/* 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-rtpL24depay * @see_also: rtpL24pay * * Extract raw audio from RTP packets according to RFC 3190, section 4. * For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt * * <refsect2> * <title>Example pipeline</title> * |[ * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink * ]| This example pipeline will depayload an RTP raw audio stream. Refer to * the rtpL24pay example to create the RTP stream. * </refsect2> */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <string.h> #include <stdlib.h> #include <gst/audio/audio.h> #include "gstrtpL24depay.h" #include "gstrtpchannels.h" #include "gstrtputils.h" GST_DEBUG_CATEGORY_STATIC (rtpL24depay_debug); #define GST_CAT_DEFAULT (rtpL24depay_debug) static GstStaticPadTemplate gst_rtp_L24_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) S24BE, " "layout = (string) interleaved, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") ); static GstStaticPadTemplate gst_rtp_L24_depay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "clock-rate = (int) [ 1, MAX ], " "encoding-name = (string) \"L24\"") ); #define gst_rtp_L24_depay_parent_class parent_class G_DEFINE_TYPE (GstRtpL24Depay, gst_rtp_L24_depay, GST_TYPE_RTP_BASE_DEPAYLOAD); static gboolean gst_rtp_L24_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_L24_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp); static void gst_rtp_L24_depay_class_init (GstRtpL24DepayClass * klass) { GstElementClass *gstelement_class; GstRTPBaseDepayloadClass *gstrtpbasedepayload_class; gstelement_class = (GstElementClass *) klass; gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass; gstrtpbasedepayload_class->set_caps = gst_rtp_L24_depay_setcaps; gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_L24_depay_process; gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_L24_depay_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_L24_depay_sink_template); gst_element_class_set_static_metadata (gstelement_class, "RTP audio depayloader", "Codec/Depayloader/Network/RTP", "Extracts raw 24-bit audio from RTP packets", "Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>," "David Holroyd <dave@badgers-in-foil.co.uk>"); GST_DEBUG_CATEGORY_INIT (rtpL24depay_debug, "rtpL24depay", 0, "Raw Audio RTP Depayloader"); } static void gst_rtp_L24_depay_init (GstRtpL24Depay * rtpL24depay) { } static gint gst_rtp_L24_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_L24_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) { GstStructure *structure; GstRtpL24Depay *rtpL24depay; gint clock_rate, payload; gint channels; GstCaps *srccaps; gboolean res; const gchar *channel_order; const GstRTPChannelOrder *order; GstAudioInfo *info; rtpL24depay = GST_RTP_L24_DEPAY (depayload); structure = gst_caps_get_structure (caps, 0); payload = 96; gst_structure_get_int (structure, "payload", &payload); /* no fixed mapping, we need clock-rate */ channels = 0; clock_rate = 0; /* caps can overwrite defaults */ clock_rate = gst_rtp_L24_depay_parse_int (structure, "clock-rate", clock_rate); if (clock_rate == 0) goto no_clockrate; channels = gst_rtp_L24_depay_parse_int (structure, "encoding-params", channels); if (channels == 0) { channels = gst_rtp_L24_depay_parse_int (structure, "channels", channels); if (channels == 0) { /* channels defaults to 1 otherwise */ channels = 1; } } depayload->clock_rate = clock_rate; info = &rtpL24depay->info; gst_audio_info_init (info); info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S24BE); info->rate = clock_rate; info->channels = channels; info->bpf = (info->finfo->width / 8) * channels; /* add channel positions */ channel_order = gst_structure_get_string (structure, "channel-order"); order = gst_rtp_channels_get_by_order (channels, channel_order); rtpL24depay->order = order; if (order) { memcpy (info->position, order->pos, sizeof (GstAudioChannelPosition) * channels); gst_audio_channel_positions_to_valid_order (info->position, info->channels); } else { GST_ELEMENT_WARNING (rtpL24depay, STREAM, DECODE, (NULL), ("Unknown channel order '%s' for %d channels", GST_STR_NULL (channel_order), channels)); /* create default NONE layout */ gst_rtp_channels_create_default (channels, info->position); } srccaps = gst_audio_info_to_caps (info); 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; } } static GstBuffer * gst_rtp_L24_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp) { GstRtpL24Depay *rtpL24depay; GstBuffer *outbuf; gint payload_len; gboolean marker; rtpL24depay = GST_RTP_L24_DEPAY (depayload); payload_len = gst_rtp_buffer_get_payload_len (rtp); if (payload_len <= 0) goto empty_packet; GST_DEBUG_OBJECT (rtpL24depay, "got payload of %d bytes", payload_len); outbuf = gst_rtp_buffer_get_payload_buffer (rtp); marker = gst_rtp_buffer_get_marker (rtp); if (marker) { /* mark talk spurt with RESYNC */ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC); } outbuf = gst_buffer_make_writable (outbuf); if (outbuf) { gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpL24depay), outbuf, g_quark_from_static_string (GST_META_TAG_AUDIO_STR)); } if (rtpL24depay->order && !gst_audio_buffer_reorder_channels (outbuf, rtpL24depay->info.finfo->format, rtpL24depay->info.channels, rtpL24depay->info.position, rtpL24depay->order->pos)) { goto reorder_failed; } return outbuf; /* ERRORS */ empty_packet: { GST_ELEMENT_WARNING (rtpL24depay, STREAM, DECODE, ("Empty Payload."), (NULL)); return NULL; } reorder_failed: { GST_ELEMENT_ERROR (rtpL24depay, STREAM, DECODE, ("Channel reordering failed."), (NULL)); return NULL; } } gboolean gst_rtp_L24_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpL24depay", GST_RANK_SECONDARY, GST_TYPE_RTP_L24_DEPAY); }