static GstBuffer *
gst_rtp_mp4v_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpMP4VDepay *rtpmp4vdepay;
GstBuffer *outbuf;
rtpmp4vdepay = GST_RTP_MP4V_DEPAY (depayload);
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf))
gst_adapter_clear (rtpmp4vdepay->adapter);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
gst_adapter_push (rtpmp4vdepay->adapter, outbuf);
/* if this was the last packet of the VOP, create and push a buffer */
if (gst_rtp_buffer_get_marker (buf)) {
guint avail;
avail = gst_adapter_available (rtpmp4vdepay->adapter);
outbuf = gst_adapter_take_buffer (rtpmp4vdepay->adapter, avail);
GST_DEBUG ("gst_rtp_mp4v_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
}
return NULL;
}
static GstBuffer *
gst_rtp_pcma_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf = NULL;
gboolean marker;
guint len;
marker = gst_rtp_buffer_get_marker (buf);
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
GST_BUFFER_SIZE (buf), marker,
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
len = gst_rtp_buffer_get_payload_len (buf);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
if (outbuf) {
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (len, GST_SECOND, depayload->clock_rate);
if (marker) {
/* mark start of talkspurt with DISCONT */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
}
return outbuf;
}
static GstBuffer *
gst_rtp_mp4v_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpMP4VDepay *rtpmp4vdepay;
GstBuffer *pbuf, *outbuf = NULL;
GstRTPBuffer rtp = { NULL };
gboolean marker;
rtpmp4vdepay = GST_RTP_MP4V_DEPAY (depayload);
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf))
gst_adapter_clear (rtpmp4vdepay->adapter);
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
pbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
marker = gst_rtp_buffer_get_marker (&rtp);
gst_rtp_buffer_unmap (&rtp);
gst_adapter_push (rtpmp4vdepay->adapter, pbuf);
/* if this was the last packet of the VOP, create and push a buffer */
if (marker) {
guint avail;
avail = gst_adapter_available (rtpmp4vdepay->adapter);
outbuf = gst_adapter_take_buffer (rtpmp4vdepay->adapter, avail);
GST_DEBUG ("gst_rtp_mp4v_depay_chain: pushing buffer of size %"
G_GSIZE_FORMAT, gst_buffer_get_size (outbuf));
}
return outbuf;
}
static GstBuffer *
gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpG722Depay *rtpg722depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
rtpg722depay = GST_RTP_G722_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (buf);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
marker = gst_rtp_buffer_get_marker (buf);
if (marker) {
/* mark talk spurt with DISCONT */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
("Empty Payload."), (NULL));
return NULL;
}
}
static GstBuffer *
gst_rtp_pcmu_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf = NULL;
guint len;
gboolean marker;
GstRTPBuffer rtp = { NULL };
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
marker = gst_rtp_buffer_get_marker (&rtp);
GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
gst_buffer_get_size (buf), marker,
gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp));
len = gst_rtp_buffer_get_payload_len (&rtp);
outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
gst_rtp_buffer_unmap (&rtp);
if (outbuf) {
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (len, GST_SECOND, depayload->clock_rate);
if (marker) {
/* mark start of talkspurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
}
return outbuf;
}
static GstBuffer *
gst_rtp_ilbc_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf;
gboolean marker;
GstRTPBuffer rtp = { NULL };
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
marker = gst_rtp_buffer_get_marker (&rtp);
GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
gst_buffer_get_size (buf), marker,
gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp));
outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
gst_rtp_buffer_unmap (&rtp);
if (marker && outbuf) {
/* mark start of talkspurt with DISCONT */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
return outbuf;
}
static GstBuffer *
gst_rtp_gsm_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstBuffer *outbuf = NULL;
gboolean marker;
marker = gst_rtp_buffer_get_marker (rtp);
GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
gst_buffer_get_size (rtp->buffer), marker,
gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
if (marker && outbuf) {
/* mark start of talkspurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
if (outbuf) {
gst_rtp_drop_meta (GST_ELEMENT_CAST (depayload), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
}
return outbuf;
}
static GstBuffer *
gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpL16Depay *rtpL16depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
GstRTPBuffer rtp = { NULL };
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
payload_len = gst_rtp_buffer_get_payload_len (&rtp);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
marker = gst_rtp_buffer_get_marker (&rtp);
if (marker) {
/* mark talk spurt with DISCONT */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
outbuf = gst_buffer_make_writable (outbuf);
if (rtpL16depay->order &&
!gst_audio_buffer_reorder_channels (outbuf,
rtpL16depay->info.finfo->format, rtpL16depay->info.channels,
rtpL16depay->info.position, rtpL16depay->order->pos)) {
goto reorder_failed;
}
gst_rtp_buffer_unmap (&rtp);
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
