Commit acddbd83 authored by Wim Taymans's avatar Wim Taymans

gst/rtp/gstrtpamrdepay.c: Fix depayloader clock_rate and some cleanups.

Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
(gst_rtp_amr_depay_process):
Fix depayloader clock_rate and some cleanups.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
* gst/rtp/gstrtph264depay.h:
Don't push codec_data in the adapter because it might get flushed when
we get a discont.
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
Handle multiple AU per packet.
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process),
(gst_rtp_sv3v_depay_plugin_init):
Disable rank, this one does not work.
Remove timestamping, base class does that.
parent 497d589d
2007-04-10 Wim Taymans <wim@fluendo.com>
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
(gst_rtp_amr_depay_process):
Fix depayloader clock_rate and some cleanups.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
* gst/rtp/gstrtph264depay.h:
Don't push codec_data in the adapter because it might get flushed when
we get a discont.
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
Handle multiple AU per packet.
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process),
(gst_rtp_sv3v_depay_plugin_init):
Disable rank, this one does not work.
Remove timestamping, base class does that.
2007-04-10 Stefan Kost <ensonic@users.sf.net>
* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
......
......@@ -197,6 +197,7 @@ gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = 8000;
depayload->clock_rate = clock_rate;
/* we require 1 channel, 8000 Hz, octet aligned, no CRC,
* no robust sorting, no interleaving for now */
......@@ -233,24 +234,20 @@ gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpAMRDepay *rtpamrdepay;
GstBuffer *outbuf = NULL;
gint payload_len;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
if (!rtpamrdepay->negotiated)
goto not_negotiated;
if (!gst_rtp_buffer_validate (buf)) {
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP packet did not validate"));
goto bad_packet;
}
if (!gst_rtp_buffer_validate (buf))
goto invalid_packet;
/* when we get here, 1 channel, 8000 Hz, octet aligned, no CRC,
* no robust sorting, no interleaving data is to be depayloaded */
{
gint payload_len;
guint8 *payload, *p, *dp;
guint32 timestamp;
guint8 CMR;
gint i, num_packets, num_nonempty_packets;
gint amr_len;
......@@ -259,11 +256,8 @@ gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
payload_len = gst_rtp_buffer_get_payload_len (buf);
/* need at least 2 bytes for the header */
if (payload_len < 2) {
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP payload too small (%d)", payload_len));
goto bad_packet;
}
if (payload_len < 2)
goto too_small;
payload = gst_rtp_buffer_get_payload (buf);
......@@ -290,11 +284,8 @@ gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
payload_len -= 1;
payload += 1;
if (ILP > ILL) {
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP wrong interleaving"));
goto bad_packet;
}
if (ILP > ILL)
goto wrong_interleaving;
}
/*
......@@ -317,11 +308,8 @@ gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
fr_size = frame_size[FT];
GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
if (fr_size == -1) {
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP frame size == -1"));
goto bad_packet;
}
if (fr_size == -1)
goto wrong_framesize;
if (fr_size > 0) {
amr_len += fr_size;
......@@ -335,26 +323,15 @@ gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
if (rtpamrdepay->crc) {
/* data len + CRC len + header bytes should be smaller than payload_len */
if (num_packets + num_nonempty_packets + amr_len > payload_len) {
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP wrong length 1"));
goto bad_packet;
}
if (num_packets + num_nonempty_packets + amr_len > payload_len)
goto wrong_length_1;
} else {
/* data len + header bytes should be smaller than payload_len */
if (num_packets + amr_len > payload_len) {
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP wrong length 2"));
goto bad_packet;
}
if (num_packets + amr_len > payload_len)
goto wrong_length_2;
}
timestamp = gst_rtp_buffer_get_timestamp (buf);
outbuf = gst_buffer_new_and_alloc (payload_len);
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale_int (timestamp, GST_SECOND,
depayload->clock_rate);
/* point to destination */
p = GST_BUFFER_DATA (outbuf);
......@@ -386,16 +363,51 @@ gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
GST_DEBUG ("gst_rtp_amr_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
}
return outbuf;
/* ERRORS */
invalid_packet:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP packet did not validate"));
goto bad_packet;
}
not_negotiated:
{
GST_ELEMENT_ERROR (rtpamrdepay, STREAM, NOT_IMPLEMENTED,
(NULL), ("not negotiated"));
return NULL;
}
too_small:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP payload too small (%d)", payload_len));
goto bad_packet;
}
wrong_interleaving:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP wrong interleaving"));
goto bad_packet;
}
wrong_framesize:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP frame size == -1"));
goto bad_packet;
}
wrong_length_1:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP wrong length 1"));
goto bad_packet;
}
wrong_length_2:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP wrong length 2"));
goto bad_packet;
}
bad_packet:
{
/* no fatal error */
......
