mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-24 18:51:11 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtp/gstrtph264pay.c
Jan Alexander Steffens (heftig) 20758215b5 rtph26*pay: Update param set timestamp even if parameters unchanged 
rtph264pay and rtph265pay skip updating the parameter set timestamp if
the units they see contain no new configuration. This can result in
them injecting duplicate parameters.

https://bugzilla.gnome.org/show_bug.cgi?id=796748
2018-08-16 16:49:16 +03:00

1432 lines
42 KiB
C
Raw Blame History

/* ex: set tabstop=2 shiftwidth=2 expandtab: */
/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/pbutils/pbutils.h>
#include <gst/video/video.h>
/* Included to not duplicate gst_rtp_h264_add_sps_pps () */
#include "gstrtph264depay.h"
#include "gstrtph264pay.h"
#include "gstrtputils.h"
#define IDR_TYPE_ID 5
#define SPS_TYPE_ID 7
#define PPS_TYPE_ID 8
GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug);
#define GST_CAT_DEFAULT (rtph264pay_debug)
/* references:
*
* RFC 3984
*/
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, "
"stream-format = (string) avc, alignment = (string) au;"
"video/x-h264, "
"stream-format = (string) byte-stream, alignment = (string) { nal, au }")
);
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\"")
);
#define DEFAULT_SPROP_PARAMETER_SETS NULL
#define DEFAULT_CONFIG_INTERVAL 0
enum
{
PROP_0,
PROP_SPROP_PARAMETER_SETS,
PROP_CONFIG_INTERVAL
};
#define IS_ACCESS_UNIT(x) (((x) > 0x00) && ((x) < 0x06))
static void gst_rtp_h264_pay_finalize (GObject * object);
static void gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_rtp_h264_pay_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter);
static gboolean gst_rtp_h264_pay_setcaps (GstRTPBasePayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * pad,
GstBuffer * buffer);
static gboolean gst_rtp_h264_pay_sink_event (GstRTPBasePayload * payload,
GstEvent * event);
static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement *
element, GstStateChange transition);
#define gst_rtp_h264_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpH264Pay, gst_rtp_h264_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_h264_pay_class_init (GstRtpH264PayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->set_property = gst_rtp_h264_pay_set_property;
gobject_class->get_property = gst_rtp_h264_pay_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_SPROP_PARAMETER_SETS, g_param_spec_string ("sprop-parameter-sets",
"sprop-parameter-sets",
"The base64 sprop-parameter-sets to set in out caps (set to NULL to "
"extract from stream)",
DEFAULT_SPROP_PARAMETER_SETS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_CONFIG_INTERVAL,
g_param_spec_int ("config-interval",
"SPS PPS Send Interval",
"Send SPS and PPS Insertion Interval in seconds (sprop parameter sets "
"will be multiplexed in the data stream when detected.) "
"(0 = disabled, -1 = send with every IDR frame)",
-1, 3600, DEFAULT_CONFIG_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
gobject_class->finalize = gst_rtp_h264_pay_finalize;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_h264_pay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_h264_pay_sink_template);
gst_element_class_set_static_metadata (gstelement_class, "RTP H264 payloader",
"Codec/Payloader/Network/RTP",
"Payload-encode H264 video into RTP packets (RFC 3984)",
"Laurent Glayal <spglegle@yahoo.fr>");
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_h264_pay_change_state);
gstrtpbasepayload_class->get_caps = gst_rtp_h264_pay_getcaps;
gstrtpbasepayload_class->set_caps = gst_rtp_h264_pay_setcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer;
gstrtpbasepayload_class->sink_event = gst_rtp_h264_pay_sink_event;
GST_DEBUG_CATEGORY_INIT (rtph264pay_debug, "rtph264pay", 0,
"H264 RTP Payloader");
}
static void
gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay)
{
rtph264pay->queue = g_array_new (FALSE, FALSE, sizeof (guint));
rtph264pay->profile = 0;
rtph264pay->sps = g_ptr_array_new_with_free_func (
(GDestroyNotify) gst_buffer_unref);
rtph264pay->pps = g_ptr_array_new_with_free_func (
(GDestroyNotify) gst_buffer_unref);
rtph264pay->last_spspps = -1;
rtph264pay->spspps_interval = DEFAULT_CONFIG_INTERVAL;
rtph264pay->delta_unit = FALSE;
rtph264pay->discont = FALSE;
rtph264pay->adapter = gst_adapter_new ();
}
static void
