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gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtph264pay.c

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/* 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 "gstrtpelements.h"
#include "gstrtph264pay.h"
#include "gstrtputils.h"
#include "gstbuffermemory.h"
#define IDR_TYPE_ID 5
#define SPS_TYPE_ID 7
#define PPS_TYPE_ID 8
#define AUD_TYPE_ID 9
#define STAP_A_TYPE_ID 24
#define FU_A_TYPE_ID 28
GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug);
#define GST_CAT_DEFAULT (rtph264pay_debug)
#define GST_TYPE_RTP_H264_AGGREGATE_MODE \
(gst_rtp_h264_aggregate_mode_get_type ())
static GType
gst_rtp_h264_aggregate_mode_get_type (void)
{
static GType type = 0;
static const GEnumValue values[] = {
{GST_RTP_H264_AGGREGATE_NONE, "Do not aggregate NAL units", "none"},
{GST_RTP_H264_AGGREGATE_ZERO_LATENCY,
"Aggregate NAL units until a VCL unit is included", "zero-latency"},
{GST_RTP_H264_AGGREGATE_MAX_STAP,
"Aggregate all NAL units with the same timestamp (adds one frame of"
" latency)", "max-stap"},
{0, NULL, NULL},
};
if (!type) {
type = g_enum_register_static ("GstRtpH264AggregateMode", values);
}
return type;
}
/* 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
#define DEFAULT_AGGREGATE_MODE GST_RTP_H264_AGGREGATE_NONE
enum
{
PROP_0,
PROP_SPROP_PARAMETER_SETS,
PROP_CONFIG_INTERVAL,
PROP_AGGREGATE_MODE,
};
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);
static gboolean gst_rtp_h264_pay_src_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static void gst_rtp_h264_pay_reset_bundle (GstRtpH264Pay * rtph264pay);
#define gst_rtp_h264_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpH264Pay, gst_rtp_h264_pay, GST_TYPE_RTP_BASE_PAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtph264pay, "rtph264pay",
GST_RANK_SECONDARY, GST_TYPE_RTP_H264_PAY, rtp_element_init (plugin));
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)
);
/**
* GstRtpH264Pay:aggregate-mode
*
* Bundle suitable SPS/PPS NAL units into STAP-A aggregate packets.
*
* This can potentially reduce RTP packetization overhead but not all
* RTP implementations handle it correctly.
*
* For best compatibility, it is recommended to set this to "none" (the
* default) for RTSP and for WebRTC to "zero-latency".
*
* Since: 1.18
*/
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_AGGREGATE_MODE,
g_param_spec_enum ("aggregate-mode",
"Attempt to use aggregate packets",
"Bundle suitable SPS/PPS NAL units into STAP-A "
"aggregate packets",
GST_TYPE_RTP_H264_AGGREGATE_MODE,
DEFAULT_AGGREGATE_MODE, 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");
gst_type_mark_as_plugin_api (GST_TYPE_RTP_H264_AGGREGATE_MODE, 0);
}
static void
gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay)
{
rtph264pay->queue = g_array_new (FALSE, FALSE, sizeof (guint));
rtph264pay->profile_level = 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->aggregate_mode = DEFAULT_AGGREGATE_MODE;
rtph264pay->delta_unit = FALSE;
rtph264pay->discont = FALSE;
rtph264pay->adapter = gst_adapter_new ();
gst_pad_set_query_function (GST_RTP_BASE_PAYLOAD_SRCPAD (rtph264pay),
gst_rtp_h264_pay_src_query);
}
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);
gst_rtp_h264_pay_reset_bundle (rtph264pay);
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;
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 if ((profile = gst_structure_get_string (s, "profile"))) {
gst_structure_set (new_s, "profile", G_TYPE_STRING, profile, 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;
}
static gboolean
