/* GStreamer * Copyright (C) <2005> Wim Taymans * * 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 "gstrtpamrpay.h" GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug); #define GST_CAT_DEFAULT (rtpamrpay_debug) /* references: * * RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File * Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive * Multi-Rate Wideband (AMR-WB) Audio Codecs. * * ETSI TS 126 201 V6.0.0 (2004-12) - Digital cellular telecommunications system (Phase 2+); * Universal Mobile Telecommunications System (UMTS); * AMR speech codec, wideband; * Frame structure * (3GPP TS 26.201 version 6.0.0 Release 6) */ /* elementfactory information */ static const GstElementDetails gst_rtp_amrpay_details = GST_ELEMENT_DETAILS ("RTP AMR payloader", "Codec/Payloader/Network", "Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267)", "Wim Taymans "); static GstStaticPadTemplate gst_rtp_amr_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/AMR, channels=(int)1, rate=(int)8000; " "audio/AMR-WB, channels=(int)1, rate=(int)16000") ); static GstStaticPadTemplate gst_rtp_amr_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 8000, " "encoding-name = (string) \"AMR\", " "encoding-params = (string) \"1\", " "octet-align = (string) \"1\", " "crc = (string) \"0\", " "robust-sorting = (string) \"0\", " "interleaving = (string) \"0\", " "mode-set = (int) [ 0, 7 ], " "mode-change-period = (int) [ 1, MAX ], " "mode-change-neighbor = (string) { \"0\", \"1\" }, " "maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ];" "application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 16000, " "encoding-name = (string) \"AMR-WB\", " "encoding-params = (string) \"1\", " "octet-align = (string) \"1\", " "crc = (string) \"0\", " "robust-sorting = (string) \"0\", " "interleaving = (string) \"0\", " "mode-set = (int) [ 0, 7 ], " "mode-change-period = (int) [ 1, MAX ], " "mode-change-neighbor = (string) { \"0\", \"1\" }, " "maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ]") ); static gboolean gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps); static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * pad, GstBuffer * buffer); GST_BOILERPLATE (GstRtpAMRPay, gst_rtp_amr_pay, GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD); static void gst_rtp_amr_pay_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_amr_pay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_amr_pay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_amrpay_details); } static void gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * klass) { GstBaseRTPPayloadClass *gstbasertppayload_class; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gstbasertppayload_class->set_caps = gst_rtp_amr_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_amr_pay_handle_buffer; GST_DEBUG_CATEGORY_INIT (rtpamrpay_debug, "rtpamrpay", 0, "AMR/AMR-WB RTP Payloader"); } static void gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay, GstRtpAMRPayClass * klass) { /* needed because of GST_BOILERPLATE */ } static gboolean gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps) { GstRtpAMRPay *rtpamrpay; const GstStructure *s; const gchar *str; rtpamrpay = GST_RTP_AMR_PAY (basepayload); /* figure out the mode Narrow or Wideband */ s = gst_caps_get_structure (caps, 0); if ((str = gst_structure_get_name (s))) { if (strcmp (str, "audio/AMR") == 0) rtpamrpay->mode = GST_RTP_AMR_P_MODE_NB; else if (strcmp (str, "audio/AMR-WB") == 0) rtpamrpay->mode = GST_RTP_AMR_P_MODE_WB; else goto wrong_type; } else goto wrong_type; if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB) gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000); else gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR-WB", 16000); gst_basertppayload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1", /* don't set the defaults * * "crc", G_TYPE_STRING, "0", * "robust-sorting", G_TYPE_STRING, "0", * "interleaving", G_TYPE_STRING, "0", */ NULL); return TRUE; /* ERRORS */ wrong_type: { GST_ERROR_OBJECT (rtpamrpay, "unsupported media type '%s'", GST_STR_NULL (str)); return