gst_rtp_h264_pay_clear_sps_pps (GstRtpH264Pay * rtph264pay)
{
g_ptr_array_set_size (rtph264pay->sps, 0);
g_ptr_array_set_size (rtph264pay->pps, 0);
}
static void
gst_rtp_h264_pay_finalize (GObject * object)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
g_array_free (rtph264pay->queue, TRUE);
g_ptr_array_free (rtph264pay->sps, TRUE);
g_ptr_array_free (rtph264pay->pps, TRUE);
g_free (rtph264pay->sprop_parameter_sets);
g_object_unref (rtph264pay->adapter);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static const gchar all_levels[][4] = {
"1",
"1b",
"1.1",
"1.2",
"1.3",
"2",
"2.1",
"2.2",
"3",
"3.1",
"3.2",
"4",
"4.1",
"4.2",
"5",
"5.1"
};
static GstCaps *
gst_rtp_h264_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
GstCaps * filter)
{
GstCaps *template_caps;
GstCaps *allowed_caps;
GstCaps *caps, *icaps;
gboolean append_unrestricted;
guint i;
allowed_caps =
gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), NULL);
if (allowed_caps == NULL)
return NULL;
template_caps =
gst_static_pad_template_get_caps (&gst_rtp_h264_pay_sink_template);
if (gst_caps_is_any (allowed_caps)) {
caps = gst_caps_ref (template_caps);
goto done;
}
if (gst_caps_is_empty (allowed_caps)) {
caps = gst_caps_ref (allowed_caps);
goto done;
}
caps = gst_caps_new_empty ();
append_unrestricted = FALSE;
for (i = 0; i < gst_caps_get_size (allowed_caps); i++) {
GstStructure *s = gst_caps_get_structure (allowed_caps, i);
GstStructure *new_s = gst_structure_new_empty ("video/x-h264");
const gchar *profile_level_id;
profile_level_id = gst_structure_get_string (s, "profile-level-id");
if (profile_level_id && strlen (profile_level_id) == 6) {
const gchar *profile;
const gchar *level;
long int spsint;
guint8 sps[3];
spsint = strtol (profile_level_id, NULL, 16);
sps[0] = spsint >> 16;
sps[1] = spsint >> 8;
sps[2] = spsint;
profile = gst_codec_utils_h264_get_profile (sps, 3);
level = gst_codec_utils_h264_get_level (sps, 3);
if (profile && level) {
GST_LOG_OBJECT (payload, "In caps, have profile %s and level %s",
profile, level);
if (!strcmp (profile, "constrained-baseline"))
gst_structure_set (new_s, "profile", G_TYPE_STRING, profile, NULL);
else {
GValue val = { 0, };
GValue profiles = { 0, };
g_value_init (&profiles, GST_TYPE_LIST);
g_value_init (&val, G_TYPE_STRING);
g_value_set_static_string (&val, profile);
gst_value_list_append_value (&profiles, &val);
g_value_set_static_string (&val, "constrained-baseline");
gst_value_list_append_value (&profiles, &val);
gst_structure_take_value (new_s, "profile", &profiles);
}
if (!strcmp (level, "1"))
gst_structure_set (new_s, "level", G_TYPE_STRING, level, NULL);
else {
GValue levels = { 0, };
GValue val = { 0, };
int j;
g_value_init (&levels, GST_TYPE_LIST);
g_value_init (&val, G_TYPE_STRING);
for (j = 0; j < G_N_ELEMENTS (all_levels); j++) {
g_value_set_static_string (&val, all_levels[j]);
gst_value_list_prepend_value (&levels, &val);
if (!strcmp (level, all_levels[j]))
break;
}
gst_structure_take_value (new_s, "level", &levels);
}
} else {
/* Invalid profile-level-id means baseline */
gst_structure_set (new_s,
"profile", G_TYPE_STRING, "constrained-baseline", NULL);
}
} else {
/* No profile-level-id means baseline or unrestricted */
gst_structure_set (new_s,
"profile", G_TYPE_STRING, "constrained-baseline", NULL);
append_unrestricted = TRUE;
}
caps = gst_caps_merge_structure (caps, new_s);
}
if (append_unrestricted) {
caps =
gst_caps_merge_structure (caps, gst_structure_new ("video/x-h264", NULL,
NULL));
}
icaps = gst_caps_intersect (caps, template_caps);
gst_caps_unref (caps);
caps = icaps;
done:
if (filter) {
GST_DEBUG_OBJECT (payload, "Intersect %" GST_PTR_FORMAT " and filter %"
GST_PTR_FORMAT, caps, filter);
icaps = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = icaps;
}
gst_caps_unref (template_caps);
gst_caps_unref (allowed_caps);
GST_LOG_OBJECT (payload, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
}
/* take the currently configured SPS and PPS lists and set them on the caps as
* sprop-parameter-sets */
static gboolean
gst_rtp_h264_pay_set_sps_pps (GstRTPBasePayload * basepayload)
{
GstRtpH264Pay *payloader = GST_RTP_H264_PAY (basepayload);
gchar *profile;
gchar *set;
GString *sprops;
guint count;
gboolean res;
GstMapInfo map;
guint i;
sprops = g_string_new ("");
count = 0;
/* build the sprop-parameter-sets */
for (i = 0; i < payloader->sps->len; i++) {