("Empty Payload."), (NULL));
gst_rtp_buffer_unmap (&rtp);
return NULL;
}
reorder_failed:
{
GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
("Channel reordering failed."), (NULL));
gst_rtp_buffer_unmap (&rtp);
return NULL;
}
}
static GstBuffer *
gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf;
GstRTPBuffer rtpbuf = { NULL, };
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf);
outbuf = gst_rtp_buffer_get_payload_buffer (&rtpbuf);
gst_rtp_buffer_unmap (&rtpbuf);
return outbuf;
}
static GstBuffer *
gst_rtp_mp1s_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf;
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
if (outbuf)
GST_DEBUG ("gst_rtp_mp1s_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
}
static GstBuffer *
gst_rtp_g729_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpG729Depay *rtpg729depay;
GstBuffer *outbuf = NULL;
gint payload_len;
gboolean marker;
GstRTPBuffer rtp = { NULL };
rtpg729depay = GST_RTP_G729_DEPAY (depayload);
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
payload_len = gst_rtp_buffer_get_payload_len (&rtp);
/* At least 2 bytes (CNG from G729 Annex B) */
if (payload_len < 2) {
GST_ELEMENT_WARNING (rtpg729depay, STREAM, DECODE,
(NULL), ("G729 RTP payload too small (%d)", payload_len));
goto bad_packet;
}
GST_LOG_OBJECT (rtpg729depay, "payload len %d", payload_len);
if ((payload_len % 10) == 2) {
GST_LOG_OBJECT (rtpg729depay, "G729 payload contains CNG frame");
}
outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
marker = gst_rtp_buffer_get_marker (&rtp);
gst_rtp_buffer_unmap (&rtp);
if (marker) {
/* marker bit starts talkspurt */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
GST_LOG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (outbuf));
return outbuf;
/* ERRORS */
bad_packet:
{
/* no fatal error */
return NULL;
}
}
static GstBuffer *
gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf;
GstRTPBuffer rtpbuf = { NULL, };
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf);
outbuf = gst_rtp_buffer_get_payload_buffer (&rtpbuf);
gst_rtp_buffer_unmap (&rtpbuf);
outbuf = gst_buffer_make_writable (outbuf);
/* Filter away all metas that are not sensible to copy */
gst_buffer_foreach_meta (outbuf, foreach_metadata, depayload);
return outbuf;
}
static GstBuffer *
gst_rtp_g722_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpG722Depay *rtpg722depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
GstRTPBuffer rtp = { NULL };
rtpg722depay = GST_RTP_G722_DEPAY (depayload);
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
payload_len = gst_rtp_buffer_get_payload_len (&rtp);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
marker = gst_rtp_buffer_get_marker (&rtp);
gst_rtp_buffer_unmap (&rtp);
if (marker && outbuf) {
/* mark talk spurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
if (outbuf) {
gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpg722depay), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
}
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
("Empty Payload."), (NULL));
gst_rtp_buffer_unmap (&rtp);
return NULL;
}
}
static GstBuffer *
gst_rtp_g723_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpG723Depay *rtpg723depay;
GstBuffer *outbuf = NULL;
gint payload_len;
gboolean marker;
rtpg723depay = GST_RTP_G723_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (buf);
/* At least 4 bytes */
if (payload_len < 4)
goto too_small;
GST_LOG_OBJECT (rtpg723depay, "payload len %d", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
marker = gst_rtp_buffer_get_marker (buf);
if (marker) {
/* marker bit starts talkspurt */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
GST_LOG_OBJECT (depayload, "pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
/* ERRORS */
too_small:
{
GST_ELEMENT_WARNING (rtpg723depay, STREAM, DECODE,
(NULL), ("G723 RTP payload too small (%d)", payload_len));
goto bad_packet;
}
bad_packet:
{
/* no fatal error */
return NULL;
}
}
static GstBuffer *
gst_rtp_ilbc_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf;
gboolean marker;
marker = gst_rtp_buffer_get_marker (buf);
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
GST_BUFFER_SIZE (buf), marker,
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
if (marker && outbuf) {
/* mark start of talkspurt with DISCONT */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
return outbuf;
}
static GstBuffer *
gst_rtp_ilbc_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstBuffer *outbuf;
gboolean marker;
marker = gst_rtp_buffer_get_marker (rtp);
GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
gst_buffer_get_size (rtp->buffer), marker,
gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
if (marker && outbuf) {
/* mark start of talkspurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
return outbuf;
}
static GstBuffer *
gst_rtp_speex_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf = NULL;
GstRTPBuffer rtp = { NULL };
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
gst_buffer_get_size (buf),
gst_rtp_buffer_get_marker (&rtp),
gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp));
/* nothing special to be done */
outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
gst_rtp_buffer_unmap (&rtp);
if (outbuf)
GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
return outbuf;
}
/* Process one RTP packet. Accumulate RTP payload in the proper place in a DV
* frame, and return that frame if we detect a new frame, or NULL otherwise.