FALSE; } } /* -1 is invalid */ static gint nb_frame_size[16] = { 12, 13, 15, 17, 19, 20, 26, 31, 5, -1, -1, -1, -1, -1, -1, 0 }; static gint wb_frame_size[16] = { 17, 23, 32, 36, 40, 46, 50, 58, 60, -1, -1, -1, -1, -1, -1, 0 }; static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstRtpAMRPay *rtpamrpay; GstFlowReturn ret; guint size, payload_len; GstBuffer *outbuf; guint8 *payload, *data, *payload_amr; GstClockTime timestamp, duration; guint packet_len, mtu; gint i, num_packets, num_nonempty_packets; gint amr_len; gint *frame_size; rtpamrpay = GST_RTP_AMR_PAY (basepayload); mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpamrpay); size = GST_BUFFER_SIZE (buffer); data = GST_BUFFER_DATA (buffer); timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); /* setup frame size pointer */ if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB) frame_size = nb_frame_size; else frame_size = wb_frame_size; GST_DEBUG_OBJECT (basepayload, "got %d bytes", size); /* FIXME, only * octet aligned, no interleaving, single channel, no CRC, * no robust-sorting. To fix this you need to implement the downstream * negotiation function. */ /* first count number of packets and total amr frame size */ amr_len = num_packets = num_nonempty_packets = 0; for (i = 0; i < size; i++) { guint8 FT; gint fr_size; FT = (data[i] & 0x78) >> 3; fr_size = frame_size[FT]; GST_DEBUG_OBJECT (basepayload, "frame size %d", fr_size); /* FIXME, we don't handle this yet.. */ if (fr_size <= 0) goto wrong_size; amr_len += fr_size; num_nonempty_packets++; num_packets++; i += fr_size; } if (amr_len > size) goto incomplete_frame; /* we need one extra byte for the CMR, the ToC is in the input * data */ payload_len = size + 1; /* get packet len to check against MTU */ packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0); if (packet_len > mtu) goto too_big; /* now alloc output buffer */ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); /* copy timestamp */ GST_BUFFER_TIMESTAMP (outbuf) = timestamp; /* FIXME: when we do more than one AMR frame per packet, fix this */ if (duration != GST_CLOCK_TIME_NONE) GST_BUFFER_DURATION (outbuf) = duration; else { GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND; } if (GST_BUFFER_IS_DISCONT (buffer)) { GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit"); GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); gst_rtp_buffer_set_marker (outbuf, TRUE); } /* get payload, this is now writable */ payload = gst_rtp_buffer_get_payload (outbuf); /* 0 1 2 3 4 5 6 7 * +-+-+-+-+-+-+-+-+ * | CMR |R|R|R|R| * +-+-+-+-+-+-+-+-+ */ payload[0] = 0xF0; /* CMR, no specific mode requested */ /* this is where we copy the AMR data, after num_packets FTs and the * CMR. */ payload_amr = payload + num_packets + 1; /* copy data in payload, first we copy all the FTs then all * the AMR data. The last FT has to have the F flag cleared. */ for (i = 1; i <= num_packets; i++) { guint8 FT; gint fr_size; /* 0 1 2 3 4 5 6 7 * +-+-+-+-+-+-+-+-+ * |F| FT |Q|P|P| more FT... * +-+-+-+-+-+-+-+-+ */ FT = (*data & 0x78) >> 3; fr_size = frame_size[FT]; if (i == num_packets) /* last packet, clear F flag */ payload[i] = *data & 0x7f; else /* set F flag */ payload[i] = *data | 0x80; memcpy (payload_amr, &data[1], fr_size); /* all sizes are > 0 since we checked for that above */ data += fr_size + 1; payload_amr += fr_size; } gst_buffer_unref (buffer); ret = gst_basertppayload_push (basepayload, outbuf); return ret; /* ERRORS */ wrong_size: { GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT, (NULL), ("received AMR frame with size <= 0")); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } incomplete_frame: { GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT, (NULL), ("received incomplete AMR frames")); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } too_big: { GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT, (NULL), ("received too many AMR frames for MTU")); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } } gboolean gst_rtp_amr_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpamrpay", GST_RANK_NONE, GST_TYPE_RTP_AMR_PAY); }