GstBuffer *sps_buf =
GST_BUFFER_CAST (g_ptr_array_index (payloader->sps, i));
gst_buffer_map (sps_buf, &map, GST_MAP_READ);
set = g_base64_encode (map.data, map.size);
gst_buffer_unmap (sps_buf, &map);
g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
g_free (set);
count++;
}
for (i = 0; i < payloader->pps->len; i++) {
GstBuffer *pps_buf =
GST_BUFFER_CAST (g_ptr_array_index (payloader->pps, i));
gst_buffer_map (pps_buf, &map, GST_MAP_READ);
set = g_base64_encode (map.data, map.size);
gst_buffer_unmap (pps_buf, &map);
g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
g_free (set);
count++;
}
if (G_LIKELY (count)) {
if (payloader->profile != 0) {
/* profile is 24 bit. Force it to respect the limit */
profile = g_strdup_printf ("%06x", payloader->profile & 0xffffff);
/* combine into output caps */
res = gst_rtp_base_payload_set_outcaps (basepayload,
"packetization-mode", G_TYPE_STRING, "1",
"profile-level-id", G_TYPE_STRING, profile,
"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
g_free (profile);
} else {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"packetization-mode", G_TYPE_STRING, "1",
"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
}
} else {
res = gst_rtp_base_payload_set_outcaps (basepayload, NULL);
}
g_string_free (sprops, TRUE);
return res;
}
static gboolean
gst_rtp_h264_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpH264Pay *rtph264pay;
GstStructure *str;
const GValue *value;
GstMapInfo map;
guint8 *data;
gsize size;
GstBuffer *buffer;
const gchar *alignment, *stream_format;
rtph264pay = GST_RTP_H264_PAY (basepayload);
str = gst_caps_get_structure (caps, 0);
/* we can only set the output caps when we found the sprops and profile
* NALs */
gst_rtp_base_payload_set_options (basepayload, "video", TRUE, "H264", 90000);
rtph264pay->alignment = GST_H264_ALIGNMENT_UNKNOWN;
alignment = gst_structure_get_string (str, "alignment");
if (alignment) {
if (g_str_equal (alignment, "au"))
rtph264pay->alignment = GST_H264_ALIGNMENT_AU;
if (g_str_equal (alignment, "nal"))
rtph264pay->alignment = GST_H264_ALIGNMENT_NAL;
}
rtph264pay->stream_format = GST_H264_STREAM_FORMAT_UNKNOWN;
stream_format = gst_structure_get_string (str, "stream-format");
if (stream_format) {
if (g_str_equal (stream_format, "avc"))
rtph264pay->stream_format = GST_H264_STREAM_FORMAT_AVC;
if (g_str_equal (stream_format, "byte-stream"))
rtph264pay->stream_format = GST_H264_STREAM_FORMAT_BYTESTREAM;
}
/* packetized AVC video has a codec_data */
if ((value = gst_structure_get_value (str, "codec_data"))) {
guint num_sps, num_pps;
gint i, nal_size;
GST_DEBUG_OBJECT (rtph264pay, "have packetized h264");
buffer = gst_value_get_buffer (value);
gst_buffer_map (buffer, &map, GST_MAP_READ);
data = map.data;
size = map.size;
/* parse the avcC data */
if (size < 7)
goto avcc_too_small;
/* parse the version, this must be 1 */
if (data[0] != 1)
goto wrong_version;
/* AVCProfileIndication */
/* profile_compat */
/* AVCLevelIndication */
rtph264pay->profile = (data[1] << 16) | (data[2] << 8) | data[3];
GST_DEBUG_OBJECT (rtph264pay, "profile %06x", rtph264pay->profile);
/* 6 bits reserved | 2 bits lengthSizeMinusOne */
/* this is the number of bytes in front of the NAL units to mark their
* length */
rtph264pay->nal_length_size = (data[4] & 0x03) + 1;
GST_DEBUG_OBJECT (rtph264pay, "nal length %u", rtph264pay->nal_length_size);
/* 3 bits reserved | 5 bits numOfSequenceParameterSets */
num_sps = data[5] & 0x1f;
GST_DEBUG_OBJECT (rtph264pay, "num SPS %u", num_sps);
data += 6;
size -= 6;
/* create the sprop-parameter-sets */
for (i = 0; i < num_sps; i++) {
GstBuffer *sps_buf;
if (size < 2)
goto avcc_error;
nal_size = (data[0] << 8) | data[1];
data += 2;
size -= 2;
GST_LOG_OBJECT (rtph264pay, "SPS %d size %d", i, nal_size);
if (size < nal_size)
goto avcc_error;
/* make a buffer out of it and add to SPS list */
sps_buf = gst_buffer_new_and_alloc (nal_size);
gst_buffer_fill (sps_buf, 0, data, nal_size);
gst_rtp_h264_add_sps_pps (GST_ELEMENT (rtph264pay), rtph264pay->sps,
rtph264pay->pps, sps_buf);
data += nal_size;
size -= nal_size;
}
if (size < 1)
goto avcc_error;
/* 8 bits numOfPictureParameterSets */
num_pps = data[0];
data += 1;
size -= 1;
GST_DEBUG_OBJECT (rtph264pay, "num PPS %u", num_pps);
for (i = 0; i < num_pps; i++) {
GstBuffer *pps_buf;
if (size < 2)
goto avcc_error;
nal_size = (data[0] << 8) | data[1];