gst_rtp_h264_pay_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (parent);
if (GST_QUERY_TYPE (query) == GST_QUERY_LATENCY) {
gboolean retval;
gboolean live;
GstClockTime min_latency, max_latency;
retval = gst_pad_query_default (pad, parent, query);
if (!retval)
return retval;
if (rtph264pay->stream_format == GST_H264_STREAM_FORMAT_UNKNOWN ||
rtph264pay->alignment == GST_H264_ALIGNMENT_UNKNOWN)
return FALSE;
gst_query_parse_latency (query, &live, &min_latency, &max_latency);
if (rtph264pay->aggregate_mode == GST_RTP_H264_AGGREGATE_MAX_STAP &&
rtph264pay->alignment != GST_H264_ALIGNMENT_AU && rtph264pay->fps_num) {
GstClockTime one_frame = gst_util_uint64_scale_int (GST_SECOND,
rtph264pay->fps_denum, rtph264pay->fps_num);
min_latency += one_frame;
max_latency += one_frame;
gst_query_set_latency (query, live, min_latency, max_latency);
}
return TRUE;
}
return gst_pad_query_default (pad, parent, query);
}
/* 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)
{
GstStructure *s = gst_structure_new_empty ("unused");
GstRtpH264Pay *payloader = GST_RTP_H264_PAY (basepayload);
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)) {
gchar *profile_level;
gst_structure_set (s,
"packetization-mode", G_TYPE_STRING, "1",
"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
if (payloader->profile_level != 0) {
guint8 sps[2] = {
payloader->profile_level >> 16,
payloader->profile_level >> 8,
};
profile_level =
g_strdup_printf ("%06x", payloader->profile_level & 0xffffff);
gst_structure_set (s,
"profile-level-id", G_TYPE_STRING, profile_level,
"profile", G_TYPE_STRING, gst_codec_utils_h264_get_profile (sps, 2),
NULL);
g_free (profile_level);
}
/* combine into output caps */
res = gst_rtp_base_payload_set_outcaps_structure (basepayload, s);
} else {
res = gst_rtp_base_payload_set_outcaps (basepayload, NULL);
}
gst_structure_free (s);
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;
}
if (!gst_structure_get_fraction (str, "framerate", &rtph264pay->fps_num,
&rtph264pay->fps_denum))
rtph264pay->fps_num = rtph264pay->fps_denum = 0;
/* 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_level = (data[1] << 16) | (data[2] << 8) | data[3];
GST_DEBUG_OBJECT (rtph264pay, "profile %06x", rtph264pay->profile_level);
/* 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_payload_nal_single (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont);
static GstFlowReturn
gst_rtp_h264_pay_payload_nal_fragment (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont, guint8 nal_header);
static GstFlowReturn
gst_rtp_h264_pay_payload_nal_bundle (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont, guint8 nal_header);
static GstFlowReturn
gst_rtp_h264_pay_send_sps_pps (GstRTPBasePayload * basepayload,
GstClockTime dts, GstClockTime pts, gboolean delta_unit, gboolean discont)
{
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (basepayload);
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, delta_unit, discont);
/* 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, TRUE, 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;
guint8 nal_header, nal_type;
gboolean first_slice = FALSE;
gboolean send_spspps;
guint size;
rtph264pay = GST_RTP_H264_PAY (basepayload);
size = gst_buffer_get_size (paybuf);
gst_buffer_extract (paybuf, 0, &nal_header, 1);
nal_type = nal_header & 0x1f;
/* These payload type are reserved for STAP-A, STAP-B, MTAP16, and MTAP24
* as internally used NAL types */
switch (nal_type) {
case 24:
case 25:
case 26:
case 27:
GST_WARNING_OBJECT (rtph264pay, "Ignoring reserved NAL TYPE=%d",
