/* GStreamer * Copyright (C) <2010> 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 "gstrtpg722pay.h" #include "gstrtpchannels.h" GST_DEBUG_CATEGORY_STATIC (rtpg722pay_debug); #define GST_CAT_DEFAULT (rtpg722pay_debug) static GstStaticPadTemplate gst_rtp_g722_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/G722, " "rate = (int) 16000, " "channels = (int) 1") ); static GstStaticPadTemplate gst_rtp_g722_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "encoding-name = (string) \"G722\", " "payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", " "clock-rate = (int) 8000") ); static gboolean gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps); static GstCaps *gst_rtp_g722_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad, GstCaps * filter); #define gst_rtp_g722_pay_parent_class parent_class G_DEFINE_TYPE (GstRtpG722Pay, gst_rtp_g722_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD); static void gst_rtp_g722_pay_class_init (GstRtpG722PayClass * klass) { GstElementClass *gstelement_class; GstRTPBasePayloadClass *gstrtpbasepayload_class; GST_DEBUG_CATEGORY_INIT (rtpg722pay_debug, "rtpg722pay", 0, "G722 RTP Payloader"); gstelement_class = (GstElementClass *) klass; gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_g722_pay_src_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_g722_pay_sink_template)); gst_element_class_set_details_simple (gstelement_class, "RTP audio payloader", "Codec/Payloader/Network/RTP", "Payload-encode Raw audio into RTP packets (RFC 3551)", "Wim Taymans "); gstrtpbasepayload_class->set_caps = gst_rtp_g722_pay_setcaps; gstrtpbasepayload_class->get_caps = gst_rtp_g722_pay_getcaps; } static void gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay) { GstRTPBaseAudioPayload *rtpbaseaudiopayload; rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg722pay); /* tell rtpbaseaudiopayload that this is a sample based codec */ gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); } static gboolean gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps) { GstRtpG722Pay *rtpg722pay; GstStructure *structure; gint rate, channels, clock_rate; gboolean res; gchar *params; #if 0 GstAudioChannelPosition *pos; const GstRTPChannelOrder *order; #endif GstRTPBaseAudioPayload *rtpbaseaudiopayload; rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload); rtpg722pay = GST_RTP_G722_PAY (basepayload); structure = gst_caps_get_structure (caps, 0); /* first parse input caps */ if (!gst_structure_get_int (structure, "rate", &rate)) goto no_rate; if (!gst_structure_get_int (structure, "channels", &channels)) goto no_channels; /* FIXME: Do something with the channel positions */ #if 0 /* get the channel order */ pos = gst_audio_get_channel_positions (structure); if (pos) order = gst_rtp_channels_get_by_pos (channels, pos); else order = NULL; #endif /* Clock rate is always 8000 Hz for G722 according to * RFC 3551 although the sampling rate is 16000 Hz */ clock_rate = 8000; gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "G722", clock_rate); params = g_strdup_printf ("%d", channels); #if 0 if (!order && channels > 2) { GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE, (NULL), ("Unknown channel order for %d channels", channels)); } if (order && order->name) { res = gst_rtp_base_payload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, channels, "channel-order", G_TYPE_STRING, order->name, NULL); } else { #endif res = gst_rtp_base_payload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, channels, NULL); #if 0 } #endif g_free (params); #if 0 g_free (pos); #endif rtpg722pay->rate = rate; rtpg722pay->channels = channels; /* bits-per-sample is 4 * channels for G722, but as the RTP clock runs at * half speed (8 instead of 16 khz), pretend it's 8 bits per sample * channels. */ gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload, 8 * rtpg722pay->channels); return res; /* ERRORS */ no_rate: { GST_DEBUG_OBJECT (rtpg722pay, "no rate given"); return FALSE; } no_channels: { GST_DEBUG_OBJECT (rtpg722pay, "no channels given"); return FALSE; } } static GstCaps * gst_rtp_g722_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad, GstCaps * filter) { GstCaps *otherpadcaps; GstCaps *caps; otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad); caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad)); if (otherpadcaps) { if (!gst_caps_is_empty (otherpadcaps)) { gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL); gst_caps_set_simple (caps, "rate", G_TYPE_INT, 16000, NULL); } gst_caps_unref (otherpadcaps); } return caps; } gboolean gst_rtp_g722_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg722pay", GST_RANK_SECONDARY, GST_TYPE_RTP_G722_PAY); }