
}
break;
}
case GST_VIDEO_FORMAT_Y41B:
{
gint i;
guint uvoff;
guint8 *ydp, *udp, *vdp;
ydp = yp + (lin * ystride) + offs;
uvoff =
(lin / rtpvrawpay->yinc * uvstride) + (offs / rtpvrawpay->xinc);
udp = up + uvoff;
vdp = vp + uvoff;
for (i = 0; i < pixels; i++) {
*outdata++ = *udp++;
*outdata++ = *ydp++;
*outdata++ = *ydp++;
*outdata++ = *vdp++;
*outdata++ = *ydp++;
*outdata++ = *ydp++;
}
break;
}
default:
gst_buffer_unref (out);
goto unknown_sampling;
}
if (!cont)
break;
}
if (line >= height) {
GST_LOG_OBJECT (rtpvrawpay, "field/frame complete, set marker");
gst_rtp_buffer_set_marker (out, TRUE);
}
if (left > 0) {
GST_LOG_OBJECT (rtpvrawpay, "we have %u bytes left", left);
GST_BUFFER_SIZE (out) -= left;
}
/* push buffer */
ret = gst_basertppayload_push (payload, out);
}
}
gst_buffer_unref (buffer);
return ret;
/* ERRORS */
unknown_sampling:
{
GST_ELEMENT_ERROR (payload, STREAM, FORMAT,
(NULL), ("unimplemented sampling"));
gst_buffer_unref (buffer);
return GST_FLOW_NOT_SUPPORTED;
}
}
static GstFlowReturn
gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpAMRPay *rtpamrpay;
GstFlowReturn ret;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data, *payload_amr;
GstClockTime timestamp, duration;
guint packet_len, mtu;
gint i, num_packets, num_nonempty_packets;
gint amr_len;
gint *frame_size;
rtpamrpay = GST_RTP_AMR_PAY (basepayload);
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpamrpay);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
/* setup frame size pointer */
if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
frame_size = nb_frame_size;
else
frame_size = wb_frame_size;
GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
/* FIXME, only
* octet aligned, no interleaving, single channel, no CRC,
* no robust-sorting. To fix this you need to implement the downstream
* negotiation function. */
/* first count number of packets and total amr frame size */
amr_len = num_packets = num_nonempty_packets = 0;
for (i = 0; i < size; i++) {
guint8 FT;
gint fr_size;
FT = (data[i] & 0x78) >> 3;
fr_size = frame_size[FT];
GST_DEBUG_OBJECT (basepayload, "frame size %d", fr_size);
/* FIXME, we don't handle this yet.. */
if (fr_size <= 0)
goto wrong_size;
amr_len += fr_size;
num_nonempty_packets++;
num_packets++;
i += fr_size;
}
if (amr_len > size)
goto incomplete_frame;
/* we need one extra byte for the CMR, the ToC is in the input
* data */
payload_len = size + 1;
/* get packet len to check against MTU */
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
if (packet_len > mtu)
goto too_big;
/* now alloc output buffer */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy timestamp, or fabricate one */
if (timestamp != GST_CLOCK_TIME_NONE)
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
else {
/* AMR (nb) and AMR-WB both have 20 ms per frame */
/* FIXME: when we do more than one AMR frame per packet, fix this */
gint count = basepayload->seqnum - basepayload->seqnum_base;
GST_BUFFER_TIMESTAMP (outbuf) = count * 20 * GST_MSECOND;
}
if (duration != GST_CLOCK_TIME_NONE)
GST_BUFFER_DURATION (outbuf) = duration;
else {
GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
}
/* get payload, this is now writable */
payload = gst_rtp_buffer_get_payload (outbuf);
/* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* | CMR |R|R|R|R|
* +-+-+-+-+-+-+-+-+
*/
payload[0] = 0xF0; /* CMR, no specific mode requested */
/* this is where we copy the AMR data, after num_packets FTs and the
* CMR. */
payload_amr = payload + num_packets + 1;
/* copy data in payload, first we copy all the FTs then all
* the AMR data. The last FT has to have the F flag cleared. */
for (i = 1; i <= num_packets; i++) {
guint8 FT;
gint fr_size;
/* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* |F| FT |Q|P|P| more FT...