data += 2;
size -= 2;
GST_LOG_OBJECT (rtph264pay, "PPS %d size %d", i, nal_size);
if (size < nal_size)
goto avcc_error;
/* make a buffer out of it and add to PPS list */
pps_buf = gst_buffer_new_and_alloc (nal_size);
gst_buffer_fill (pps_buf, 0, data, nal_size);
gst_rtp_h264_add_sps_pps (GST_ELEMENT (rtph264pay), rtph264pay->sps,
rtph264pay->pps, pps_buf);
data += nal_size;
size -= nal_size;
}
/* and update the caps with the collected data */
if (!gst_rtp_h264_pay_set_sps_pps (basepayload))
goto set_sps_pps_failed;
gst_buffer_unmap (buffer, &map);
} else {
GST_DEBUG_OBJECT (rtph264pay, "have bytestream h264");
}
return TRUE;
avcc_too_small:
{
GST_ERROR_OBJECT (rtph264pay, "avcC size %" G_GSIZE_FORMAT " < 7", size);
goto error;
}
wrong_version:
{
GST_ERROR_OBJECT (rtph264pay, "wrong avcC version");
goto error;
}
avcc_error:
{
GST_ERROR_OBJECT (rtph264pay, "avcC too small ");
goto error;
}
set_sps_pps_failed:
{
GST_ERROR_OBJECT (rtph264pay, "failed to set sps/pps");
goto error;
}
error:
{
gst_buffer_unmap (buffer, &map);
return FALSE;
}
}
static void
gst_rtp_h264_pay_parse_sprop_parameter_sets (GstRtpH264Pay * rtph264pay)
{
const gchar *ps;
gchar **params;
guint len;
gint i;
GstBuffer *buf;
ps = rtph264pay->sprop_parameter_sets;
if (ps == NULL)
return;
gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
params = g_strsplit (ps, ",", 0);
len = g_strv_length (params);
GST_DEBUG_OBJECT (rtph264pay, "we have %d params", len);
for (i = 0; params[i]; i++) {
gsize nal_len;
GstMapInfo map;
guint8 *nalp;
guint save = 0;
gint state = 0;
nal_len = strlen (params[i]);
buf = gst_buffer_new_and_alloc (nal_len);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
nalp = map.data;
nal_len = g_base64_decode_step (params[i], nal_len, nalp, &state, &save);
gst_buffer_unmap (buf, &map);
gst_buffer_resize (buf, 0, nal_len);
if (!nal_len) {
gst_buffer_unref (buf);
continue;
}
gst_rtp_h264_add_sps_pps (GST_ELEMENT (rtph264pay), rtph264pay->sps,
rtph264pay->pps, buf);
}
g_strfreev (params);
}
static guint
next_start_code (const 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 3th byte */
guint offset = 2;
while (offset < size) {
if (1 == data[offset]) {
unsigned int shift = offset;
if (0 == data[--shift]) {
if (0 == data[--shift]) {
return shift;
}
}
/* The jump is always 3 because of the 1 previously matched.
* All the 0's must be after this '1' matched at offset */
offset += 3;
} else if (0 == data[offset]) {
/* maybe next byte is 1? */
offset++;
} else {
/* can jump 3 bytes forward */
offset += 3;
}
/* at each iteration, we rescan in a backward manner until
* we match 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;
}
static gboolean
gst_rtp_h264_pay_decode_nal (GstRtpH264Pay * payloader,
const guint8 * data, guint size, GstClockTime dts, GstClockTime pts)
{
guint8 header, type;
gboolean updated;
/* default is no update */
updated = FALSE;
GST_DEBUG ("NAL payload len=%u", size);
header = data[0];
type = header & 0x1f;
/* We record the timestamp of the last SPS/PPS so
* that we can insert them at regular intervals and when needed. */
if (SPS_TYPE_ID == type || PPS_TYPE_ID == type) {
GstBuffer *nal;
/* trailing 0x0 are not part of the SPS/PPS */
while (size > 0 && data[size - 1] == 0x0)
size--;
/* encode the entire SPS NAL in base64 */
GST_DEBUG ("Found %s %x %x %x Len=%u", type == SPS_TYPE_ID ? "SPS" : "PPS",
(header >> 7), (header >> 5) & 3, type, size);
nal = gst_buffer_new_allocate (NULL, size, NULL);
gst_buffer_fill (nal, 0, data, size);
updated = gst_rtp_h264_add_sps_pps (GST_ELEMENT (payloader),
payloader->sps, payloader->pps, nal);
/* remember when we last saw SPS */
if (pts != -1)
payloader->last_spspps =
gst_segment_to_running_time (&GST_RTP_BASE_PAYLOAD_CAST
(payloader)->segment, GST_FORMAT_TIME, pts);
} else {
GST_DEBUG ("NAL: %x %x %x Len = %u", (header >> 7),
(header >> 5) & 3, type, size);
}
return updated;
}
static GstFlowReturn
gst_rtp_h264_pay_payload_nal (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont);
static GstFlowReturn
gst_rtp_h264_pay_send_sps_pps (GstRTPBasePayload * basepayload,
GstRtpH264Pay * rtph264pay, GstClockTime dts, GstClockTime pts)
{
GstFlowReturn ret = GST_FLOW_OK;
gboolean sent_all_sps_pps = TRUE;
guint i;
for (i = 0; i < rtph264pay->sps->len; i++) {
GstBuffer *sps_buf =
GST_BUFFER_CAST (g_ptr_array_index (rtph264pay->sps, i));
