static GstBufferList *
create_buffer_list (void)
{
guint len;
GstBuffer *buffer;
mylist = gst_buffer_list_new ();
fail_if (mylist == NULL);
mycaps = gst_caps_from_string ("application/x-gst-check");
fail_if (mycaps == NULL);
len = gst_buffer_list_length (mylist);
fail_if (len != 0);
buffer = gst_buffer_new_and_alloc (sizeof (gint));
gst_buffer_fill (buffer, 0, &values[0], sizeof (gint));
gst_buffer_list_add (mylist, buffer);
buffer = gst_buffer_new_and_alloc (sizeof (gint));
gst_buffer_fill (buffer, 0, &values[1], sizeof (gint));
gst_buffer_list_add (mylist, buffer);
buffer = gst_buffer_new_and_alloc (sizeof (gint));
gst_buffer_fill (buffer, 0, &values[2], sizeof (gint));
gst_buffer_list_add (mylist, buffer);
gst_pad_set_caps (mysrcpad, mycaps);
gst_caps_unref (mycaps);
return mylist;
}
static GstFlowReturn
gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf;
guint payload_len;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
payload_len = gst_buffer_get_size (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
/* create just the RTP header buffer */
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
/* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
if (priv->buffer_list) {
GstBufferList *list;
guint i, len;
list = gst_buffer_list_new ();
len = gst_buffer_list_length (list);
for (i = 0; i < len; i++) {
/* FIXME */
g_warning ("bufferlist not implemented");
gst_buffer_list_add (list, outbuf);
gst_buffer_list_add (list, buffer);
}
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
ret = gst_rtp_base_payload_push_list (basepayload, list);
} else {
CopyMetaData data;
/* copy payload */
data.pay = baseaudiopayload;
data.outbuf = outbuf;
gst_buffer_foreach_meta (buffer, foreach_metadata, &data);
outbuf = gst_buffer_append (outbuf, buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
}
return ret;
}
static gboolean
new_packet_cb (guint8 * data, guint len, void *user_data)
{
/* Called when the PsMux has prepared a packet for output. Return FALSE
* on error */
MpegPsMux *mux = (MpegPsMux *) user_data;
GstBuffer *buf;
GstFlowReturn ret;
GST_LOG_OBJECT (mux, "Outputting a packet of length %d", len);
data = g_memdup (data, len);
buf = gst_buffer_new_wrapped (data, len);
GST_BUFFER_TIMESTAMP (buf) = mux->last_ts;
if (mux->aggregate_gops) {
if (mux->gop_list == NULL)
mux->gop_list = gst_buffer_list_new ();
gst_buffer_list_add (mux->gop_list, buf);
return TRUE;
}
ret = gst_pad_push (mux->srcpad, buf);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
mux->last_flow_ret = ret;
return FALSE;
}
return TRUE;
}
/**
* gst_adapter_take_buffer_list:
* @adapter: a #GstAdapter
* @nbytes: the number of bytes to take
*
* Returns a #GstBufferList of buffers containing the first @nbytes bytes of
* the @adapter. The returned bytes will be flushed from the adapter.
* When the caller can deal with individual buffers, this function is more
* performant because no memory should be copied.
*
* Caller owns the returned list. Call gst_buffer_list_unref() to free
* the list after usage.