* +-+-+-+-+-+-+-+-+
*/
FT = (*data & 0x78) >> 3;
fr_size = frame_size[FT];
if (i == num_packets)
/* last packet, clear F flag */
payload[i] = *data & 0x7f;
else
/* set F flag */
payload[i] = *data | 0x80;
memcpy (payload_amr, &data[1], fr_size);
/* all sizes are > 0 since we checked for that above */
data += fr_size + 1;
payload_amr += fr_size;
}
gst_buffer_unref (buffer);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
/* ERRORS */
wrong_size:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received AMR frame with size <= 0"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
incomplete_frame:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received incomplete AMR frames"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
too_big:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received too many AMR frames for MTU"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
guint16 frag_offset;
/* 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;
frag_offset = 0;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
/* this will be the total lenght 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_BASE_RTP_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 (payload_len, 0, 0);
payload_len -= 4;
gst_rtp_buffer_set_payload_type (outbuf, 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 (outbuf);
payload[0] = 0;
payload[1] = 0;
payload[2] = frag_offset >> 8;
payload[3] = frag_offset & 0xff;
gst_adapter_copy (rtpmpapay->adapter, &payload[4], 0, payload_len);
gst_adapter_flush (rtpmpapay->adapter, payload_len);
avail -= payload_len;
frag_offset += payload_len;
if (avail == 0)
gst_rtp_buffer_set_marker (outbuf, TRUE);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpapay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpapay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_h263p_pay_flush (GstRtpH263PPay * rtph263ppay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
gboolean fragmented;
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;
guint payload_len;
gint header_len;
guint next_gop = 0;
gboolean found_gob = FALSE;
if (rtph263ppay->fragmentation_mode == GST_FRAGMENTATION_MODE_SYNC) {
/* start after 1st gop possible */
guint parsed_len = 3;
const guint8 *parse_data = NULL;
parse_data = gst_adapter_peek (rtph263ppay->adapter, avail);
/* Check if we have a gob or eos , eossbs */
/* FIXME EOS and EOSSBS packets should never contain any gobs and vice-versa */
if (avail >= 3 && *parse_data == 0 && *(parse_data + 1) == 0
&& *(parse_data + 2) >= 0x80) {
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 */
while (parsed_len + 2 < avail) {
if (parse_data[parsed_len] == 0 && parse_data[parsed_len + 1] == 0
&& parse_data[parsed_len + 2] >= 0x80) {
next_gop = parsed_len;
GST_DEBUG_OBJECT (rtph263ppay, " Next GOB Detected at : %d",
next_gop);
break;
}
parsed_len++;
}
}
/* for picture start frames (non-fragmented), we need to remove the first
* two 0x00 bytes and set P=1 */
header_len = (fragmented && !found_gob) ? 2 : 0;
towrite = MIN (avail, gst_rtp_buffer_calc_payload_len
(GST_BASE_RTP_PAYLOAD_MTU (rtph263ppay) - header_len, 0, 0));
if (next_gop > 0)
towrite = MIN (next_gop, towrite);
payload_len = header_len + towrite;
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* last fragment gets the marker bit set */
gst_rtp_buffer_set_marker (outbuf, avail > towrite ? 0 : 1);
payload = gst_rtp_buffer_get_payload (outbuf);
gst_adapter_copy (rtph263ppay->adapter, &payload[header_len], 0, towrite);
/* 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_TIMESTAMP (outbuf) = rtph263ppay->first_timestamp;
GST_BUFFER_DURATION (outbuf) = rtph263ppay->first_duration;
gst_adapter_flush (rtph263ppay->adapter, towrite);
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtph263ppay), outbuf);
avail -= towrite;
fragmented = TRUE;
}
return ret;
}
/**
* gst_base_rtp_audio_payload_flush:
* @baseaudiopayload: a #GstBaseRTPPayload
* @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
*
* Since: 0.10.25
*/
GstFlowReturn
gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
guint payload_len, GstClockTime timestamp)
{
GstBaseRTPPayload *basepayload;
GstBaseRTPAudioPayloadPrivate *priv;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
GstAdapter *adapter;
guint64 distance;
priv = baseaudiopayload->priv;
adapter = priv->adapter;
basepayload = GST_BASE_RTP_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_timestamp (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_base_rtp_audio_payload_push_buffer (baseaudiopayload, buffer);
} else {
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
gst_adapter_copy (adapter, payload, 0, payload_len);
gst_adapter_flush (adapter, payload_len);
/* set metadata */
gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
ret = gst_basertppayload_push (basepayload, outbuf);
}
return ret;
}
static GstFlowReturn
gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload *
baseaudiopayload, GstBuffer * buffer)
{
GstBaseRTPPayload *basepayload;
GstBaseRTPAudioPayloadPrivate *priv;
GstBuffer *outbuf;
GstClockTime timestamp;
guint8 *payload;
guint payload_len;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
payload_len = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
if (priv->buffer_list) {
/* create just the RTP header buffer */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
} else {
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
}
/* set metadata */
gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