* We assume a DV frame is 144000 bytes. That should accomodate PAL as well as
* NTSC.
*/
static GstBuffer *
gst_rtp_dv_depay_process (GstRTPBaseDepayload * base, GstRTPBuffer * rtp)
{
GstBuffer *out = NULL;
GstBuffer *payload_buf;
guint8 *payload;
guint32 rtp_ts;
guint payload_len, location;
GstRTPDVDepay *dvdepay = GST_RTP_DV_DEPAY (base);
gboolean marker;
GstMapInfo map;
marker = gst_rtp_buffer_get_marker (rtp);
/* Check if the received packet contains (the start of) a new frame, we do
* this by checking the RTP timestamp. */
rtp_ts = gst_rtp_buffer_get_timestamp (rtp);
/* we cannot copy the packet yet if the marker is set, we will do that below
* after taking out the data */
if (dvdepay->prev_ts != -1 && rtp_ts != dvdepay->prev_ts && !marker) {
/* the timestamp changed */
GST_DEBUG_OBJECT (dvdepay, "new frame with ts %u, old ts %u", rtp_ts,
dvdepay->prev_ts);
/* return copy of accumulator. */
out = gst_buffer_copy (dvdepay->acc);
gst_buffer_foreach_meta (dvdepay->acc, foreach_metadata_drop, NULL);
}
/* Extract the payload */
payload_len = gst_rtp_buffer_get_payload_len (rtp);
payload = gst_rtp_buffer_get_payload (rtp);
payload_buf = gst_rtp_buffer_get_payload_buffer (rtp);
/* copy all DIF chunks in their place. */
gst_buffer_map (dvdepay->acc, &map, GST_MAP_READWRITE);
while (payload_len >= 80) {
guint offset;
/* Calculate where in the frame the payload should go */
location = calculate_difblock_location (payload);
if (location < 6) {
/* part of a header, set the flag to mark that we have the header. */
dvdepay->header_mask |= (1 << location);
GST_LOG_OBJECT (dvdepay, "got header at location %d, now %02x", location,
dvdepay->header_mask);
} else {
GST_LOG_OBJECT (dvdepay, "got block at location %d", location);
}
if (location != -1) {
/* get the byte offset of the dif block */
offset = location * 80;
/* And copy it in, provided the location is sane. */
if (offset <= dvdepay->frame_size - 80) {
memcpy (map.data + offset, payload, 80);
gst_rtp_copy_meta (GST_ELEMENT_CAST (dvdepay), dvdepay->acc,
payload_buf, 0);
}
}
payload += 80;
payload_len -= 80;
}
gst_buffer_unmap (dvdepay->acc, &map);
gst_buffer_unref (payload_buf);
if (marker) {
GST_DEBUG_OBJECT (dvdepay, "marker bit complete frame %u", rtp_ts);
/* only copy the frame when we have a complete header */
if (dvdepay->header_mask == 0x3f) {
/* The marker marks the end of a frame that we need to push. The next frame
* will change the timestamp but we won't copy the accumulator again because
* we set the prev_ts to -1. */
out = gst_buffer_copy (dvdepay->acc);
gst_buffer_foreach_meta (dvdepay->acc, foreach_metadata_drop, NULL);
} else {
GST_WARNING_OBJECT (dvdepay, "waiting for frame headers %02x",
dvdepay->header_mask);
}
dvdepay->prev_ts = -1;
} else {
/* save last timestamp */
dvdepay->prev_ts = rtp_ts;
}
return out;
}
static GstBuffer *
gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpL16Depay *rtpL16depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
GstAudioInfo *info;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (rtp);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
marker = gst_rtp_buffer_get_marker (rtp);
if (marker) {
/* mark talk spurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
outbuf = gst_buffer_make_writable (outbuf);
info = &rtpL16depay->info;
if (payload_len % ((info->finfo->width * info->channels) / 8) != 0)
goto wrong_payload_size;
if (rtpL16depay->order &&
!gst_audio_buffer_reorder_channels (outbuf,
info->finfo->format, info->channels,
info->position, rtpL16depay->order->pos)) {
goto reorder_failed;
}
gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpL16depay), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
("Empty Payload."), (NULL));
return NULL;
}
wrong_payload_size:
{
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
("Wrong Payload Size."), (NULL));
return NULL;
}
reorder_failed:
{
GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
("Channel reordering failed."), (NULL));
return NULL;
}
}