nal_type);
gst_buffer_unref (paybuf);
return GST_FLOW_OK;
default:
break;
}
GST_DEBUG_OBJECT (rtph264pay,
"payloading NAL Unit: datasize=%u type=%d pts=%" GST_TIME_FORMAT,
size, nal_type, GST_TIME_ARGS (pts));
/* 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;
if (nal_type == IDR_TYPE_ID) {
guint8 first_mb_in_slice;
gst_buffer_extract (paybuf, 1, &first_mb_in_slice, 1);
/* 'first_mb_in_slice' specifies the address of the first macroblock
* in the slice. if 'first_mb_in_slice' is 0 (note that it's exp golomb
* code), the current slice is the first slice of the frame */
first_slice = ((first_mb_in_slice >> 7) & 0x01) == 1;
}
/* check if we need to emit an SPS/PPS now */
if (first_slice && nal_type == 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 (first_slice && nal_type == 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. */
GstFlowReturn ret;
rtph264pay->send_spspps = FALSE;
ret = gst_rtp_h264_pay_send_sps_pps (basepayload, dts, pts, delta_unit,
discont);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (paybuf);
return ret;
}
delta_unit = TRUE;
discont = FALSE;
}
if (rtph264pay->aggregate_mode != GST_RTP_H264_AGGREGATE_NONE)
return gst_rtp_h264_pay_payload_nal_bundle (basepayload, paybuf, dts, pts,
end_of_au, delta_unit, discont, nal_header);
return gst_rtp_h264_pay_payload_nal_fragment (basepayload, paybuf, dts, pts,
end_of_au, delta_unit, discont, nal_header);
}
static GstFlowReturn
gst_rtp_h264_pay_payload_nal_fragment (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont, guint8 nal_header)
{
GstRtpH264Pay *rtph264pay;
guint mtu, size, max_fragment_size, max_fragments, ii, pos;
GstBuffer *outbuf;
guint8 *payload;
GstBufferList *list = NULL;
GstRTPBuffer rtp = { NULL };
rtph264pay = GST_RTP_H264_PAY (basepayload);
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtph264pay);
size = gst_buffer_get_size (paybuf);
if (gst_rtp_buffer_calc_packet_len (size, 0, 0) <= mtu) {
/* We don't need to fragment this packet */
GST_DEBUG_OBJECT (rtph264pay,
"sending NAL Unit: datasize=%u mtu=%u", size, mtu);
return gst_rtp_h264_pay_payload_nal_single (basepayload, paybuf, dts, pts,
end_of_au, delta_unit, discont);
}
GST_DEBUG_OBJECT (basepayload,
"using FU-A fragmentation for NAL Unit: datasize=%u mtu=%u", size, mtu);
/* We keep 2 bytes for FU indicator and FU Header */
max_fragment_size = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
max_fragments = (size + max_fragment_size - 2) / max_fragment_size;
list = gst_buffer_list_new_sized (max_fragments);
/* Start at the NALU payload */
for (pos = 1, ii = 0; pos < size; pos += max_fragment_size, ii++) {
guint remaining, fragment_size;
gboolean first_fragment, last_fragment;
remaining = size - pos;
fragment_size = MIN (remaining, max_fragment_size);
first_fragment = (pos == 1);
last_fragment = (remaining <= max_fragment_size);
GST_DEBUG_OBJECT (basepayload,
"creating FU-A packet %u/%u, size %u",
ii + 1, max_fragments, fragment_size);
/* use buffer lists
* create buffer without payload containing only the RTP header
* (memory block at index 0) */
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 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 it's the last fragment and the end of this au, mark the end of
* slice */
gst_rtp_buffer_set_marker (&rtp, last_fragment && end_of_au);
if (last_fragment && end_of_au)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MARKER);
/* FU indicator */
payload[0] = (nal_header & 0x60) | FU_A_TYPE_ID;
/* FU Header */
payload[1] = (first_fragment << 7) | (last_fragment << 6) |