*
* Returns: (transfer full) (nullable): a #GstBufferList of buffers containing
* the first @nbytes of the adapter, or %NULL if @nbytes bytes are not
* available
*
* Since: 1.6
*/
GstBufferList *
gst_adapter_take_buffer_list (GstAdapter * adapter, gsize nbytes)
{
GstBufferList *buffer_list;
GstBuffer *cur;
gsize hsize, skip, cur_size;
guint n_bufs;
g_return_val_if_fail (GST_IS_ADAPTER (adapter), NULL);
if (nbytes > adapter->size)
return NULL;
GST_LOG_OBJECT (adapter, "taking %" G_GSIZE_FORMAT " bytes", nbytes);
/* try to create buffer list with sufficient size, so no resize is done later */
if (adapter->count < 64)
n_bufs = adapter->count;
else
n_bufs = (adapter->count * nbytes * 1.2 / adapter->size) + 1;
buffer_list = gst_buffer_list_new_sized (n_bufs);
while (nbytes > 0) {
cur = adapter->buflist->data;
skip = adapter->skip;
cur_size = gst_buffer_get_size (cur);
hsize = MIN (nbytes, cur_size - skip);
gst_buffer_list_add (buffer_list, gst_adapter_take_buffer (adapter, hsize));
nbytes -= hsize;
}
return buffer_list;
}
static GstFlowReturn
rtp_pipeline_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstBufferList *list;
list = gst_buffer_list_new_sized (1);
gst_buffer_list_add (list, buf);
return rtp_pipeline_chain_list (pad, parent, list);
}
static GstBufferList *
create_buffer_list (guint * data_size)
{
GstBufferList *list;
GstBuffer *rtp_buffer;
GstBuffer *data_buffer;
list = gst_buffer_list_new ();
/*** First group, i.e. first packet. **/
/* Create the RTP header buffer */
rtp_buffer = gst_buffer_new_allocate (NULL, RTP_HEADER_SIZE, NULL);
gst_buffer_memset (rtp_buffer, 0, 0, RTP_HEADER_SIZE);
/* Create the buffer that holds the payload */
data_buffer = gst_buffer_new_allocate (NULL, RTP_PAYLOAD_SIZE, NULL);
gst_buffer_memset (data_buffer, 0, 0, RTP_PAYLOAD_SIZE);
/* Create a new group to hold the rtp header and the payload */
gst_buffer_list_add (list, gst_buffer_append (rtp_buffer, data_buffer));
/*** Second group, i.e. second packet. ***/
/* Create the RTP header buffer */
rtp_buffer = gst_buffer_new_allocate (NULL, RTP_HEADER_SIZE, NULL);
gst_buffer_memset (rtp_buffer, 0, 0, RTP_HEADER_SIZE);
/* Create the buffer that holds the payload */
data_buffer = gst_buffer_new_allocate (NULL, RTP_PAYLOAD_SIZE, NULL);
gst_buffer_memset (data_buffer, 0, 0, RTP_PAYLOAD_SIZE);
/* Create a new group to hold the rtp header and the payload */
gst_buffer_list_add (list, gst_buffer_append (rtp_buffer, data_buffer));
/* Calculate the size of the data */
*data_size = 2 * RTP_HEADER_SIZE + 2 * RTP_PAYLOAD_SIZE;
return list;
}
GstFlowReturn AudioFileReader::handleSample(GstAppSink* sink)
{
GstSample* sample = gst_app_sink_pull_sample(sink);
if (!sample)
return GST_FLOW_ERROR;
GstBuffer* buffer = gst_sample_get_buffer(sample);
if (!buffer) {
gst_sample_unref(sample);
return GST_FLOW_ERROR;
}
GstCaps* caps = gst_sample_get_caps(sample);
if (!caps) {
gst_sample_unref(sample);
return GST_FLOW_ERROR;
}
GstAudioInfo info;
gst_audio_info_from_caps(&info, caps);
int frames = gst_buffer_get_size(buffer) / info.bpf;
// Check the first audio channel. The buffer is supposed to store
// data of a single channel anyway.
switch (GST_AUDIO_INFO_POSITION(&info, 0)) {
case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
case GST_AUDIO_CHANNEL_POSITION_MONO:
gst_buffer_list_add(m_frontLeftBuffers, gst_buffer_ref(buffer));
m_channelSize += frames;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
gst_buffer_list_add(m_frontRightBuffers, gst_buffer_ref(buffer));
break;
default:
break;
}
gst_sample_unref(sample);
return GST_FLOW_OK;
}
/**
* gst_adapter_get_buffer_list:
* @adapter: a #GstAdapter
* @nbytes: the number of bytes to get
*
* Returns a #GstBufferList of buffers containing the first @nbytes bytes of
* the @adapter but does not flush them from the adapter. See
* gst_adapter_take_buffer_list() for details.
*
* Caller owns the returned list. Call gst_buffer_list_unref() to free
* the list after usage.
