gstreamer/gst/rtp/gstrtph264pay.c
Wim Taymans 4a7cbe8489 fixes: #514889
Original commit message from CVS:
patch by:  Wim Taymans  <wim.taymans@collabora.co.uk>
fixes: #514889
* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4gpay.h:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix various leaks shown up in valgrind
- free sprops and buffer in error cases in H264 payloader
- fix leak in mp4g depayloader when construction the caps
- don't leak config string in the mp4g payloader
- don't leak buffers and headers in theora and vorbis payloaders
* tests/check/elements/rtp-payloading.c:
Fix the RTP data test
- Actually send valid amr data to the payloader instead of 20
zero-bytes
- The mp4g payloader expects codec_data on the caps
2008-02-12 23:38:19 +00:00

562 lines
16 KiB
C

/* GStreamer
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtph264pay.h"
#define SPS_TYPE_ID 7
#define PPS_TYPE_ID 8
#define USE_MEMCMP
GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug);
#define GST_CAT_DEFAULT (rtph264pay_debug)
/* references:
*
* RFC 3984
*/
/* elementfactory information */
static const GstElementDetails gst_rtp_h264pay_details =
GST_ELEMENT_DETAILS ("RTP packet payloader",
"Codec/Payloader/Network",
"Payload-encode H264 video into RTP packets (RFC 3984)",
"Laurent Glayal <spglegle@yahoo.fr>");
static GstStaticPadTemplate gst_rtp_h264_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-h264")
);
static GstStaticPadTemplate gst_rtp_h264_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"video\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"H264\"")
);
static void gst_rtp_h264_pay_finalize (GObject * object);
static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * pad,
GstBuffer * buffer);
GST_BOILERPLATE (GstRtpH264Pay, gst_rtp_h264_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_h264_pay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_h264_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_h264_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_h264pay_details);
}
static void
gst_rtp_h264_pay_class_init (GstRtpH264PayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gobject_class->finalize = gst_rtp_h264_pay_finalize;
gstelement_class->change_state = gst_rtp_h264_pay_change_state;
gstbasertppayload_class->set_caps = gst_rtp_h264_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtph264pay_debug, "rtph264pay", 0,
"H264 RTP Payloader");
}
static void
gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay, GstRtpH264PayClass * klass)
{
rtph264pay->profile = 0;
rtph264pay->sps = NULL;
rtph264pay->pps = NULL;
}
static void
gst_rtp_h264_pay_finalize (GObject * object)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
if (rtph264pay->sps)
g_free (rtph264pay->sps);
if (rtph264pay->pps)
g_free (rtph264pay->pps);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (basepayload);
/* we can only set the output caps when we found the sprops and profile
* NALs */
gst_basertppayload_set_options (basepayload, "video", TRUE, "H264", 90000);
return TRUE;
}
static guint
next_start_code (guint8 * data, guint size)
{
/* Boyer-Moore string matching algorithm, in a degenerative
* sense because our search 'alphabet' is binary - 0 & 1 only.
* This allow us to simplify the general BM algorithm to a very
* simple form. */
/* assume 1 is in the 4th byte */
guint offset = 3;
while (offset < size) {
if (1 == data[offset]) {
unsigned int shift = offset;
if (0 == data[--shift]) {
if (0 == data[--shift]) {
if (0 == data[--shift]) {
return shift;
}
}
}
/* The jump is always 4 because of the 1 previously matched.
* All the 0's must be after this '1' matched at offset */
offset += 4;
} else if (0 == data[offset]) {
/* maybe next byte is 1? */
offset++;
} else {
/* can jump 4 bytes forward */
offset += 4;
}
/* at each iteration, we rescan in a backward manner until
* we match 0.0.0.1 in reverse order. Since our search string
* has only 2 'alpabets' (i.e. 0 & 1), we know that any
* mismatch will force us to shift a fixed number of steps */
}
GST_DEBUG ("Cannot find next NAL start code. returning %u", size);
return size;
}
/* we don't use memcpy but this faster version (around 20%) because we need to
* perform it on all data. */
static gboolean
is_nal_equal (const guint8 * nal1, const guint8 * nal2, guint len)
{
/* if we have a 64-bit processor, we may use guint64 to make
* this go faster. Otherwise with 32 bits, we are already
* going faster than byte to byte compare.