(nal_header & 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,
fragment_size);
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);
}
GST_DEBUG_OBJECT (rtph264pay,
"sending FU-A fragments: n=%u datasize=%u mtu=%u", ii, size, mtu);
gst_buffer_unref (paybuf);
return gst_rtp_base_payload_push_list (basepayload, list);
}
static GstFlowReturn
gst_rtp_h264_pay_payload_nal_single (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont)
{
GstRtpH264Pay *rtph264pay;
GstBuffer *outbuf;
GstRTPBuffer rtp = { NULL };
rtph264pay = GST_RTP_H264_PAY (basepayload);
/* create buffer without payload containing only the RTP header
* (memory block at index 0) */
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* Mark the end of a frame */
gst_rtp_buffer_set_marker (&rtp, end_of_au);
if (end_of_au)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MARKER);
/* timestamp the outbuffer */
GST_BUFFER_PTS (outbuf) = pts;
GST_BUFFER_DTS (outbuf) = dts;
if (delta_unit)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DELTA_UNIT);
if (discont)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
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 */
return gst_rtp_base_payload_push (basepayload, outbuf);
}
static void
gst_rtp_h264_pay_reset_bundle (GstRtpH264Pay * rtph264pay)
{
g_clear_pointer (&rtph264pay->bundle, gst_buffer_list_unref);
rtph264pay->bundle_size = 0;
rtph264pay->bundle_contains_vcl = FALSE;
}
static GstFlowReturn
gst_rtp_h264_pay_send_bundle (GstRtpH264Pay * rtph264pay, gboolean end_of_au)
{
GstRTPBasePayload *basepayload;
GstBufferList *bundle;
guint length, bundle_size;
GstBuffer *first, *outbuf;
GstClockTime dts, pts;
gboolean delta, discont;
bundle_size = rtph264pay->bundle_size;
if (bundle_size == 0) {
GST_DEBUG_OBJECT (rtph264pay, "no bundle, nothing to send");
return GST_FLOW_OK;
}
basepayload = GST_RTP_BASE_PAYLOAD (rtph264pay);
bundle = rtph264pay->bundle;
length = gst_buffer_list_length (bundle);
first = gst_buffer_list_get (bundle, 0);
dts = GST_BUFFER_DTS (first);
pts = GST_BUFFER_PTS (first);
delta = GST_BUFFER_FLAG_IS_SET (first, GST_BUFFER_FLAG_DELTA_UNIT);
discont = GST_BUFFER_FLAG_IS_SET (first, GST_BUFFER_FLAG_DISCONT);
if (length == 1) {
/* Push unaggregated NALU */
outbuf = gst_buffer_ref (first);
GST_DEBUG_OBJECT (rtph264pay,
"sending NAL Unit unaggregated: datasize=%u", bundle_size - 2);
} else {
guint8 stap_header;
guint i;
outbuf = gst_buffer_new_allocate (NULL, sizeof stap_header, NULL);
stap_header = STAP_A_TYPE_ID;
for (i = 0; i < length; i++) {
GstBuffer *buf = gst_buffer_list_get (bundle, i);
guint8 nal_header;
GstMemory *size_header;
GstMapInfo map;
gst_buffer_extract (buf, 0, &nal_header, sizeof nal_header);
/* Propagate F bit */
if ((nal_header & 0x80))
stap_header |= 0x80;
/* Select highest nal_ref_idc */
if ((nal_header & 0x60) > (stap_header & 0x60))
stap_header = (stap_header & 0x9f) | (nal_header & 0x60);
/* append NALU size */
size_header = gst_allocator_alloc (NULL, 2, NULL);
gst_memory_map (size_header, &map, GST_MAP_WRITE);
GST_WRITE_UINT16_BE (map.data, gst_buffer_get_size (buf));
gst_memory_unmap (size_header, &map);
gst_buffer_append_memory (outbuf, size_header);
/* append NALU data */
outbuf = gst_buffer_append (outbuf, gst_buffer_ref (buf));
}
gst_buffer_fill (outbuf, 0, &stap_header, sizeof stap_header);
GST_DEBUG_OBJECT (rtph264pay,
"sending STAP-A bundle: n=%u header=%02x datasize=%u",
length, stap_header, bundle_size);
}
gst_rtp_h264_pay_reset_bundle (rtph264pay);
return gst_rtp_h264_pay_payload_nal_single (basepayload, outbuf, dts, pts,
end_of_au, delta, discont);