*
* Returns: (transfer full) (nullable): a #GstBufferList of buffers containing
* the first @nbytes of the adapter, or %NULL if @nbytes bytes are not
* available
*
* Since: 1.6
*/
GstBufferList *
gst_adapter_get_buffer_list (GstAdapter * adapter, gsize nbytes)
{
GstBufferList *buffer_list;
GstBuffer *cur, *buffer;
gsize hsize, skip, cur_size;
guint n_bufs;
GSList *g = NULL;
g_return_val_if_fail (GST_IS_ADAPTER (adapter), NULL);
if (nbytes > adapter->size)
return NULL;
GST_LOG_OBJECT (adapter, "getting %" G_GSIZE_FORMAT " bytes", nbytes);
/* try to create buffer list with sufficient size, so no resize is done later */
if (adapter->count < 64)
n_bufs = adapter->count;
else
n_bufs = (adapter->count * nbytes * 1.2 / adapter->size) + 1;
buffer_list = gst_buffer_list_new_sized (n_bufs);
g = adapter->buflist;
skip = adapter->skip;
while (nbytes > 0) {
cur = g->data;
cur_size = gst_buffer_get_size (cur);
hsize = MIN (nbytes, cur_size - skip);
if (skip == 0 && cur_size == hsize) {
GST_LOG_OBJECT (adapter,
"inserting a buffer of %" G_GSIZE_FORMAT " bytes", hsize);
buffer = gst_buffer_ref (cur);
} else {
GST_LOG_OBJECT (adapter, "inserting a buffer of %" G_GSIZE_FORMAT " bytes"
" via region copy", hsize);
buffer = gst_buffer_copy_region (cur, GST_BUFFER_COPY_ALL, skip, hsize);
}
gst_buffer_list_add (buffer_list, buffer);
nbytes -= hsize;
skip = 0;
g = g_slist_next (g);
}
return buffer_list;
}
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;
}
static GstFlowReturn
gst_rtp_vraw_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
{
GstRtpVRawPay *rtpvrawpay;
GstFlowReturn ret = GST_FLOW_OK;
gfloat packets_per_packline;
guint pgroups_per_packet;
guint packlines_per_list, buffers_per_list;
guint lines_delay; /* after how many packed lines we push out a buffer list */
guint last_line; /* last pack line number we pushed out a buffer list */
guint line, offset;
guint8 *p0, *yp, *up, *vp;
guint ystride, uvstride;
guint xinc, yinc;
guint pgroup;
guint mtu;
guint width, height;
gint field, fields;
GstVideoFormat format;
GstVideoFrame frame;
gint interlaced;
gboolean use_buffer_lists;
GstBufferList *list = NULL;
GstRTPBuffer rtp = { NULL, };
rtpvrawpay = GST_RTP_VRAW_PAY (payload);
gst_video_frame_map (&frame, &rtpvrawpay->vinfo, buffer, GST_MAP_READ);
GST_LOG_OBJECT (rtpvrawpay, "new frame of %" G_GSIZE_FORMAT " bytes",
gst_buffer_get_size (buffer));
/* get pointer and strides of the planes */
p0 = GST_VIDEO_FRAME_PLANE_DATA (&frame, 0);
yp = GST_VIDEO_FRAME_COMP_DATA (&frame, 0);
up = GST_VIDEO_FRAME_COMP_DATA (&frame, 1);
vp = GST_VIDEO_FRAME_COMP_DATA (&frame, 2);
ystride = GST_VIDEO_FRAME_COMP_STRIDE (&frame, 0);
uvstride = GST_VIDEO_FRAME_COMP_STRIDE (&frame, 1);
mtu = GST_RTP_BASE_PAYLOAD_MTU (payload);
/* amount of bytes for one pixel */
pgroup = rtpvrawpay->pgroup;
width = GST_VIDEO_INFO_WIDTH (&rtpvrawpay->vinfo);
height = GST_VIDEO_INFO_HEIGHT (&rtpvrawpay->vinfo);
interlaced = GST_VIDEO_INFO_IS_INTERLACED (&rtpvrawpay->vinfo);
format = GST_VIDEO_INFO_FORMAT (&rtpvrawpay->vinfo);
yinc = rtpvrawpay->yinc;
xinc = rtpvrawpay->xinc;
/* after how many packed lines we push out a buffer list */
lines_delay = GST_ROUND_UP_4 (height / rtpvrawpay->chunks_per_frame);
/* calculate how many buffers we expect to store in a single buffer list */
pgroups_per_packet = (mtu - (12 + 14)) / pgroup;
packets_per_packline = width / (xinc * pgroups_per_packet * 1.0);
packlines_per_list = height / (yinc * rtpvrawpay->chunks_per_frame);
buffers_per_list = packlines_per_list * packets_per_packline;
buffers_per_list = GST_ROUND_UP_8 (buffers_per_list);
use_buffer_lists = (rtpvrawpay->chunks_per_frame < (height / yinc));
fields = 1 + interlaced;
/* start with line 0, offset 0 */
for (field = 0; field < fields; field++) {
line = field;
offset = 0;
last_line = 0;
if (use_buffer_lists)
list = gst_buffer_list_new_sized (buffers_per_list);
/* write all lines */