GST_DEBUG_OBJECT (rtph264pay, "inserting SPS in the stream");
/* resend SPS */
ret = gst_rtp_h264_pay_payload_nal (basepayload, gst_buffer_ref (sps_buf),
dts, pts, FALSE, FALSE, FALSE);
/* Not critical here; but throw a warning */
if (ret != GST_FLOW_OK) {
sent_all_sps_pps = FALSE;
GST_WARNING_OBJECT (basepayload, "Problem pushing SPS");
}
}
for (i = 0; i < rtph264pay->pps->len; i++) {
GstBuffer *pps_buf =
GST_BUFFER_CAST (g_ptr_array_index (rtph264pay->pps, i));
GST_DEBUG_OBJECT (rtph264pay, "inserting PPS in the stream");
/* resend PPS */
ret = gst_rtp_h264_pay_payload_nal (basepayload, gst_buffer_ref (pps_buf),
dts, pts, FALSE, FALSE, FALSE);
/* Not critical here; but throw a warning */
if (ret != GST_FLOW_OK) {
sent_all_sps_pps = FALSE;
GST_WARNING_OBJECT (basepayload, "Problem pushing PPS");
}
}
if (pts != -1 && sent_all_sps_pps)
rtph264pay->last_spspps =
gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME,
pts);
return ret;
}
/* @delta_unit: if %FALSE the first packet sent won't have the
* GST_BUFFER_FLAG_DELTA_UNIT flag.
* @discont: if %TRUE the first packet sent will have the
* GST_BUFFER_FLAG_DISCONT flag.
*/
static GstFlowReturn
gst_rtp_h264_pay_payload_nal (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
guint8 nalHeader;
guint8 nalType;
guint packet_len, payload_len, mtu;
GstBuffer *outbuf;
guint8 *payload;
GstBufferList *list = NULL;
gboolean send_spspps;
GstRTPBuffer rtp = { NULL };
guint size = gst_buffer_get_size (paybuf);
rtph264pay = GST_RTP_H264_PAY (basepayload);
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtph264pay);
gst_buffer_extract (paybuf, 0, &nalHeader, 1);
nalType = nalHeader & 0x1f;
GST_DEBUG_OBJECT (rtph264pay, "Processing Buffer with NAL TYPE=%d", nalType);
/* should set src caps before pushing stuff,
* and if we did not see enough SPS/PPS, that may not be the case */
if (G_UNLIKELY (!gst_pad_has_current_caps (GST_RTP_BASE_PAYLOAD_SRCPAD
(basepayload))))
gst_rtp_h264_pay_set_sps_pps (basepayload);
send_spspps = FALSE;
/* check if we need to emit an SPS/PPS now */
if (nalType == IDR_TYPE_ID && rtph264pay->spspps_interval > 0) {
if (rtph264pay->last_spspps != -1) {
guint64 diff;
GstClockTime running_time =
gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME,
pts);
GST_LOG_OBJECT (rtph264pay,
"now %" GST_TIME_FORMAT ", last SPS/PPS %" GST_TIME_FORMAT,
GST_TIME_ARGS (running_time),
GST_TIME_ARGS (rtph264pay->last_spspps));
/* calculate diff between last SPS/PPS in milliseconds */
if (running_time > rtph264pay->last_spspps)
diff = running_time - rtph264pay->last_spspps;
else
diff = 0;
GST_DEBUG_OBJECT (rtph264pay,
"interval since last SPS/PPS %" GST_TIME_FORMAT,
GST_TIME_ARGS (diff));
/* bigger than interval, queue SPS/PPS */
if (GST_TIME_AS_SECONDS (diff) >= rtph264pay->spspps_interval) {
GST_DEBUG_OBJECT (rtph264pay, "time to send SPS/PPS");
send_spspps = TRUE;
}
} else {
/* no know previous SPS/PPS time, send now */
GST_DEBUG_OBJECT (rtph264pay, "no previous SPS/PPS time, send now");
send_spspps = TRUE;
}
} else if (nalType == IDR_TYPE_ID && rtph264pay->spspps_interval == -1) {
GST_DEBUG_OBJECT (rtph264pay, "sending SPS/PPS before current IDR frame");
/* send SPS/PPS before every IDR frame */
send_spspps = TRUE;
}
if (send_spspps || rtph264pay->send_spspps) {
/* we need to send SPS/PPS now first. FIXME, don't use the pts for
* checking when we need to send SPS/PPS but convert to running_time first. */
rtph264pay->send_spspps = FALSE;
ret = gst_rtp_h264_pay_send_sps_pps (basepayload, rtph264pay, dts, pts);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (paybuf);
return ret;
}
}
packet_len = gst_rtp_buffer_calc_packet_len (size, 0, 0);
if (packet_len < mtu) {
/* will fit in one packet */
GST_DEBUG_OBJECT (basepayload,
"NAL Unit fit in one packet datasize=%d mtu=%d", size, mtu);
/* create buffer without payload containing only the RTP header
* (memory block at index 0) */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* only set the marker bit on packets containing access units */
if (IS_ACCESS_UNIT (nalType) && end_of_au) {
gst_rtp_buffer_set_marker (&rtp, 1);
}
/* timestamp the outbuffer */
GST_BUFFER_PTS (outbuf) = pts;
GST_BUFFER_DTS (outbuf) = dts;
if (!delta_unit)
/* Only the first packet sent should not have the flag */
delta_unit = TRUE;
else
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DELTA_UNIT);