*/
guint remainder = len & 0x3;
guint num_int = len >> 2;
guint32 *pu1 = (guint32 *) nal1, *pu2 = (guint32 *) nal2;
guint i;
/* compare by groups of sizeof(guint32) bytes */
for (i = 0; i < num_int; i++) {
/* XOR is faster than CMP?... */
if (pu1[i] ^ pu2[i])
return FALSE;
}
/* check that the remaining bytes are still equal */
if (!remainder) {
return TRUE;
} else if (1 == remainder) {
return (nal1[--len] == nal2[len]);
} else { /* 2 or 3 */
if (remainder & 1) { /* -1 if 3 bytes left */
if (nal1[--len] != nal2[len])
return FALSE;
}
/* last 2 bytes */
return ((nal1[--len] == nal2[len]) /* -1 */
&&(nal1[--len] == nal2[len])); /* -2 */
}
}
static void
gst_rtp_h264_pay_decode_nal (GstRtpH264Pay * payloader,
guint8 * data, guint size, gboolean * updated)
{
guint8 *sps = NULL, *pps = NULL;
guint sps_len = 0, pps_len = 0;
/* default is no update */
*updated = FALSE;
if (size <= 3) {
GST_WARNING ("Encoded buffer len %u <= 3", size);
} else {
GST_DEBUG ("NAL payload len=%u", size);
/* loop through all NAL units and save the locations of any
* SPS / PPS for later processing. Only the last seen SPS
* or PPS will be considered */
while (size > 5) {
guint8 header, type;
guint len;
len = next_start_code (data, size);
header = data[0];
type = header & 0x1f;
/* keep sps & pps separately so that we can update either one
* independently */
if (SPS_TYPE_ID == type) {
/* encode the entire SPS NAL in base64 */
GST_DEBUG ("Found SPS %x %x %x Len=%u\n", (header >> 7),
(header >> 5) & 3, type, len);
sps = data;
sps_len = len;
} else if (PPS_TYPE_ID == type) {
/* encoder the entire PPS NAL in base64 */
GST_DEBUG ("Found PPS %x %x %x Len = %u\n",
(header >> 7), (header >> 5) & 3, type, len);
pps = data;
pps_len = len;
} else {
GST_DEBUG ("NAL: %x %x %x Len = %u\n", (header >> 7),
(header >> 5) & 3, type, len);
}
/* end of loop */
if (len >= size - 4) {
break;
}
/* next NAL start */
data += len + 4;
size -= len + 4;
}
/* If we encountered an SPS and/or a PPS, check if it's the
* same as the one we have. If not, update our version and
* set *updated to TRUE
*/
if (sps_len > 0) {
if ((payloader->sps_len != sps_len)
|| !is_nal_equal (payloader->sps, sps, sps_len)) {
payloader->profile = (sps[1] << 16) + (sps[2] << 8) + sps[3];
GST_DEBUG ("Profile level IDC = %06x", payloader->profile);
if (payloader->sps_len)
g_free (payloader->sps);
payloader->sps = sps_len ? g_new (guint8, sps_len) : NULL;
memcpy (payloader->sps, sps, sps_len);
payloader->sps_len = sps_len;
*updated = TRUE;
}
}
if (pps_len > 0) {
if ((payloader->pps_len != pps_len)
|| !is_nal_equal (payloader->pps, pps, pps_len)) {
if (payloader->pps_len)
g_free (payloader->pps);
payloader->pps = pps_len ? g_new (guint8, pps_len) : NULL;
memcpy (payloader->pps, pps, pps_len);
payloader->pps_len = pps_len;
*updated = TRUE;
}
}
}
}
static gchar *
encode_base64 (const guint8 * in, guint size, guint * len)
{
gchar *ret, *d;
static const gchar *v =
"ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
*len = ((size + 2) / 3) * 4;
d = ret = (gchar *) g_malloc (*len + 1);
for (; size; in += 3) { /* process tuplets */
*d++ = v[in[0] >> 2]; /* byte 1: high 6 bits (1) */
/* byte 2: low 2 bits (1), high 4 bits (2) */
*d++ = v[((in[0] << 4) + (--size ? (in[1] >> 4) : 0)) & 0x3f];
/* byte 3: low 4 bits (2), high 2 bits (3) */
*d++ = size ? v[((in[1] << 2) + (--size ? (in[2] >> 6) : 0)) & 0x3f] : '=';
/* byte 4: low 6 bits (3) */
*d++ = size ? v[in[2] & 0x3f] : '=';
if (size)
size--; /* count third character if processed */
}
*d = '\0'; /* tie off string */
return ret; /* return the resulting string */
}
static void
gst_rtp_h264_pay_parse_sps_pps (GstBaseRTPPayload * basepayload,
guint8 * data, guint size)
{
gboolean update = FALSE;
GstRtpH264Pay *payloader = GST_RTP_H264_PAY (basepayload);
gst_rtp_h264_pay_decode_nal (payloader, data, size, &update);
/* if has new SPS & PPS, update the output caps */
if (update) {
gchar *profile;
gchar *sps;
gchar *pps;
gchar *sprops;
guint len;
/* profile is 24 bit. Force it to respect the limit */
profile = g_strdup_printf ("%06x", payloader->profile & 0xffffff);
/* build the sprop-parameter-sets */
sps = (payloader->sps_len > 0)
? encode_base64 (payloader->sps, payloader->sps_len, &len) : NULL;
pps = (payloader->pps_len > 0)
? encode_base64 (payloader->pps, payloader->pps_len, &len) : NULL;
if (sps)
sprops = g_strjoin (",", sps, pps, NULL);
else
sprops = g_strdup (pps);
gst_basertppayload_set_outcaps (basepayload, "profile-level-id",
G_TYPE_STRING, profile,
"sprop-parameter-sets", G_TYPE_STRING, sprops, NULL);
GST_DEBUG ("outcaps udpate: profile=%s, sps=%s, pps=%s\n",
profile, sps, pps);
g_free (sprops);
g_free (profile);
g_free (sps);
g_free (pps);
}
}
static GstFlowReturn
gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
guint size, idxdata;
GstBuffer *outbuf;
guint8 *payload, *data, *pdata;
guint8 nalType;
GstClockTime timestamp;
guint packet_len, payload_len, mtu;
rtph264pay = GST_RTP_H264_PAY (basepayload);
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtph264pay);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
/* H264 stream analysis */
pdata = data;
/* use next_start_code() to scan buffer.
