/* 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 #include "gstrtpamrdepay.h" /* 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. */ /* elementfactory information */ static const GstElementDetails gst_rtp_amrdepay_details = GST_ELEMENT_DETAILS ("RTP packet depayloader", "Codec/Depayloader/Network", "Extracts AMR audio from RTP packets (RFC 3267)", "Wim Taymans "); /* RtpAMRDepay signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { ARG_0 }; /* input is an RTP packet * * params see RFC 3267, section 8.1 */ static GstStaticPadTemplate gst_rtp_amr_depay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "clock-rate = (int) 8000, " "encoding-name = (string) \"AMR\", " "encoding-params = (string) \"1\", " /* NOTE that all values must be strings in orde to be able to do SDP <-> * GstCaps mapping. */ "octet-align = (string) \"1\", " "crc = (string) { \"0\", \"1\" }, " "robust-sorting = (string) \"0\", " "interleaving = (string) \"0\"" /* following options are not needed for a decoder * "mode-set = (int) [ 0, 7 ], " "mode-change-period = (int) [ 1, MAX ], " "mode-change-neighbor = (boolean) { TRUE, FALSE }, " "maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ]" */ ) ); static GstStaticPadTemplate gst_rtp_amr_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000") ); static gboolean gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf); GST_BOILERPLATE (GstRtpAMRDepay, gst_rtp_amr_depay, GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD); static void gst_rtp_amr_depay_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_depay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_amr_depay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_amrdepay_details); } static void gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseRTPDepayloadClass *gstbasertpdepayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gstbasertpdepayload_class->process = gst_rtp_amr_depay_process; gstbasertpdepayload_class->set_caps = gst_rtp_amr_depay_setcaps; } static void gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay, GstRtpAMRDepayClass * klass) { GstBaseRTPDepayload *depayload; depayload = GST_BASE_RTP_DEPAYLOAD (rtpamrdepay); depayload->clock_rate = 8000; gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload)); } static gboolean gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) { GstStructure *structure; GstCaps *srccaps; GstRtpAMRDepay *rtpamrdepay; const gchar *params; const gchar *str; gint clock_rate; rtpamrdepay = GST_RTP_AMR_DEPAY (depayload); structure = gst_caps_get_structure (caps, 0); if (!(str = gst_structure_get_string (structure, "octet-align"))) rtpamrdepay->octet_align = FALSE; else rtpamrdepay->octet_align = (atoi (str) == 1); if (!(str = gst_structure_get_string (structure, "crc"))) rtpamrdepay->crc = FALSE; else rtpamrdepay->crc = (atoi (str) == 1); if (rtpamrdepay->crc) { /* crc mode implies octet aligned mode */ rtpamrdepay->octet_align = TRUE; } if (!(str = gst_structure_get_string (structure, "robust-sorting"))) rtpamrdepay->robust_sorting = FALSE; else rtpamrdepay->robust_sorting = (atoi (str) == 1); if (rtpamrdepay->robust_sorting) { /* robust_sorting mode implies octet aligned mode */ rtpamrdepay->octet_align = TRUE; } if (!(str = gst_structure_get_string (structure, "interleaving"))) rtpamrdepay->interleaving = FALSE; else rtpamrdepay->interleaving = (atoi (str) == 1); if (rtpamrdepay->interleaving) { /* interleaving mode implies octet aligned mode */ rtpamrdepay->octet_align = TRUE; } if (!(params = gst_structure_get_string (structure, "encoding-params"))) rtpamrdepay->channels = 1; else { rtpamrdepay->channels = atoi (params); } if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) clock_rate = 8000; /* we require 1 channel, 8000 Hz, octet aligned, no CRC, * no robust sorting, no interleaving for now */ if (rtpamrdepay->channels != 1) return FALSE; if (clock_rate != 8000) return FALSE; if (rtpamrdepay->octet_align != TRUE) return FALSE; if (rtpamrdepay->robust_sorting != FALSE) return FALSE; if (rtpamrdepay->interleaving != FALSE) return FALSE; srccaps = gst_caps_new_simple ("audio/AMR", "channels", G_TYPE_INT, rtpamrdepay->channels, "rate", G_TYPE_INT, clock_rate, NULL); gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps); gst_caps_unref (srccaps); rtpamrdepay->negotiated = TRUE; return TRUE; } /* -1 is invalid */ static gint frame_size[16] = { 12, 13, 15, 17, 19, 20, 26, 31, 5, -1, -1, -1, -1, -1, -1, 0 }; static GstBuffer * gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) { GstRtpAMRDepay *rtpamrdepay; GstBuffer *outbuf = NULL; rtpamrdepay = GST_RTP_AMR_DEPAY (depayload); if (!rtpamrdepay->negotiated) goto not_negotiated; if (!gst_rtp_buffer_validate (buf)) { GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, (NULL), ("AMR RTP packet did not validate")); goto bad_packet; } /* when we get here, 1 channel, 8000 Hz, octet aligned, no CRC, * no robust sorting, no interleaving data is to be depayloaded */ { gint payload_len; guint8 *payload, *p, *dp; guint32 timestamp; guint8 CMR; gint i, num_packets, num_nonempty_packets; gint amr_len; gint ILL, ILP; payload_len = gst_rtp_buffer_get_payload_len (buf); /* need at least 2 bytes for the header */ if (payload_len < 2) { GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, (NULL), ("AMR RTP payload too small (%d)", payload_len)); goto bad_packet; } payload = gst_rtp_buffer_get_payload (buf); /* depay CMR. The CMR is used by the sender to request * a new encoding mode. * * 0 1 2 3 4 5 6 7 * +-+-+-+-+-+-+-+-+ * | CMR |R|R|R|R| * +-+-+-+-+-+-+-+-+ */ CMR = (payload[0] & 0xf0) >> 4; /* strip CMR header now, pack FT and the data for the decoder */ payload_len -= 1; payload += 1; GST_DEBUG_OBJECT (rtpamrdepay, "payload len %d", payload_len); if (rtpamrdepay->interleaving) { ILL = (payload[0] & 0xf0) >> 4; ILP = (payload[0] & 0x0f); payload_len -= 1; payload += 1; if (ILP > ILL) { GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, (NULL), ("AMR RTP wrong interleaving")); goto bad_packet; } } /* * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 * +-+-+-+-+-+-+-+-+.. * |F| FT |Q|P|P| more FT.. * +-+-+-+-+-+-+-+-+.. */ /* count number of packets by counting the FTs. Also * count number of amr data bytes and number of non-empty * packets (this is also the number of CRCs if present). */ amr_len = 0; num_nonempty_packets = 0; num_packets = 0; for (i = 0; i < payload_len; i++) { gint fr_size; guint8 FT; FT = (payload[i] & 0x78) >> 3; fr_size = frame_size[FT]; GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size); if (fr_size == -1) { GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, (NULL), ("AMR RTP frame size == -1")); goto bad_packet; } if (fr_size > 0) { amr_len += fr_size; num_nonempty_packets++; } num_packets++; if ((payload[i] & 0x80) == 0) break; } if (rtpamrdepay->crc) { /* data len + CRC len + header bytes should be smaller than payload_len */ if (num_packets + num_nonempty_packets + amr_len > payload_len) { GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, (NULL), ("AMR RTP wrong length 1")); goto bad_packet; } } else { /* data len + header bytes should be smaller than payload_len */ if (num_packets + amr_len > payload_len) { GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE, (NULL), ("AMR RTP wrong length 2")); goto bad_packet; } } timestamp = gst_rtp_buffer_get_timestamp (buf); outbuf = gst_buffer_new_and_alloc (payload_len); GST_BUFFER_TIMESTAMP (outbuf) = gst_util_uint64_scale_int (timestamp, GST_SECOND, depayload->clock_rate); /* point to destination */ p = GST_BUFFER_DATA (outbuf); /* point to first data packet */ dp = payload + num_packets; if (rtpamrdepay->crc) { /* skip CRC if present */ dp += num_nonempty_packets; } for (i = 0; i < num_packets; i++) { gint fr_size; /* copy FT, clear F bit */ *p++ = payload[i] & 0x7f; fr_size = frame_size[(payload[i] & 0x78) >> 3]; if (fr_size > 0) { /* copy data packet, FIXME, calc CRC here. */ memcpy (p, dp, fr_size); p += fr_size; dp += fr_size; } } gst_buffer_set_caps (outbuf, GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload))); GST_DEBUG ("gst_rtp_amr_depay_chain: pushing buffer of size %d", GST_BUFFER_SIZE (outbuf)); } return outbuf; /* ERRORS */ not_negotiated: { GST_ELEMENT_ERROR (rtpamrdepay, STREAM, NOT_IMPLEMENTED, (NULL), ("not negotiated")); return NULL; } bad_packet: { /* no fatal error */ return NULL; } } gboolean gst_rtp_amr_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpamrdepay", GST_RANK_MARGINAL, GST_TYPE_RTP_AMR_DEPAY); }