/* GStreamer * Copyright (C) <2007> 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 #include "gstrtpL16pay.h" #include "gstrtpchannels.h" GST_DEBUG_CATEGORY_STATIC (rtpL16pay_debug); #define GST_CAT_DEFAULT (rtpL16pay_debug) static GstStaticPadTemplate gst_rtp_L16_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "endianness = (int) BIG_ENDIAN, " "signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") ); static GstStaticPadTemplate gst_rtp_L16_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) [ 1, MAX ], " "encoding-name = (string) \"L16\", " "channels = (int) [ 1, MAX ];" "application/x-rtp, " "media = (string) \"audio\", " "encoding-name = (string) \"L16\", " "payload = (int) " GST_RTP_PAYLOAD_L16_STEREO_STRING ", " "clock-rate = (int) 44100;" "application/x-rtp, " "media = (string) \"audio\", " "encoding-name = (string) \"L16\", " "payload = (int) " GST_RTP_PAYLOAD_L16_MONO_STRING ", " "clock-rate = (int) 44100") ); static gboolean gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps); static GstCaps *gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad, GstCaps * filter); #define gst_rtp_L16_pay_parent_class parent_class G_DEFINE_TYPE (GstRtpL16Pay, gst_rtp_L16_pay, GST_TYPE_BASE_RTP_AUDIO_PAYLOAD); static void gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass) { GstElementClass *gstelement_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gstelement_class = (GstElementClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; gstbasertppayload_class->set_caps = gst_rtp_L16_pay_setcaps; gstbasertppayload_class->get_caps = gst_rtp_L16_pay_getcaps; gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_L16_pay_src_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_L16_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 "); GST_DEBUG_CATEGORY_INIT (rtpL16pay_debug, "rtpL16pay", 0, "L16 RTP Payloader"); } static void gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay) { GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpL16pay); /* tell basertpaudiopayload that this is a sample based codec */ gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload); } static gboolean gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps) { GstRtpL16Pay *rtpL16pay; GstStructure *structure; gint channels, rate; gboolean res; gchar *params; GstAudioChannelPosition *pos; const GstRTPChannelOrder *order; GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); rtpL16pay = GST_RTP_L16_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; /* 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; gst_basertppayload_set_options (basepayload, "audio", TRUE, "L16", rate); params = g_strdup_printf ("%d", channels); if (!order && channels > 2) { GST_ELEMENT_WARNING (rtpL16pay, STREAM, DECODE, (NULL), ("Unknown channel order for %d channels", channels)); } if (order && order->name) { res = gst_basertppayload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, channels, "channel-order", G_TYPE_STRING, order->name, NULL); } else { res = gst_basertppayload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, channels, NULL); } g_free (params); g_free (pos); rtpL16pay->rate = rate; rtpL16pay->channels = channels; /* octet-per-sample is 2 * channels for L16 */ gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 2 * rtpL16pay->channels); return res; /* ERRORS */ no_rate: { GST_DEBUG_OBJECT (rtpL16pay, "no rate given"); return FALSE; } no_channels: { GST_DEBUG_OBJECT (rtpL16pay, "no channels given"); return FALSE; } } static GstCaps * gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * 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)) { GstStructure *structure; gint channels; gint pt; gint rate; structure = gst_caps_get_structure (otherpadcaps, 0); if (gst_structure_get_int (structure, "channels", &channels)) { gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL); } else if (gst_structure_get_int (structure, "payload", &pt)) { if (pt == 10) gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL); else if (pt == 11) gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL); } if (gst_structure_get_int (structure, "clock-rate", &rate)) { gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL); } else if (gst_structure_get_int (structure, "payload", &pt)) { if (pt == 10 || pt == 11) gst_caps_set_simple (caps, "rate", G_TYPE_INT, 44100, NULL); } } gst_caps_unref (otherpadcaps); } if (filter) { GstCaps *tcaps = caps; caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (tcaps); } return caps; } gboolean gst_rtp_L16_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpL16pay", GST_RANK_SECONDARY, GST_TYPE_RTP_L16_PAY); }