if (discont) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
/* Only the first packet sent should have the flag */
discont = FALSE;
}
gst_rtp_buffer_unmap (&rtp);
/* insert payload memory block */
gst_rtp_copy_video_meta (rtph264pay, outbuf, paybuf);
outbuf = gst_buffer_append (outbuf, paybuf);
/* push the buffer to the next element */
ret = gst_rtp_base_payload_push (basepayload, outbuf);
} else {
/* fragmentation Units FU-A */
guint limitedSize;
int ii = 0, start = 1, end = 0, pos = 0;
GST_DEBUG_OBJECT (basepayload,
"NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", size, mtu);
pos++;
size--;
ret = GST_FLOW_OK;
GST_DEBUG_OBJECT (basepayload, "Using FU-A fragmentation for data size=%d",
size);
/* We keep 2 bytes for FU indicator and FU Header */
payload_len = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
list = gst_buffer_list_new_sized ((size / payload_len) + 1);
while (end == 0) {
limitedSize = size < payload_len ? size : payload_len;
GST_DEBUG_OBJECT (basepayload,
"Inside FU-A fragmentation limitedSize=%d iteration=%d", limitedSize,
ii);
/* use buffer lists
* create buffer without payload containing only the RTP header
* (memory block at index 0) */
outbuf = gst_rtp_buffer_new_allocate (2, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
GST_BUFFER_DTS (outbuf) = dts;
GST_BUFFER_PTS (outbuf) = pts;
payload = gst_rtp_buffer_get_payload (&rtp);
if (limitedSize == size) {
GST_DEBUG_OBJECT (basepayload, "end size=%d iteration=%d", size, ii);
end = 1;
}
if (IS_ACCESS_UNIT (nalType)) {
gst_rtp_buffer_set_marker (&rtp, end && end_of_au);
}
/* FU indicator */
payload[0] = (nalHeader & 0x60) | 28;
/* FU Header */
payload[1] = (start << 7) | (end << 6) | (nalHeader & 0x1f);
gst_rtp_buffer_unmap (&rtp);
/* insert payload memory block */
gst_rtp_copy_video_meta (rtph264pay, outbuf, paybuf);
gst_buffer_copy_into (outbuf, paybuf, GST_BUFFER_COPY_MEMORY, pos,
limitedSize);
if (!delta_unit)
/* Only the first packet sent should not have the flag */
delta_unit = TRUE;
else
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DELTA_UNIT);
if (discont) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
/* Only the first packet sent should have the flag */
discont = FALSE;
}
/* add the buffer to the buffer list */
gst_buffer_list_add (list, outbuf);
size -= limitedSize;
pos += limitedSize;
ii++;
start = 0;
}
ret = gst_rtp_base_payload_push_list (basepayload, list);
gst_buffer_unref (paybuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
gsize size;
guint nal_len, i;
GstMapInfo map;
const guint8 *data;
GstClockTime dts, pts;
GArray *nal_queue;
gboolean avc;
GstBuffer *paybuf = NULL;
gsize skip;
gboolean delayed_not_delta_unit = FALSE;
gboolean delayed_discont = FALSE;
rtph264pay = GST_RTP_H264_PAY (basepayload);
/* the input buffer contains one or more NAL units */
avc = rtph264pay->stream_format == GST_H264_STREAM_FORMAT_AVC;
if (avc) {
/* In AVC mode, there is no adapter, so nothign to flush */
if (buffer == NULL)
return GST_FLOW_OK;
gst_buffer_map (buffer, &map, GST_MAP_READ);
data = map.data;
size = map.size;
pts = GST_BUFFER_PTS (buffer);
dts = GST_BUFFER_DTS (buffer);
rtph264pay->delta_unit = GST_BUFFER_FLAG_IS_SET (buffer,
GST_BUFFER_FLAG_DELTA_UNIT);
rtph264pay->discont = GST_BUFFER_IS_DISCONT (buffer);
GST_DEBUG_OBJECT (basepayload, "got %" G_GSIZE_FORMAT " bytes", size);
} else {
dts = gst_adapter_prev_dts (rtph264pay->adapter, NULL);
pts = gst_adapter_prev_pts (rtph264pay->adapter, NULL);
if (buffer) {
if (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT)) {
if (gst_adapter_available (rtph264pay->adapter) == 0)
rtph264pay->delta_unit = FALSE;
else
/* This buffer contains a key frame but the adapter isn't empty. So
* we'll purge it first by sending a first packet and then the second
* one won't have the DELTA_UNIT flag. */
delayed_not_delta_unit = TRUE;
}
if (GST_BUFFER_IS_DISCONT (buffer)) {
if (gst_adapter_available (rtph264pay->adapter) == 0)
rtph264pay->discont = TRUE;
else
/* This buffer has the DISCONT flag but the adapter isn't empty. So
* we'll purge it first by sending a first packet and then the second
* one will have the DISCONT flag set. */
delayed_discont = TRUE;
}
if (!GST_CLOCK_TIME_IS_VALID (dts))
dts = GST_BUFFER_DTS (buffer);
if (!GST_CLOCK_TIME_IS_VALID (pts))
pts = GST_BUFFER_PTS (buffer);
gst_adapter_push (rtph264pay->adapter, buffer);
}
size = gst_adapter_available (rtph264pay->adapter);