......@@ -154,8 +154,10 @@ gst_rtp_h264_depay_finalize (GObject * object)
rtph264depay = GST_RTP_H264_DEPAY (object);
if (rtph264depay->codec_data)
gst_buffer_unref (rtph264depay->codec_data);
g_object_unref (rtph264depay->adapter);
rtph264depay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
......@@ -255,9 +257,12 @@ gst_rtp_h264_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
}
GST_BUFFER_SIZE (codec_data) = total;
/* don't set codec_data, we send unpacketized data so let the decoder
* packetize for us */
gst_adapter_push (rtph264depay->adapter, codec_data);
/* keep the codec_data, we need to send it as the first buffer. We cannot
* push it in the adapter because the adapter might be flushed on discont.
*/
if (rtph264depay->codec_data)
gst_buffer_unref (rtph264depay->codec_data);
rtph264depay->codec_data = codec_data;
}
gst_pad_set_caps (depayload->srcpad, srccaps);
......@@ -333,6 +338,12 @@ gst_rtp_h264_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
rtph264depay->wait_start = FALSE;
/* prepend codec_data */
if (rtph264depay->codec_data) {
gst_adapter_push (rtph264depay->adapter, rtph264depay->codec_data);
rtph264depay->codec_data = NULL;
}
/* STAP-A Single-time aggregation packet 5.7.1 */
while (payload_len > 2) {
/* 1
......@@ -446,13 +457,20 @@ gst_rtp_h264_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
/* if NAL unit ends, flush the adapter */
if (E) {
GST_DEBUG_OBJECT (rtph264depay, "output %d bytes", outsize);
outsize = gst_adapter_available (rtph264depay->adapter);
outbuf = gst_adapter_take_buffer (rtph264depay->adapter, outsize);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
GST_DEBUG_OBJECT (rtph264depay, "output %d bytes", outsize);
/* push codec_data first */
if (rtph264depay->codec_data) {
gst_buffer_set_caps (rtph264depay->codec_data,
GST_PAD_CAPS (depayload->srcpad));
gst_base_rtp_depayload_push (depayload, rtph264depay->codec_data);
rtph264depay->codec_data = NULL;
}
return outbuf;
}
break;
......@@ -473,6 +491,13 @@ gst_rtp_h264_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
/* push codec_data first */
if (rtph264depay->codec_data) {
gst_buffer_set_caps (rtph264depay->codec_data,
GST_PAD_CAPS (depayload->srcpad));
gst_base_rtp_depayload_push (depayload, rtph264depay->codec_data);
rtph264depay->codec_data = NULL;
}
return outbuf;
}
}
......
......@@ -44,6 +44,7 @@ struct _GstRtpH264Depay
{
GstBaseRTPDepayload depayload;
GstBuffer *codec_data;
GstAdapter *adapter;
gboolean wait_start;
};
......
......@@ -287,64 +287,92 @@ gst_rtp_mp4g_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
guint32 timestamp;
guint AU_headers_len;
guint AU_size, AU_index;
gboolean M;
payload_len = gst_rtp_buffer_get_payload_len (buf);
payload = gst_rtp_buffer_get_payload (buf);
payload_header = 0;
timestamp = gst_rtp_buffer_get_timestamp (buf);
M = gst_rtp_buffer_get_marker (buf);
if (rtpmp4gdepay->sizelength > 0) {
gint num_AU_headers, AU_headers_bytes, i;
/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
* |AU-headers-length|AU-header|AU-header| |AU-header|padding|
* | | (1) | (2) | | (n) * | bits |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
*
* The lenght is 2 bytes and contains the length of the following
* The length is 2 bytes and contains the length of the following
* AU-headers in bits.
