/* 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include #include "gstrtph264pay.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, PROP_LAST }; #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_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CONFIG_INTERVAL, g_param_spec_uint ("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)", 0, 3600, DEFAULT_CONFIG_INTERVAL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS) ); gobject_class->finalize = gst_rtp_h264_pay_finalize; gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_h264_pay_src_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&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 "); 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 = NULL; rtph264pay->pps = NULL; rtph264pay->last_spspps = -1; rtph264pay->spspps_interval = DEFAULT_CONFIG_INTERVAL; rtph264pay->adapter = gst_adapter_new (); } static void gst_rtp_h264_pay_clear_sps_pps (GstRtpH264Pay * rtph264pay) { g_list_foreach (rtph264pay->sps, (GFunc) gst_mini_object_unref, NULL); g_list_free (rtph264pay->sps); rtph264pay->sps = NULL; g_list_foreach (rtph264pay->pps, (GFunc) gst_mini_object_unref, NULL); g_list_free (rtph264pay->pps); rtph264pay->pps = NULL; } static void gst_rtp_h264_pay_finalize (GObject * object) { GstRtpH264Pay *rtph264pay; rtph264pay = GST_RTP_H264_PAY (object); g_array_free (rtph264pay->queue, TRUE); gst_rtp_h264_pay_clear_sps_pps (rtph264pay); 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), filter); 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: 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; GList *walk; GString *sprops; guint count; gboolean res; GstMapInfo map; sprops = g_string_new (""); count = 0; /* build the sprop-parameter-sets */ for (walk = payloader->sps; walk; walk = g_list_next (walk)) { GstBuffer *sps_buf = GST_BUFFER_CAST (walk->data); 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 (walk = payloader->pps; walk; walk = g_list_next (walk)) { GstBuffer *pps_buf = GST_BUFFER_CAST (walk->data); 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)) { /* 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, "sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL); g_free (profile); } 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, "bytestream")) 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); rtph264pay->sps = g_list_append (rtph264pay->sps, 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); rtph264pay->pps = g_list_append (rtph264pay->pps, pps_buf); data += nal_size; size -= nal_size; } gst_buffer_unmap (buffer, &map); /* and update the caps with the collected data */ if (!gst_rtp_h264_pay_set_sps_pps (basepayload)) goto set_sps_pps_failed; } 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, num_sps, num_pps; 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); num_sps = num_pps = 0; for (i = 0; params[i]; i++) { gsize nal_len; GstMapInfo map; guint8 *nalp; guint save = 0; gint state = 0; guint8 nal_type; 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); nal_type = nalp[0]; gst_buffer_unmap (buf, &map); gst_buffer_resize (buf, 0, nal_len); if (!nal_len) { gst_buffer_unref (buf); continue; } /* append to the right list */ if ((nal_type & 0x1f) == 7) { GST_DEBUG_OBJECT (rtph264pay, "adding param %d as SPS %d", i, num_sps); rtph264pay->sps = g_list_append (rtph264pay->sps, buf); num_sps++; } else { GST_DEBUG_OBJECT (rtph264pay, "adding param %d as PPS %d", i, num_pps); rtph264pay->pps = g_list_append (rtph264pay->pps, buf); num_pps++; } } 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) { const guint8 *sps = NULL, *pps = NULL; guint sps_len = 0, pps_len = 0; guint8 header, type; guint len; gboolean updated; /* default is no update */ updated = FALSE; GST_DEBUG ("NAL payload len=%u", size); len = size; header = data[0]; type = header & 0x1f; /* keep sps & pps separately so that we can update either one * independently. We also 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) { /* encode