/* Nothing to do here if the adapter is empty, e.g. on EOS */
if (size == 0)
return GST_FLOW_OK;
data = gst_adapter_map (rtph264pay->adapter, size);
GST_DEBUG_OBJECT (basepayload,
"got %" G_GSIZE_FORMAT " bytes (%" G_GSIZE_FORMAT ")", size,
buffer ? gst_buffer_get_size (buffer) : 0);
}
ret = GST_FLOW_OK;
/* now loop over all NAL units and put them in a packet
* FIXME, we should really try to pack multiple NAL units into one RTP packet
* if we can, especially for the config packets that wont't cause decoder
* latency. */
if (avc) {
guint nal_length_size;
gsize offset = 0;
nal_length_size = rtph264pay->nal_length_size;
while (size > nal_length_size) {
gint i;
gboolean end_of_au = FALSE;
nal_len = 0;
for (i = 0; i < nal_length_size; i++) {
nal_len = ((nal_len << 8) + data[i]);
}
/* skip the length bytes, make sure we don't run past the buffer size */
data += nal_length_size;
offset += nal_length_size;
size -= nal_length_size;
if (size >= nal_len) {
GST_DEBUG_OBJECT (basepayload, "got NAL of size %u", nal_len);
} else {
nal_len = size;
GST_DEBUG_OBJECT (basepayload, "got incomplete NAL of size %u",
nal_len);
}
/* If we're at the end of the buffer, then we're at the end of the
* access unit
*/
if (rtph264pay->alignment == GST_H264_ALIGNMENT_AU
&& size - nal_len <= nal_length_size) {
end_of_au = TRUE;
}
paybuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, offset,
nal_len);
ret =
gst_rtp_h264_pay_payload_nal (basepayload, paybuf, dts, pts,
end_of_au, rtph264pay->delta_unit, rtph264pay->discont);
if (!rtph264pay->delta_unit)
/* Only the first outgoing packet doesn't have the DELTA_UNIT flag */
rtph264pay->delta_unit = TRUE;
if (rtph264pay->discont)
/* Only the first outgoing packet have the DISCONT flag */
rtph264pay->discont = FALSE;
if (ret != GST_FLOW_OK)
break;
data += nal_len;
offset += nal_len;
size -= nal_len;
}
} else {
guint next;
gboolean update = FALSE;
/* get offset of first start code */
next = next_start_code (data, size);
/* skip to start code, if no start code is found, next will be size and we
* will not collect data. */
data += next;
size -= next;
nal_queue = rtph264pay->queue;
skip = next;
/* array must be empty when we get here */
g_assert (nal_queue->len == 0);
GST_DEBUG_OBJECT (basepayload,
"found first start at %u, bytes left %" G_GSIZE_FORMAT, next, size);
/* first pass to locate NALs and parse SPS/PPS */
while (size > 4) {
/* skip start code */
data += 3;
size -= 3;
/* 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.
* data is unchanged by the call to next_start_code()
*/
next = next_start_code (data, size);
/* nal or au aligned input needs no delaying until next time */
if (next == size && buffer != NULL &&
rtph264pay->alignment == GST_H264_ALIGNMENT_UNKNOWN) {
/* Didn't find the start of next NAL and it's not EOS,
* handle it next time */
break;
}
/* nal length is distance to next start code */
nal_len = next;
GST_DEBUG_OBJECT (basepayload, "found next start at %u of size %u", next,
nal_len);
if (rtph264pay->sprop_parameter_sets != NULL) {
/* explicitly set profile and sprop, use those */
if (rtph264pay->update_caps) {
if (!gst_rtp_base_payload_set_outcaps (basepayload,
"sprop-parameter-sets", G_TYPE_STRING,
rtph264pay->sprop_parameter_sets, NULL))
goto caps_rejected;
/* parse SPS and PPS from provided parameter set (for insertion) */
gst_rtp_h264_pay_parse_sprop_parameter_sets (rtph264pay);
rtph264pay->update_caps = FALSE;
GST_DEBUG ("outcaps update: sprop-parameter-sets=%s",
rtph264pay->sprop_parameter_sets);
}
} else {
/* We know our stream is a valid H264 NAL packet,
* go parse it for SPS/PPS to enrich the caps */
/* order: make sure to check nal */
update =
gst_rtp_h264_pay_decode_nal (rtph264pay, data, nal_len, dts, pts)
|| update;
}
/* move to next NAL packet */
data += nal_len;
size -= nal_len;
g_array_append_val (nal_queue, nal_len);
}
/* if has new SPS & PPS, update the output caps */
if (G_UNLIKELY (update))
if (!gst_rtp_h264_pay_set_sps_pps (basepayload))
goto caps_rejected;
/* second pass to payload and push */
if (nal_queue->len != 0)
gst_adapter_flush (rtph264pay->adapter, skip);
for (i = 0; i < nal_queue->len; i++) {
guint size;
gboolean end_of_au = FALSE;
nal_len = g_array_index (nal_queue, guint, i);
/* skip start code */
gst_adapter_flush (rtph264pay->adapter, 3);
/* Trim the end unless we're the last NAL in the stream.