*/
AU_headers_len = (payload[0] << 8) | payload[1];
AU_headers_bytes = (AU_headers_len + 7) / 8;
num_AU_headers = AU_headers_len / 16;
GST_DEBUG_OBJECT (rtpmp4gdepay, "AU headers len %d, bytes %d, num %d",
AU_headers_len, AU_headers_bytes, num_AU_headers);
/* skip header */
payload += 2;
payload_header += 2;
payload_len -= 2;
/* FIXME, use bits */
AU_size = ((payload[0] << 8) | payload[1]) >> 3;
AU_index = payload[1] & 0x7;
GST_DEBUG_OBJECT (rtpmp4gdepay, "len, %d, size %d, index %d",
AU_headers_len, AU_size, AU_index);
/* skip special headers */
payload += (AU_headers_len + 7) / 8;
payload_header += (AU_headers_len + 7) / 8;
payload_len = AU_size;
}
timestamp = gst_rtp_buffer_get_timestamp (buf);
/* strip header from payload and push in the adapter */
outbuf =
gst_rtp_buffer_get_payload_subbuffer (buf, payload_header, payload_len);
gst_adapter_push (rtpmp4gdepay->adapter, outbuf);
payload_header = 2 + AU_headers_bytes;
for (i = 0; i < num_AU_headers; i++) {
/* FIXME, use bits */
AU_size = ((payload[0] << 8) | payload[1]) >> 3;
AU_index = payload[1] & 0x7;
payload += 2;
GST_DEBUG_OBJECT (rtpmp4gdepay, "len, %d, size %d, index %d",
AU_headers_len, AU_size, AU_index);
/* collect stuff in the adapter, strip header from payload and push in
* the adapter */
outbuf =
gst_rtp_buffer_get_payload_subbuffer (buf, payload_header, AU_size);
gst_adapter_push (rtpmp4gdepay->adapter, outbuf);
if (M) {
guint avail;
/* packet is complete, flush */
avail = gst_adapter_available (rtpmp4gdepay->adapter);
outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
/* only apply the timestamp for the first buffer */
if (i == 0)
gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
else
gst_base_rtp_depayload_push (depayload, outbuf);
}
payload_header += AU_size;
}
} else {
/* push complete buffer in adapter */
outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 0, payload_len);
gst_adapter_push (rtpmp4gdepay->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;
/* if this was the last packet of the VOP, create and push a buffer */
if (M) {
guint avail;
avail = gst_adapter_available (rtpmp4gdepay->adapter);
avail = gst_adapter_available (rtpmp4gdepay->adapter);
outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
GST_BUFFER_TIMESTAMP (outbuf) = gst_util_uint64_scale_int
(timestamp, GST_SECOND, depayload->clock_rate);
outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
} else {
return NULL;
return outbuf;
}
}
}
return NULL;
......
......@@ -252,16 +252,11 @@ gst_rtp_sv3v_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
if (M) {
/* frame is completed: push contents of adapter */
guint avail;
guint32 timestamp;
avail = gst_adapter_available (rtpsv3vdepay->adapter);
outbuf = gst_adapter_take_buffer (rtpsv3vdepay->adapter, avail);
/* timestamp for complete buffer is that of last buffer as well */
timestamp = gst_rtp_buffer_get_timestamp (buf);
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale_int (timestamp, GST_SECOND,
depayload->clock_rate);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
return outbuf;
......@@ -342,5 +337,5 @@ gboolean
gst_rtp_sv3v_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpsv3vdepay",
GST_RANK_MARGINAL, GST_TYPE_RTP_SV3V_DEPAY);
GST_RANK_NONE, GST_TYPE_RTP_SV3V_DEPAY);
}
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