the entire SPS NAL in base64 */ GST_DEBUG ("Found SPS %x %x %x Len=%u", (header >> 7), (header >> 5) & 3, type, len); sps = data; sps_len = len; /* remember when we last saw SPS */ if (pts != -1) payloader->last_spspps = pts; } else if (PPS_TYPE_ID == type) { /* encoder the entire PPS NAL in base64 */ GST_DEBUG ("Found PPS %x %x %x Len = %u", (header >> 7), (header >> 5) & 3, type, len); pps = data; pps_len = len; /* remember when we last saw PPS */ if (pts != -1) payloader->last_spspps = pts; } else { GST_DEBUG ("NAL: %x %x %x Len = %u", (header >> 7), (header >> 5) & 3, type, len); } /* 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) { GstBuffer *sps_buf; if (payloader->sps != NULL) { sps_buf = GST_BUFFER_CAST (payloader->sps->data); if (gst_buffer_memcmp (sps_buf, 0, sps, sps_len)) { /* something changed, update */ payloader->profile = (sps[1] << 16) + (sps[2] << 8) + sps[3]; GST_DEBUG ("Profile level IDC = %06x", payloader->profile); updated = TRUE; } } else { /* no previous SPS, update */ updated = TRUE; } if (updated) { sps_buf = gst_buffer_new_and_alloc (sps_len); gst_buffer_fill (sps_buf, 0, sps, sps_len); if (payloader->sps) { /* replace old buffer */ gst_buffer_unref (payloader->sps->data); payloader->sps->data = sps_buf; } else { /* add new buffer */ payloader->sps = g_list_prepend (payloader->sps, sps_buf); } } } if (pps_len > 0) { GstBuffer *pps_buf; if (payloader->pps != NULL) { pps_buf = GST_BUFFER_CAST (payloader->pps->data); if (gst_buffer_memcmp (pps_buf, 0, pps, pps_len)) { /* something changed, update */ updated = TRUE; } } else { /* no previous SPS, update */ updated = TRUE; } if (updated) { pps_buf = gst_buffer_new_and_alloc (pps_len); gst_buffer_fill (pps_buf, 0, pps, pps_len); if (payloader->pps) { /* replace old buffer */ gst_buffer_unref (payloader->pps->data); payloader->pps->data = pps_buf; } else { /* add new buffer */ payloader->pps = g_list_prepend (payloader->pps, pps_buf); } } } return updated; } static GstFlowReturn gst_rtp_h264_pay_payload_nal (GstRTPBasePayload * basepayload, const guint8 * data, guint size, GstClockTime dts, GstClockTime pts, gboolean end_of_au); static GstFlowReturn gst_rtp_h264_pay_send_sps_pps (GstRTPBasePayload * basepayload, GstRtpH264Pay * rtph264pay, GstClockTime dts, GstClockTime pts) { GstFlowReturn ret = GST_FLOW_OK; GList *walk; GstMapInfo map; for (walk = rtph264pay->sps; walk; walk = g_list_next (walk)) { GstBuffer *sps_buf = GST_BUFFER_CAST (walk->data); GST_DEBUG_OBJECT (rtph264pay, "inserting SPS in the stream"); /* resend SPS */ gst_buffer_map (sps_buf, &map, GST_MAP_READ); ret = gst_rtp_h264_pay_payload_nal (basepayload, map.data, map.size, dts, pts, FALSE); gst_buffer_unmap (sps_buf, &map); /* Not critical here; but throw a warning */ if (ret != GST_FLOW_OK) GST_WARNING ("Problem pushing SPS"); } for (walk = rtph264pay->pps; walk; walk = g_list_next (walk)) { GstBuffer *pps_buf = GST_BUFFER_CAST (walk->data); GST_DEBUG_OBJECT (rtph264pay, "inserting PPS in the stream"); /* resend PPS */ gst_buffer_map (pps_buf, &map, GST_MAP_READ); ret = gst_rtp_h264_pay_payload_nal (basepayload, map.data, map.size, dts, pts, FALSE); gst_buffer_unmap (pps_buf, &map); /* Not critical here; but throw a warning */ if (ret != GST_FLOW_OK) GST_WARNING ("Problem pushing PPS"); } if (pts != -1) rtph264pay->last_spspps = pts; return ret; } static GstFlowReturn gst_rtp_h264_pay_payload_nal (GstRTPBasePayload * basepayload, const guint8 * data, guint size, GstClockTime dts, GstClockTime pts, gboolean end_of_au) { GstRtpH264Pay *rtph264pay; GstFlowReturn ret; guint8 nalType; guint packet_len, payload_len, mtu; GstBuffer *outbuf; guint8 *payload; GstBufferList *list = NULL; gboolean send_spspps; GstRTPBuffer rtp = { NULL }; rtph264pay = GST_RTP_H264_PAY (basepayload); mtu = GST_RTP_BASE_PAYLOAD_MTU (rtph264pay); nalType = data[0] & 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; GST_LOG_OBJECT (rtph264pay, "now %" GST_TIME_FORMAT ", last SPS/PPS %" GST_TIME_FORMAT, GST_TIME_ARGS (pts), GST_TIME_ARGS (rtph264pay->last_spspps)); /* calculate diff between last SPS/PPS in milliseconds */ if (pts > rtph264pay->last_spspps) diff = pts - 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; } } 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) return ret; } packet_len = gst_rtp_buffer_calc_packet_len (size, 0, 0); if (packet_len < mtu) { GST_DEBUG_OBJECT (basepayload, "NAL Unit fit in one packet datasize=%d mtu=%d", size, mtu); /* will fit in one packet */ /* use buffer lists * 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; /* insert payload memory block */ gst_buffer_append_memory (outbuf, gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY, (guint8*) data, size, 0, size, NULL, NULL)); list = gst_buffer_list_new (); /* add the buffer to the buffer list */ gst_buffer_list_add (list, outbuf); gst_rtp_buffer_unmap (&rtp); /* push the list to the next element in the pipe */ ret = gst_rtp_base_payload_push_list (basepayload, list); } else { /* fragmentation Units FU-A */ guint8 nalHeader; 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); nalHeader = *data; 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 (); 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); /* insert payload memory block */ gst_buffer_append_memory (outbuf, gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY, (guint8 *) data + pos, limitedSize, 0, limitedSize, NULL, NULL)); /* add the buffer to the buffer list */ gst_buffer_list_add (list, outbuf); gst_rtp_buffer_unmap (&rtp); size -= limitedSize; pos += limitedSize; ii++; start = 0; } ret = gst_rtp_base_payload_push_list (basepayload, list); } 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, *nal_data; GstClockTime dts, pts; GArray *nal_queue; guint pushed = 0; gboolean avc; 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) { gst_buffer_map (buffer, &map, GST_MAP_READ); data = map.data; size = map.size; pts = GST_BUFFER_PTS (buffer); dts = GST_BUFFER_DTS (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); gst_adapter_push (rtph264pay->adapter, buffer); size = gst_adapter_available (rtph264pay->adapter); data = gst_adapter_map (rtph264pay->adapter, size); GST_DEBUG_OBJECT (basepayload, "got %" G_GSIZE_FORMAT " bytes (%" G_GSIZE_FORMAT ")", size, gst_buffer_get_size (buffer)); if (!GST_CLOCK_TIME_IS_VALID (dts)) dts = GST_BUFFER_DTS (buffer); if (!GST_CLOCK_TIME_IS_VALID (pts)) pts = GST_BUFFER_PTS (buffer); } 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; 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; 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; } ret = gst_rtp_h264_pay_payload_nal (basepayload, data, nal_len, dts, pts, end_of_au); if (ret != GST_FLOW_OK) break; data += 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_data = data; nal_queue = rtph264pay->queue; /* 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); if (next == size) { /* Didn't find the start of next NAL, 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 */ data = nal_data; pushed = 0; 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 */ data += 3; /* Trim the end unless we're the last NAL in the buffer. * 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; if (i + 1 != nal_queue->len) 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 && i == nal_queue->len - 1) end_of_au = TRUE; /* put the data in one or more RTP packets */ ret = gst_rtp_h264_pay_payload_nal (basepayload, data, size, dts, pts, end_of_au); if (ret != GST_FLOW_OK) { break; } /* move to next NAL packet */ data += nal_len; size -= nal_len; pushed += nal_len + 3; } 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); gst_adapter_flush (rtph264pay->adapter, pushed); } 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; 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); 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_uint (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_uint (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); }