* In case we're not at the end of the buffer we know the next block
* starts with 0x000001 so all the 0x00 bytes at the end of this one are
* trailing 0x0 that can be discarded */
size = nal_len;
data = gst_adapter_map (rtph264pay->adapter, size);
if (i + 1 != nal_queue->len || buffer != NULL)
for (; size > 1 && data[size - 1] == 0x0; size--)
/* skip */ ;
/* If it's the last nal unit we have in non-bytestream mode, we can
* assume it's the end of an access-unit
*
* FIXME: We need to wait until the next packet or EOS to
* actually payload the NAL so we can know if the current NAL is
* the last one of an access unit or not if we are in bytestream mode
*/
if ((rtph264pay->alignment == GST_H264_ALIGNMENT_AU || buffer == NULL) &&
i == nal_queue->len - 1)
end_of_au = TRUE;
paybuf = gst_adapter_take_buffer (rtph264pay->adapter, size);
g_assert (paybuf);
/* put the data in one or more RTP packets */
ret =
gst_rtp_h264_pay_payload_nal (basepayload, paybuf, dts, pts,
end_of_au, rtph264pay->delta_unit, rtph264pay->discont);
if (delayed_not_delta_unit) {
rtph264pay->delta_unit = FALSE;
delayed_not_delta_unit = FALSE;
} else {
/* Only the first outgoing packet doesn't have the DELTA_UNIT flag */
rtph264pay->delta_unit = TRUE;
}
if (delayed_discont) {
rtph264pay->discont = TRUE;
delayed_discont = FALSE;
} else {
/* Only the first outgoing packet have the DISCONT flag */
rtph264pay->discont = FALSE;
}
if (ret != GST_FLOW_OK) {
break;
}
/* move to next NAL packet */
/* Skips the trailing zeros */
gst_adapter_flush (rtph264pay->adapter, nal_len - size);
}
g_array_set_size (nal_queue, 0);
}
done:
if (avc) {
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
} else {
gst_adapter_unmap (rtph264pay->adapter);
}
return ret;
caps_rejected:
{
GST_WARNING_OBJECT (basepayload, "Could not set outcaps");
g_array_set_size (nal_queue, 0);
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
}
static gboolean
gst_rtp_h264_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
{
gboolean res;
const GstStructure *s;
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (payload);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_adapter_clear (rtph264pay->adapter);
break;
case GST_EVENT_CUSTOM_DOWNSTREAM:
s = gst_event_get_structure (event);
if (gst_structure_has_name (s, "GstForceKeyUnit")) {
gboolean resend_codec_data;
if (gst_structure_get_boolean (s, "all-headers",
&resend_codec_data) && resend_codec_data)
rtph264pay->send_spspps = TRUE;
}
break;
case GST_EVENT_EOS:
{
/* call handle_buffer with NULL to flush last NAL from adapter
* in byte-stream mode
*/
gst_rtp_h264_pay_handle_buffer (payload, NULL);
break;
}
case GST_EVENT_STREAM_START:
GST_DEBUG_OBJECT (rtph264pay, "New stream detected => Clear SPS and PPS");
gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
break;
default:
break;
}
res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
return res;
}
static GstStateChangeReturn
gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
rtph264pay->send_spspps = FALSE;
gst_adapter_clear (rtph264pay->adapter);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
rtph264pay->last_spspps = -1;
gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
break;
default:
break;
}
return ret;
}
static void
gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
switch (prop_id) {
case PROP_SPROP_PARAMETER_SETS:
g_free (rtph264pay->sprop_parameter_sets);
rtph264pay->sprop_parameter_sets = g_value_dup_string (value);
rtph264pay->update_caps = TRUE;
break;
case PROP_CONFIG_INTERVAL:
rtph264pay->spspps_interval = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpH264Pay *rtph264pay;
rtph264pay = GST_RTP_H264_PAY (object);
switch (prop_id) {
case PROP_SPROP_PARAMETER_SETS:
g_value_set_string (value, rtph264pay->sprop_parameter_sets);
break;
case PROP_CONFIG_INTERVAL:
g_value_set_int (value, rtph264pay->spspps_interval);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_rtp_h264_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtph264pay",
GST_RANK_SECONDARY, GST_TYPE_RTP_H264_PAY);
}
Reference in a new issue View git blame Copy permalink