while (line < height) {
guint left, pack_line;
GstBuffer *out;
guint8 *outdata, *headers;
gboolean next_line, complete = FALSE;
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_PTS (out) = GST_BUFFER_PTS (buffer);
} else {
GST_BUFFER_PTS (out) = GST_BUFFER_PTS (buffer) +
GST_BUFFER_DURATION (buffer) / 2;
}
gst_rtp_buffer_map (out, GST_MAP_WRITE, &rtp);
outdata = gst_rtp_buffer_get_payload (&rtp);
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;
/* make sure we can fit at least *one* header and pixel */
if (!(left > (6 + pgroup))) {
gst_rtp_buffer_unmap (&rtp);
gst_buffer_unref (out);
goto too_small;
}
/* 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) / 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) * xinc;
length = (pixels * pgroup) / 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 += 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 (format) {
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:
case GST_VIDEO_FORMAT_UYVP:
offs /= xinc;
memcpy (outdata, p0 + (lin * ystride) + (offs * pgroup), length);
outdata += length;
break;
case GST_VIDEO_FORMAT_AYUV:
{
gint i;
guint8 *datap;
datap = p0 + (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 / yinc * uvstride) + (offs / 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 / yinc * uvstride) + (offs / 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_rtp_buffer_unmap (&rtp);
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 (&rtp, TRUE);
complete = TRUE;
}
gst_rtp_buffer_unmap (&rtp);
if (left > 0) {
GST_LOG_OBJECT (rtpvrawpay, "we have %u bytes left", left);
gst_buffer_resize (out, 0, gst_buffer_get_size (out) - left);
}
/* Now either push out the buffer directly */
if (!use_buffer_lists) {
ret = gst_rtp_base_payload_push (payload, out);
continue;
}
/* or add the buffer to buffer list ... */
gst_buffer_list_add (list, out);
/* .. and check if we need to push out the list */
pack_line = (line - field) / fields;
if (complete || (pack_line > last_line && pack_line % lines_delay == 0)) {
GST_LOG_OBJECT (rtpvrawpay, "pushing list of %u buffers up to pack "
"line %u", gst_buffer_list_length (list), pack_line);
ret = gst_rtp_base_payload_push_list (payload, list);
list = NULL;
if (!complete)
list = gst_buffer_list_new_sized (buffers_per_list);
last_line = pack_line;
}
}
}
gst_video_frame_unmap (&frame);
gst_buffer_unref (buffer);
return ret;
/* ERRORS */
unknown_sampling:
{
GST_ELEMENT_ERROR (payload, STREAM, FORMAT,
(NULL), ("unimplemented sampling"));
gst_video_frame_unmap (&frame);
gst_buffer_unref (buffer);
return GST_FLOW_NOT_SUPPORTED;
}
too_small:
{
GST_ELEMENT_ERROR (payload, RESOURCE, NO_SPACE_LEFT,
(NULL), ("not enough space to send at least one pixel"));
gst_video_frame_unmap (&frame);
gst_buffer_unref (buffer);
return GST_FLOW_NOT_SUPPORTED;
}
}
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_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_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf;
guint8 *payload;
guint payload_len;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
payload_len = gst_buffer_get_size (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_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
if (priv->buffer_list) {
GstBufferList *list;
guint i, len;
list = gst_buffer_list_new ();
len = gst_buffer_list_length (list);
for (i = 0; i < len; i++) {
/* FIXME */
g_warning ("bufferlist not implemented");
gst_buffer_list_add (list, outbuf);
gst_buffer_list_add (list, buffer);
}
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
ret = gst_rtp_base_payload_push_list (basepayload, list);
} else {
GstRTPBuffer rtp = { NULL };
/* copy payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
payload = gst_rtp_buffer_get_payload (&rtp);
gst_buffer_extract (buffer, 0, payload, payload_len);
gst_rtp_buffer_unmap (&rtp);
gst_buffer_unref (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
}
return ret;
}