/* 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 */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include "gstrtpamrenc.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 GstElementDetails gst_rtp_amrenc_details = { "RTP packet parser", "Codec/Parser/Network", "Encode AMR audio into RTP packets (RFC 3267)", "Wim Taymans " }; static GstStaticPadTemplate gst_rtpamrenc_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_rtpamrenc_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, 255 ], " "clock_rate = (int) 8000, " "encoding_name = (string) \"AMR\", " "encoding_params = (string) \"1\", " "octet-align = (boolean) TRUE, " "crc = (boolean) FALSE, " "robust-sorting = (boolean) FALSE, " "interleaving = (boolean) FALSE, " "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 void gst_rtpamrenc_class_init (GstRtpAMREncClass * klass); static void gst_rtpamrenc_base_init (GstRtpAMREncClass * klass); static void gst_rtpamrenc_init (GstRtpAMREnc * rtpamrenc); static gboolean gst_rtpamrenc_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps); static GstFlowReturn gst_rtpamrenc_handle_buffer (GstBaseRTPPayload * pad, GstBuffer * buffer); static GstBaseRTPPayloadClass *parent_class = NULL; static GType gst_rtpamrenc_get_type (void) { static GType rtpamrenc_type = 0; if (!rtpamrenc_type) { static const GTypeInfo rtpamrenc_info = { sizeof (GstRtpAMREncClass), (GBaseInitFunc) gst_rtpamrenc_base_init, NULL, (GClassInitFunc) gst_rtpamrenc_class_init, NULL, NULL, sizeof (GstRtpAMREnc), 0, (GInstanceInitFunc) gst_rtpamrenc_init, }; rtpamrenc_type = g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpAMREnc", &rtpamrenc_info, 0); } return rtpamrenc_type; } static void gst_rtpamrenc_base_init (GstRtpAMREncClass * klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtpamrenc_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtpamrenc_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_amrenc_details); } static void gst_rtpamrenc_class_init (GstRtpAMREncClass * 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_ref (GST_TYPE_BASE_RTP_PAYLOAD); gstbasertppayload_class->set_caps = gst_rtpamrenc_setcaps; gstbasertppayload_class->handle_buffer = gst_rtpamrenc_handle_buffer; } static void gst_rtpamrenc_init (GstRtpAMREnc * rtpamrenc) { } static gboolean gst_rtpamrenc_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps) { GstRtpAMREnc *rtpamrenc; rtpamrenc = GST_RTP_AMR_ENC (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_BOOLEAN, TRUE, "crc", G_TYPE_BOOLEAN, FALSE, "robust-sorting", G_TYPE_BOOLEAN, FALSE, "interleaving", G_TYPE_BOOLEAN, FALSE, NULL); return TRUE; } static GstFlowReturn gst_rtpamrenc_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstRtpAMREnc *rtpamrenc; GstFlowReturn ret; guint size, payload_len; GstBuffer *outbuf; guint8 *payload, *data; GstClockTime timestamp; rtpamrenc = GST_RTP_AMR_ENC (basepayload); size = GST_BUFFER_SIZE (buffer); timestamp = GST_BUFFER_TIMESTAMP (buffer); /* FIXME, only one AMR frame per RTP packet for now, * octet aligned, no interleaving, single channel, no CRC, * no robust-sorting. */ /* we need one extra byte for the CMR, the ToC is in the input * data */ payload_len = size + 1; outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0); /* FIXME, assert for now */ g_assert (GST_BUFFER_SIZE (outbuf) < GST_BASE_RTP_PAYLOAD_MTU (rtpamrenc)); /* copy timestamp */ GST_BUFFER_TIMESTAMP (outbuf) = timestamp; /* get payload */ payload = gst_rtpbuffer_get_payload (outbuf); /* 0 1 2 3 4 5 6 7 * +-+-+-+-+-+-+-+-+ * | CMR |R|R|R|R| * +-+-+-+-+-+-+-+-+ */ payload[0] = 0xF0; /* CMR, no specific mode requested */ data = GST_BUFFER_DATA (buffer); /* copy data in payload */ memcpy (&payload[1], data, size); /* 0 1 2 3 4 5 6 7 * +-+-+-+-+-+-+-+-+ * |F| FT |Q|P|P| * +-+-+-+-+-+-+-+-+ */ /* clear F flag */ payload[1] = payload[1] & 0x7f; gst_buffer_unref (buffer); ret = gst_basertppayload_push (basepayload, outbuf); return ret; } gboolean gst_rtpamrenc_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpamrenc", GST_RANK_NONE, GST_TYPE_RTP_AMR_ENC); }