| /* GStreamer |
| * Copyright (C) <2005> Wim Taymans <wim@fluendo.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 "gstrtpamrpay.h" |
| |
| GST_DEBUG_CATEGORY (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. |
| */ |
| |
| /* elementfactory information */ |
| static const GstElementDetails gst_rtp_amrpay_details = |
| GST_ELEMENT_DETAILS ("RTP packet parser", |
| "Codec/Payloader/Network", |
| "Payload-encode AMR audio into RTP packets (RFC 3267)", |
| "Wim Taymans <wim@fluendo.com>"); |
| |
| 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") |
| ); |
| |
| 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) [ 96, 127 ], " |
| "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 ]") |
| ); |
| |
| static void gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * klass); |
| static void gst_rtp_amr_pay_base_init (GstRtpAMRPayClass * klass); |
| static void gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay); |
| |
| static gboolean gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, |
| GstCaps * caps); |
| static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * pad, |
| GstBuffer * buffer); |
| |
| static GstBaseRTPPayloadClass *parent_class = NULL; |
| |
| static GType |
| gst_rtp_amr_pay_get_type (void) |
| { |
| static GType rtpamrpay_type = 0; |
| |
| if (!rtpamrpay_type) { |
| static const GTypeInfo rtpamrpay_info = { |
| sizeof (GstRtpAMRPayClass), |
| (GBaseInitFunc) gst_rtp_amr_pay_base_init, |
| NULL, |
| (GClassInitFunc) gst_rtp_amr_pay_class_init, |
| NULL, |
| NULL, |
| sizeof (GstRtpAMRPay), |
| 0, |
| (GInstanceInitFunc) gst_rtp_amr_pay_init, |
| }; |
| |
| rtpamrpay_type = |
| g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpAMRPay", |
| &rtpamrpay_info, 0); |
| } |
| return rtpamrpay_type; |
| } |
| |
| static void |
| gst_rtp_amr_pay_base_init (GstRtpAMRPayClass * 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) |
| { |
| GObjectClass *gobject_class; |
| GstElementClass *gstelement_class; |
| GstBaseRTPPayloadClass *gstbasertppayload_class; |
| |
| gobject_class = (GObjectClass *) klass; |
| gstelement_class = (GstElementClass *) klass; |
| 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 RTP Payloader"); |
| |
| } |
| |
| static void |
| gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay) |
| { |
| } |
| |
| static gboolean |
| gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps) |
| { |
| GstRtpAMRPay *rtpamrpay; |
| |
| rtpamrpay = GST_RTP_AMR_PAY (basepayload); |
| |
| gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000); |
| 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; |
| } |
| |
| /* -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 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; |
| guint packet_len, mtu; |
| gint i, num_packets, num_nonempty_packets; |
| gint amr_len; |
| |
| 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); |
| |
| /* FIXME, only |
| * octet aligned, no interleaving, single channel, no CRC, |
| * no robust-sorting. */ |
| |
| GST_DEBUG_OBJECT (basepayload, "got %d bytes", size); |
| |
| /* 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, or fabricate one */ |
| if (timestamp != GST_CLOCK_TIME_NONE) |
| GST_BUFFER_TIMESTAMP (outbuf) = timestamp; |
| else { |
| /* AMR (nb) and AMR-WB both have 20 ms per frame */ |
| /* FIXME: when we do more than one AMR frame per packet, fix this */ |
| gint count = basepayload->seqnum - basepayload->seqnum_base; |
| |
| GST_BUFFER_TIMESTAMP (outbuf) = count * 20 * GST_MSECOND; |
| } |
| |
| /* 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); |
| } |