* next_start_code() returns the offset in data,
* starting from zero to the first byte of 0.0.0.1
* If no start code is found, it returns the value of the
* 'size' parameter.
* pdata is unchanged by the call to next_start_code()
*/
{
guint offset = next_start_code (pdata, size);
pdata += offset;
idxdata = size - offset;
}
if (idxdata < 5) {
GST_DEBUG_OBJECT (basepayload,
"Returning GST_FLOW_OK without creating RTP packet");
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
pdata += 4;
idxdata -= 4;
/* We know our stream is a valid H264 NAL packet,
* go parse it for SPS/PPS to enrich the caps */
gst_rtp_h264_pay_parse_sps_pps (basepayload, pdata, idxdata);
nalType = pdata[0] & 0x1f;
GST_DEBUG_OBJECT (basepayload, "Processing Buffer with NAL TYPE=%d", nalType);
packet_len = gst_rtp_buffer_calc_packet_len (idxdata, 0, 0);
if (packet_len < mtu) {
GST_DEBUG_OBJECT (basepayload,
"NAL Unit fit in one packet datasize=%d mtu=%d", idxdata, mtu);
/* will fit in one packet */
outbuf = gst_rtp_buffer_new_allocate (idxdata, 0, 0);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
gst_rtp_buffer_set_marker (outbuf, 1);
payload = gst_rtp_buffer_get_payload (outbuf);
GST_DEBUG_OBJECT (basepayload, "Copying %d bytes to outbuf", idxdata);
memcpy (payload, pdata, idxdata);
gst_buffer_unref (buffer);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
} else {
/* Fragmentation Units FU-A */
guint8 nalHeader;
guint limitedSize;
int ii = 0, start = 1, end = 0, first = 0;
GST_DEBUG_OBJECT (basepayload,
"NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", idxdata, mtu);
nalHeader = *pdata;
pdata++;
idxdata--;
ret = GST_FLOW_OK;
GST_DEBUG_OBJECT (basepayload, "Using FU-A fragmentation for data size=%d",
idxdata);
/* We keep 2 bytes for FU indicator and FU Header */
payload_len = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
while (end == 0) {
limitedSize = idxdata < payload_len ? idxdata : payload_len;
GST_DEBUG_OBJECT (basepayload,
"Inside FU-A fragmentation limitedSize=%d iteration=%d", limitedSize,
ii);
outbuf = gst_rtp_buffer_new_allocate (limitedSize + 2, 0, 0);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
payload = gst_rtp_buffer_get_payload (outbuf);
if (limitedSize == idxdata) {
GST_DEBUG_OBJECT (basepayload, "end idxdata=%d iteration=%d", idxdata,
ii);
end = 1;
}
gst_rtp_buffer_set_marker (outbuf, end);
/* FU indicator */
payload[0] = (nalHeader & 0x60) | 28;
/* FU Header */
payload[1] = (start << 7) | (end << 6) | (nalHeader & 0x1f);
memcpy (&payload[2], pdata + first, limitedSize);
GST_DEBUG_OBJECT (basepayload,
"recorded %d payload bytes into packet iteration=%d", limitedSize + 2,
ii);
ret = gst_basertppayload_push (basepayload, outbuf);
if (ret != GST_FLOW_OK)
break;
idxdata -= limitedSize;
first += limitedSize;
ii++;
start = 0;
}
gst_buffer_unref (buffer);
return ret;
}
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("Should not be there !!"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
static GstStateChangeReturn
gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition)
{
GstRtpH264Pay *rtph264pay;
GstStateChangeReturn ret;
rtph264pay = GST_RTP_H264_PAY (element);
switch (transition) {
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
default:
break;
}
return ret;
}
gboolean
gst_rtp_h264_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtph264pay",
GST_RANK_NONE, GST_TYPE_RTP_H264_PAY);
}