}
static GstFlowReturn
gst_rtp_h264_pay_payload_nal_bundle (GstRTPBasePayload * basepayload,
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
gboolean delta_unit, gboolean discont, guint8 nal_header)
{
GstRtpH264Pay *rtph264pay;
GstFlowReturn ret;
guint mtu, pay_size, bundle_size;
GstBufferList *bundle;
guint8 nal_type;
gboolean start_of_au;
rtph264pay = GST_RTP_H264_PAY (basepayload);
nal_type = nal_header & 0x1f;
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtph264pay);
pay_size = 2 + gst_buffer_get_size (paybuf);
bundle = rtph264pay->bundle;
start_of_au = FALSE;
if (bundle) {
GstBuffer *first = gst_buffer_list_get (bundle, 0);
if (nal_type == AUD_TYPE_ID) {
GST_DEBUG_OBJECT (rtph264pay, "found access delimiter");
start_of_au = TRUE;
} else if (discont) {
GST_DEBUG_OBJECT (rtph264pay, "found discont");
start_of_au = TRUE;
} else if (GST_BUFFER_PTS (first) != pts || GST_BUFFER_DTS (first) != dts) {
GST_DEBUG_OBJECT (rtph264pay, "found timestamp mismatch");
start_of_au = TRUE;
}
}
if (start_of_au) {
GST_DEBUG_OBJECT (rtph264pay, "sending bundle before start of AU");
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
if (ret != GST_FLOW_OK)
goto out;
bundle = NULL;
}
bundle_size = 1 + pay_size;
if (gst_rtp_buffer_calc_packet_len (bundle_size, 0, 0) > mtu) {
GST_DEBUG_OBJECT (rtph264pay, "NAL Unit cannot fit in a bundle");
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, FALSE);
if (ret != GST_FLOW_OK)
goto out;
return gst_rtp_h264_pay_payload_nal_fragment (basepayload, paybuf, dts, pts,
end_of_au, delta_unit, discont, nal_header);
}
bundle_size = rtph264pay->bundle_size + pay_size;
if (gst_rtp_buffer_calc_packet_len (bundle_size, 0, 0) > mtu) {
GST_DEBUG_OBJECT (rtph264pay,
"bundle overflows, sending: bundlesize=%u datasize=2+%u mtu=%u",
rtph264pay->bundle_size, pay_size - 2, mtu);
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, FALSE);
if (ret != GST_FLOW_OK)
goto out;
bundle = NULL;
}
if (!bundle) {
GST_DEBUG_OBJECT (rtph264pay, "creating new STAP-A aggregate");
bundle = rtph264pay->bundle = gst_buffer_list_new ();
bundle_size = rtph264pay->bundle_size = 1;
rtph264pay->bundle_contains_vcl = FALSE;
}
GST_DEBUG_OBJECT (rtph264pay,
"bundling NAL Unit: bundlesize=%u datasize=2+%u mtu=%u",
rtph264pay->bundle_size, pay_size - 2, mtu);
paybuf = gst_buffer_make_writable (paybuf);
GST_BUFFER_PTS (paybuf) = pts;
GST_BUFFER_DTS (paybuf) = dts;
if (delta_unit)
GST_BUFFER_FLAG_SET (paybuf, GST_BUFFER_FLAG_DELTA_UNIT);
else
GST_BUFFER_FLAG_UNSET (paybuf, GST_BUFFER_FLAG_DELTA_UNIT);
if (discont)
GST_BUFFER_FLAG_SET (paybuf, GST_BUFFER_FLAG_DISCONT);
else
GST_BUFFER_FLAG_UNSET (paybuf, GST_BUFFER_FLAG_DISCONT);
gst_buffer_list_add (bundle, gst_buffer_ref (paybuf));
rtph264pay->bundle_size += pay_size;
ret = GST_FLOW_OK;
if ((nal_type >= 1 && nal_type <= 5) || nal_type == 14 ||
(nal_type >= 20 && nal_type <= 23))
rtph264pay->bundle_contains_vcl = TRUE;
if (end_of_au) {
GST_DEBUG_OBJECT (rtph264pay, "sending bundle at end of AU");
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
}
out:
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;
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;
gboolean marker = FALSE;
gboolean draining = (buffer == NULL);
rtph264pay = GST_RTP_H264_PAY (basepayload);
/* the input buffer contains one or more NAL units */
avc = rtph264pay->stream_format == GST_H264_STREAM_FORMAT_AVC;
ret = GST_FLOW_OK;
if (avc) {
GstBufferMemoryMap memory;
gsize remaining_buffer_size;
guint nal_length_size;
gsize offset = 0;
/* In AVC mode, there is no adapter, so nothing to drain */