if (priv->buffer_list) {
GstBufferList *list;
GstBufferListIterator *it;
list = gst_buffer_list_new ();
it = gst_buffer_list_iterate (list);
/* add both buffers to the buffer list */
gst_buffer_list_iterator_add_group (it);
gst_buffer_list_iterator_add (it, outbuf);
gst_buffer_list_iterator_add (it, buffer);
gst_buffer_list_iterator_free (it);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
ret = gst_basertppayload_push_list (basepayload, list);
} else {
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
memcpy (payload, GST_BUFFER_DATA (buffer), payload_len);
gst_buffer_unref (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
ret = gst_basertppayload_push (basepayload, 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_BASE_RTP_PAYLOAD_MTU (rtpmp4gpay);
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
/* 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 (payload_len + 4, 0, 0);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
* |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 */
/* copy stuff from adapter to payload */
gst_adapter_copy (rtpmp4gpay->adapter, &payload[4], 0, payload_len);
gst_adapter_flush (rtpmp4gpay->adapter, payload_len);
/* marker only if the packet is complete */
gst_rtp_buffer_set_marker (outbuf, avail <= payload_len);
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;
}
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf);
avail -= payload_len;
}
return ret;
}
static GstFlowReturn
gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpSPEEXPay *rtpspeexpay;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp, duration;
GstFlowReturn ret;
rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
switch (rtpspeexpay->packet) {
case 0:
/* ident packet. We need to parse the headers to construct the RTP
* properties. */
if (!gst_rtp_speex_pay_parse_ident (rtpspeexpay, data, size))
goto parse_error;
ret = GST_FLOW_OK;
goto done;
case 1:
/* comment packet, we ignore it */
ret = GST_FLOW_OK;
goto done;
default:
/* other packets go in the payload */
break;
}
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP)) {
ret = GST_FLOW_OK;
goto done;
}
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
/* FIXME, only one SPEEX frame per RTP packet for now */
payload_len = size;
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* FIXME, assert for now */
g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpspeexpay));
/* copy timestamp and duration */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = duration;
/* get payload */
payload = gst_rtp_buffer_get_payload (outbuf);
/* copy data in payload */
memcpy (&payload[0], data, size);
ret = gst_basertppayload_push (basepayload, outbuf);
done:
gst_buffer_unref (buffer);
rtpspeexpay->packet++;
return ret;
/* ERRORS */
parse_error:
{
GST_ELEMENT_ERROR (rtpspeexpay, STREAM, DECODE, (NULL),
("Error parsing first identification packet."));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
/* Get a DV frame, chop it up in pieces, and push the pieces to the RTP layer.
*/
static GstFlowReturn
gst_rtp_dv_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRTPDVPay *rtpdvpay;
guint max_payload_size;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
gint hdrlen;
guint size;
guint8 *data;
guint8 *dest;
guint filled;
rtpdvpay = GST_RTP_DV_PAY (basepayload);
hdrlen = gst_rtp_buffer_calc_header_len (0);
/* DV frames are made up from a bunch of DIF blocks. DIF blocks are 80 bytes
* each, and we should put an integral number of them in each RTP packet.
* Therefore, we round the available room down to the nearest multiple of 80.
*
* The available room is just the packet MTU, minus the RTP header length. */
max_payload_size = ((GST_BASE_RTP_PAYLOAD_MTU (rtpdvpay) - hdrlen) / 80) * 80;
/* The length of the buffer to transmit. */
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
GST_DEBUG_OBJECT (rtpdvpay,
"DV RTP payloader got buffer of %u bytes, splitting in %u byte "
"payload fragments, at time %" GST_TIME_FORMAT, size, max_payload_size,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
if (!rtpdvpay->negotiated) {
gst_dv_pay_negotiate (rtpdvpay, data, size);
/* if we have not yet scanned the stream for its type, do so now */
rtpdvpay->negotiated = TRUE;
}
outbuf = NULL;
dest = NULL;
filled = 0;
/* while we have a complete DIF chunks left */
while (size >= 80) {
/* Allocate a new buffer, set the timestamp */
if (outbuf == NULL) {
outbuf = gst_rtp_buffer_new_allocate (max_payload_size, 0, 0);
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buffer);
dest = gst_rtp_buffer_get_payload (outbuf);
filled = 0;
}
/* inspect the DIF chunk, if we don't need to include it, skip to the next one. */
if (include_dif (rtpdvpay, data)) {
/* copy data in packet */
memcpy (dest, data, 80);
dest += 80;
filled += 80;
}
/* go to next dif chunk */
size -= 80;
data += 80;
/* push out the buffer if the next one would exceed the max packet size or
* when we are pushing the last packet */
if (filled + 80 > max_payload_size || size < 80) {
if (size < 160) {
guint hlen;
/* set marker */
gst_rtp_buffer_set_marker (outbuf, TRUE);
/* shrink buffer to last packet */
hlen = gst_rtp_buffer_get_header_len (outbuf);
gst_rtp_buffer_set_packet_len (outbuf, hlen + filled);
}
/* Push out the created piece, and check for errors. */
ret = gst_basertppayload_push (basepayload, outbuf);
if (ret != GST_FLOW_OK)
break;
outbuf = NULL;
}
}
gst_buffer_unref (buffer);
return ret;
}
