/* GStreamer * Copyright (C) <2006> Wim Taymans <wim.taymans@gmail.com> * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include <string.h> #include <gst/rtp/gstrtpbuffer.h> #include "gstrtpmp4gpay.h" GST_DEBUG_CATEGORY_STATIC (rtpmp4gpay_debug); #define GST_CAT_DEFAULT (rtpmp4gpay_debug) /* elementfactory information */ static const GstElementDetails gst_rtp_mp4gpay_details = GST_ELEMENT_DETAILS ("RTP MPEG4 ES payloader", "Codec/Payloader/Network", "Payload MPEG4 elementary streams as RTP packets (RFC 3640)", "Wim Taymans <wim.taymans@gmail.com>"); static GstStaticPadTemplate gst_rtp_mp4g_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("video/mpeg," "mpegversion=(int) 4," "systemstream=(boolean)false;" "audio/mpeg," "mpegversion=(int) 4") ); static GstStaticPadTemplate gst_rtp_mp4g_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) { \"video\", \"audio\", \"application\" }, " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MPEG4-GENERIC\", " /* required string params */ "streamtype = (string) { \"4\", \"5\" }, " /* 4 = video, 5 = audio */ /* "profile-level-id = (string) [1,MAX], " */ /* "config = (string) [1,MAX]" */ "mode = (string) { \"generic\", \"CELP-cbr\", \"CELP-vbr\", \"AAC-lbr\", \"AAC-hbr\" } " /* Optional general parameters */ /* "objecttype = (string) [1,MAX], " */ /* "constantsize = (string) [1,MAX], " *//* constant size of each AU */ /* "constantduration = (string) [1,MAX], " *//* constant duration of each AU */ /* "maxdisplacement = (string) [1,MAX], " */ /* "de-interleavebuffersize = (string) [1,MAX], " */ /* Optional configuration parameters */ /* "sizelength = (string) [1, 16], " *//* max 16 bits, should be enough... */ /* "indexlength = (string) [1, 8], " */ /* "indexdeltalength = (string) [1, 8], " */ /* "ctsdeltalength = (string) [1, 64], " */ /* "dtsdeltalength = (string) [1, 64], " */ /* "randomaccessindication = (string) {0, 1}, " */ /* "streamstateindication = (string) [0, 64], " */ /* "auxiliarydatasizelength = (string) [0, 64]" */ ) ); static void gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass); static void gst_rtp_mp4g_pay_base_init (GstRtpMP4GPayClass * klass); static void gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay); static void gst_rtp_mp4g_pay_finalize (GObject * object); static gboolean gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); static GstBaseRTPPayloadClass *parent_class = NULL; static GType gst_rtp_mp4g_pay_get_type (void) { static GType rtpmp4gpay_type = 0; if (!rtpmp4gpay_type) { static const GTypeInfo rtpmp4gpay_info = { sizeof (GstRtpMP4GPayClass), (GBaseInitFunc) gst_rtp_mp4g_pay_base_init, NULL, (GClassInitFunc) gst_rtp_mp4g_pay_class_init, NULL, NULL, sizeof (GstRtpMP4GPay), 0, (GInstanceInitFunc) gst_rtp_mp4g_pay_init, }; rtpmp4gpay_type = g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMP4GPay", &rtpmp4gpay_info, 0); } return rtpmp4gpay_type; } static void gst_rtp_mp4g_pay_base_init (GstRtpMP4GPayClass * klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mp4g_pay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mp4g_pay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_mp4gpay_details); } static void gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass) { GObjectClass *gobject_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gobject_class = (GObjectClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gobject_class->finalize = gst_rtp_mp4g_pay_finalize; gstbasertppayload_class->set_caps = gst_rtp_mp4g_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer; GST_DEBUG_CATEGORY_INIT (rtpmp4gpay_debug, "rtpmp4gpay", 0, "MP4-generic RTP Payloader"); } static void gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay) { rtpmp4gpay->adapter = gst_adapter_new (); rtpmp4gpay->rate = 90000; rtpmp4gpay->profile = g_strdup ("1"); rtpmp4gpay->mode = ""; } static void gst_rtp_mp4g_pay_finalize (GObject * object) { GstRtpMP4GPay *rtpmp4gpay; rtpmp4gpay = GST_RTP_MP4G_PAY (object); g_object_unref (rtpmp4gpay->adapter); rtpmp4gpay->adapter = NULL; g_free (rtpmp4gpay->params); rtpmp4gpay->params = NULL; if (rtpmp4gpay->config) gst_buffer_unref (rtpmp4gpay->config); rtpmp4gpay->config = NULL; g_free (rtpmp4gpay->profile); rtpmp4gpay->profile = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } static unsigned sampling_table[16] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, 0 }; static gboolean gst_rtp_mp4g_pay_parse_audio_config (GstRtpMP4GPay * rtpmp4gpay, GstBuffer * buffer) { guint8 *data; guint size; guint8 objectType; guint8 samplingIdx; guint8 channelCfg; data = GST_BUFFER_DATA (buffer); size = GST_BUFFER_SIZE (buffer); if (size < 2) goto too_short; /* any object type is fine, we need to copy it to the profile-level-id field. */ objectType = (data[0] & 0xf8) >> 3; if (objectType == 0) goto invalid_object; samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7); /* only fixed values for now */ if (samplingIdx > 12 && samplingIdx != 15) goto wrong_freq; channelCfg = ((data[1] & 0x78) >> 3); if (channelCfg > 7) goto wrong_channels; /* rtp rate depends on sampling rate of the audio */ if (samplingIdx == 15) { if (size < 5) goto too_short; /* index of 15 means we get the rate in the next 24 bits */ rtpmp4gpay->rate = ((data[1] & 0x7f) << 17) | ((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7); } else { /* else use the rate from the table */ rtpmp4gpay->rate = sampling_table[samplingIdx]; } /* extra rtp params contain the number of channels */ g_free (rtpmp4gpay->params); rtpmp4gpay->params = g_strdup_printf ("%d", channelCfg); /* audio stream type */ rtpmp4gpay->streamtype = "5"; /* mode only high bitrate for now */ rtpmp4gpay->mode = "AAC-hbr"; /* profile */ g_free (rtpmp4gpay->profile); rtpmp4gpay->profile = g_strdup_printf ("%d", objectType); GST_DEBUG_OBJECT (rtpmp4gpay, "objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType, samplingIdx, rtpmp4gpay->rate, channelCfg); return TRUE; /* ERROR */ too_short: { GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT, (NULL), ("config string too short, expected 2 bytes, got %d", size)); return FALSE; } invalid_object: { GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT, (NULL), ("invalid object type 0")); return FALSE; } wrong_freq: { GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED, (NULL), ("unsupported frequency index %d", samplingIdx)); return FALSE; } wrong_channels: { GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED, (NULL), ("unsupported number of channels %d, must < 8", channelCfg)); return FALSE; } } #define VOS_STARTCODE 0x000001B0 static gboolean gst_rtp_mp4g_pay_parse_video_config (GstRtpMP4GPay * rtpmp4gpay, GstBuffer * buffer) { guint8 *data; guint size; guint32 code; data = GST_BUFFER_DATA (buffer); size = GST_BUFFER_SIZE (buffer); if (size < 5) goto too_short; code = GST_READ_UINT32_BE (data); g_free (rtpmp4gpay->profile); if (code == VOS_STARTCODE) { /* get profile */ rtpmp4gpay->profile = g_strdup_printf ("%d", (gint) data[4]); } else { GST_ELEMENT_WARNING (rtpmp4gpay, STREAM, FORMAT, (NULL), ("profile not found in config string, assuming \'1\'")); rtpmp4gpay->profile = g_strdup ("1"); } /* fixed rate */ rtpmp4gpay->rate = 90000; /* video stream type */ rtpmp4gpay->streamtype = "4"; /* no params for video */ rtpmp4gpay->params = NULL; /* mode */ rtpmp4gpay->mode = "generic"; GST_LOG_OBJECT (rtpmp4gpay, "profile %s", rtpmp4gpay->profile); return TRUE; /* ERROR */ too_short: { GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT, (NULL), ("config string too short")); return FALSE; } } static gboolean gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay) { gchar *config; GValue v = { 0 }; gboolean res; #define MP4GCAPS \ "streamtype", G_TYPE_STRING, rtpmp4gpay->streamtype, \ "profile-level-id", G_TYPE_STRING, rtpmp4gpay->profile, \ "mode", G_TYPE_STRING, rtpmp4gpay->mode, \ "config", G_TYPE_STRING, config, \ "sizelength", G_TYPE_STRING, "13", \ "indexlength", G_TYPE_STRING, "3", \ "indexdeltalength", G_TYPE_STRING, "3", \ NULL g_value_init (&v, GST_TYPE_BUFFER); gst_value_set_buffer (&v, rtpmp4gpay->config); config = gst_value_serialize (&v); /* hmm, silly */ if (rtpmp4gpay->params) { res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), "encoding-params", G_TYPE_STRING, rtpmp4gpay->params, MP4GCAPS); } else { res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), MP4GCAPS); } g_value_unset (&v); g_free (config); #undef MP4GCAPS return res; } static gboolean gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) { GstRtpMP4GPay *rtpmp4gpay; GstStructure *structure; const GValue *codec_data; gchar *media_type = NULL; gboolean res; rtpmp4gpay = GST_RTP_MP4G_PAY (payload); structure = gst_caps_get_structure (caps, 0); codec_data = gst_structure_get_value (structure, "codec_data"); if (codec_data) { GST_LOG_OBJECT (rtpmp4gpay, "got codec_data"); if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) { GstBuffer *buffer; const gchar *name; buffer = gst_value_get_buffer (codec_data); GST_LOG_OBJECT (rtpmp4gpay, "configuring codec_data"); name = gst_structure_get_name (structure); /* parse buffer */ if (!strcmp (name, "audio/mpeg")) { res = gst_rtp_mp4g_pay_parse_audio_config (rtpmp4gpay, buffer); media_type = "audio"; } else if (!strcmp (name, "video/mpeg")) { res = gst_rtp_mp4g_pay_parse_video_config (rtpmp4gpay, buffer); media_type = "video"; } else { res = FALSE; } if (!res) goto config_failed; /* now we can configure the buffer */ if (rtpmp4gpay->config) gst_buffer_unref (rtpmp4gpay->config); rtpmp4gpay->config = gst_buffer_copy (buffer); } } if (media_type == NULL) goto config_failed; gst_basertppayload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC", rtpmp4gpay->rate); res = gst_rtp_mp4g_pay_new_caps (rtpmp4gpay); return res; /* ERRORS */ config_failed: { GST_DEBUG_OBJECT (rtpmp4gpay, "failed to parse config"); return FALSE; } } static GstFlowReturn gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay) { guint avail, total; GstBuffer *outbuf; GstFlowReturn ret; gboolean fragmented; guint mtu; fragmented = FALSE; /* the data available in the adapter is either smaller * than the MTU or bigger. In the case it is smaller, the complete * adapter contents can be put in one packet. In the case the * adapter has more than one MTU, we need to fragment the MPEG data * over multiple packets. */ total = avail = gst_adapter_available (rtpmp4gpay->adapter); ret = GST_FLOW_OK; mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpmp4gpay); while (avail > 0) { guint towrite; guint8 *payload; guint payload_len; guint packet_len; /* this will be the total lenght of the packet */ packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0); /* fill one MTU or all available bytes, we need 4 spare bytes for * the AU header. */ towrite = MIN (packet_len, mtu - 4); /* this is the payload length */ payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0); GST_DEBUG_OBJECT (rtpmp4gpay, "avail %d, towrite %d, packet_len %d, payload_len %d", avail, towrite, packet_len, payload_len); /* create buffer to hold the payload, also make room for the 4 header bytes. */ outbuf = gst_rtp_buffer_new_allocate (payload_len + 4, 0, 0); /* copy payload */ payload = gst_rtp_buffer_get_payload (outbuf); /* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+ * |AU-headers-length|AU-header|AU-header| |AU-header|padding| * | | (1) | (2) | | (n) | bits | * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+ */ /* AU-headers-length, we only have 1 AU-header */ payload[0] = 0x00; payload[1] = 0x10; /* we use 16 bits for the header */ /* +---------------------------------------+ * | AU-size | * +---------------------------------------+ * | AU-Index / AU-Index-delta | * +---------------------------------------+ * | CTS-flag | * +---------------------------------------+ * | CTS-delta | * +---------------------------------------+ * | DTS-flag | * +---------------------------------------+ * | DTS-delta | * +---------------------------------------+ * | RAP-flag | * +---------------------------------------+ * | Stream-state | * +---------------------------------------+ */ /* The AU-header, no CTS, DTS, RAP, Stream-state * * AU-size is always the total size of the AU, not the fragmented size */ payload[2] = (total & 0x1fe0) >> 5; payload[3] = (total & 0x1f) << 3; /* we use 13 bits for the size, 3 bits index */ /* copy stuff from adapter to payload */ gst_adapter_copy (rtpmp4gpay->adapter, &payload[4], 0, payload_len); gst_adapter_flush (rtpmp4gpay->adapter, payload_len); /* marker only if the packet is complete */ gst_rtp_buffer_set_marker (outbuf, avail <= payload_len); GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_timestamp; GST_BUFFER_DURATION (outbuf) = rtpmp4gpay->first_duration; ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf); avail -= payload_len; fragmented = TRUE; } return ret; } /* we expect buffers as exactly one complete AU */ static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstRtpMP4GPay *rtpmp4gpay; rtpmp4gpay = GST_RTP_MP4G_PAY (basepayload); rtpmp4gpay->first_timestamp = GST_BUFFER_TIMESTAMP (buffer); rtpmp4gpay->first_duration = GST_BUFFER_DURATION (buffer); /* we always encode and flush a full AU */ gst_adapter_push (rtpmp4gpay->adapter, buffer); return gst_rtp_mp4g_pay_flush (rtpmp4gpay); } gboolean gst_rtp_mp4g_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmp4gpay", GST_RANK_NONE, GST_TYPE_RTP_MP4G_PAY); }