if (draining)
return GST_FLOW_OK;
gst_buffer_memory_map (buffer, &memory);
remaining_buffer_size = gst_buffer_get_size (buffer);
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);
marker = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_MARKER);
GST_DEBUG_OBJECT (basepayload, "got %" G_GSIZE_FORMAT " bytes",
remaining_buffer_size);
nal_length_size = rtph264pay->nal_length_size;
while (remaining_buffer_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) + *memory.data;
if (!gst_buffer_memory_advance_bytes (&memory, 1))
break;
}
offset += nal_length_size;
remaining_buffer_size -= nal_length_size;
if (remaining_buffer_size >= nal_len) {
GST_DEBUG_OBJECT (basepayload, "got NAL of size %u", nal_len);
} else {
nal_len = remaining_buffer_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 (remaining_buffer_size - nal_len <= nal_length_size) {
if (rtph264pay->alignment == GST_H264_ALIGNMENT_AU || marker)
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;
/* Skip current nal. If it is split over multiple GstMemory
* advance_bytes () will switch to the correct GstMemory. The payloader
* does not access those bytes directly but uses gst_buffer_copy_region ()
* to create a sub-buffer referencing the nal instead */
if (!gst_buffer_memory_advance_bytes (&memory, nal_len))
break;
offset += nal_len;
remaining_buffer_size -= nal_len;
}
gst_buffer_memory_unmap (&memory);
gst_buffer_unref (buffer);
} else {
guint next;
gboolean update = FALSE;
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;
}
marker = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_MARKER);
gst_adapter_push (rtph264pay->adapter, buffer);
buffer = NULL;
}
/* We want to use the first TS used to construct the following NAL */
dts = gst_adapter_prev_dts (rtph264pay->adapter, NULL);
pts = gst_adapter_prev_pts (rtph264pay->adapter, NULL);
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", size);
/* 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 && !draining &&
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 || !draining)
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 (i == nal_queue->len - 1) {
if (rtph264pay->alignment == GST_H264_ALIGNMENT_AU ||
marker || draining)
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);
}
if (ret == GST_FLOW_OK && rtph264pay->bundle_size > 0 &&
rtph264pay->aggregate_mode == GST_RTP_H264_AGGREGATE_ZERO_LATENCY &&
rtph264pay->bundle_contains_vcl) {
GST_DEBUG_OBJECT (rtph264pay, "sending bundle at end incoming packet");
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, FALSE);
}
done:
if (!avc) {
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);
GstFlowReturn ret = GST_FLOW_OK;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_adapter_clear (rtph264pay->adapter);
gst_rtp_h264_pay_reset_bundle (rtph264pay);
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);
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
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);
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
break;
default:
break;
}
if (ret != GST_FLOW_OK)
return FALSE;
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);
gst_rtp_h264_pay_reset_bundle (rtph264pay);
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;
case PROP_AGGREGATE_MODE:
rtph264pay->aggregate_mode = g_value_get_enum (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;
case PROP_AGGREGATE_MODE:
g_value_set_enum (value, rtph264pay->aggregate_mode);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
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