/* GStreamer * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.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 <stdlib.h> #include <gst/audio/audio.h> #include <gst/audio/multichannel.h> #include "gstrtpL16depay.h" #include "gstrtpchannels.h" GST_DEBUG_CATEGORY_STATIC (rtpL16depay_debug); #define GST_CAT_DEFAULT (rtpL16depay_debug) /* elementfactory information */ static const GstElementDetails gst_rtp_L16_depay_details = GST_ELEMENT_DETAILS ("RTP audio depayloader", "Codec/Depayloader/Network", "Extracts raw audio from RTP packets", "Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>"); static GstStaticPadTemplate gst_rtp_L16_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, 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_depay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) [ 1, MAX ], " /* "channels = (int) [1, MAX]" */ /* "emphasis = (string) ANY" */ /* "channel-order = (string) ANY" */ "encoding-name = (string) \"L16\";" "application/x-rtp, " "media = (string) \"audio\", " "payload = (int) { " GST_RTP_PAYLOAD_L16_STEREO_STRING ", " GST_RTP_PAYLOAD_L16_MONO_STRING " }," "clock-rate = (int) [ 1, MAX ]" /* "channels = (int) [1, MAX]" */ /* "emphasis = (string) ANY" */ /* "channel-order = (string) ANY" */ ) ); GST_BOILERPLATE (GstRtpL16Depay, gst_rtp_L16_depay, GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD); static gboolean gst_rtp_L16_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_L16_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf); static void gst_rtp_L16_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_L16_depay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_L16_depay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_L16_depay_details); } static void gst_rtp_L16_depay_class_init (GstRtpL16DepayClass * klass) { GstBaseRTPDepayloadClass *gstbasertpdepayload_class; gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gstbasertpdepayload_class->set_caps = gst_rtp_L16_depay_setcaps; gstbasertpdepayload_class->process = gst_rtp_L16_depay_process; GST_DEBUG_CATEGORY_INIT (rtpL16depay_debug, "rtpL16depay", 0, "Raw Audio RTP Depayloader"); } static void gst_rtp_L16_depay_init (GstRtpL16Depay * rtpL16depay, GstRtpL16DepayClass * klass) { /* needed because of GST_BOILERPLATE */ } static gint gst_rtp_L16_depay_parse_int (GstStructure * structure, const gchar * field, gint def) { const gchar *str; gint res; if ((str = gst_structure_get_string (structure, field))) return atoi (str); if (gst_structure_get_int (structure, field, &res)) return res; return def; } static gboolean gst_rtp_L16_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) { GstStructure *structure; GstRtpL16Depay *rtpL16depay; gint clock_rate, payload; gint channels; GstCaps *srccaps; gboolean res; const gchar *channel_order; const GstRTPChannelOrder *order; rtpL16depay = GST_RTP_L16_DEPAY (depayload); structure = gst_caps_get_structure (caps, 0); payload = 96; gst_structure_get_int (structure, "payload", &payload); switch (payload) { case GST_RTP_PAYLOAD_L16_STEREO: channels = 2; clock_rate = 44100; break; case GST_RTP_PAYLOAD_L16_MONO: channels = 1; clock_rate = 44100; break; default: /* no fixed mapping, we need channels and clock-rate */ channels = 0; clock_rate = 0; break; } /* caps can overwrite defaults */ clock_rate = gst_rtp_L16_depay_parse_int (structure, "clock-rate", clock_rate); if (clock_rate == 0) goto no_clockrate; channels = gst_rtp_L16_depay_parse_int (structure, "channels", channels); if (channels == 0) goto no_channels; depayload->clock_rate = clock_rate; rtpL16depay->rate = clock_rate; rtpL16depay->channels = channels; srccaps = gst_caps_new_simple ("audio/x-raw-int", "endianness", G_TYPE_INT, G_BIG_ENDIAN, "signed", G_TYPE_BOOLEAN, TRUE, "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, "rate", G_TYPE_INT, clock_rate, "channels", G_TYPE_INT, channels, NULL); /* add channel positions */ channel_order = gst_structure_get_string (structure, "channel-order"); order = gst_rtp_channels_get_by_order (channels, channel_order); if (order) { gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), order->pos); } else { GstAudioChannelPosition *pos; GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE, (NULL), ("Unknown channel order '%s' for %d channels", GST_STR_NULL (channel_order), channels)); /* create default NONE layout */ pos = gst_rtp_channels_create_default (channels); gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos); g_free (pos); } res = gst_pad_set_caps (depayload->srcpad, srccaps); gst_caps_unref (srccaps); return res; /* ERRORS */ no_clockrate: { GST_ERROR_OBJECT (depayload, "no clock-rate specified"); return FALSE; } no_channels: { GST_ERROR_OBJECT (depayload, "no channels specified"); return FALSE; } } static GstBuffer * gst_rtp_L16_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) { GstRtpL16Depay *rtpL16depay; GstBuffer *outbuf; gint payload_len; gboolean marker; rtpL16depay = GST_RTP_L16_DEPAY (depayload); payload_len = gst_rtp_buffer_get_payload_len (buf); if (payload_len <= 0) goto empty_packet; GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len); outbuf = gst_rtp_buffer_get_payload_buffer (buf); marker = gst_rtp_buffer_get_marker (buf); if (marker) { /* mark talk spurt with DISCONT */ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); } return outbuf; /* ERRORS */ empty_packet: { GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE, ("Empty Payload."), (NULL)); return NULL; } } gboolean gst_rtp_L16_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpL16depay", GST_RANK_MARGINAL, GST_TYPE_RTP_L16_DEPAY); }