/* 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-rtpL16pay * @title: rtpL16pay * @see_also: rtpL16depay * * Payload raw audio into RTP packets according to RFC 3551. * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt * * ## Example pipeline * |[ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink * ]| This example pipeline will payload raw audio. Refer to * the rtpL16depay example to depayload and play the RTP stream. * */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include <string.h> #include <gst/audio/audio.h> #include <gst/rtp/gstrtpbuffer.h> #include "gstrtpelements.h" #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, " "format = (string) S16BE, " "layout = (string) interleaved, " "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 (GstRTPBasePayload * basepayload, GstCaps * caps); static GstCaps *gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad, GstCaps * filter); static GstFlowReturn gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload, GstBuffer * buffer); #define gst_rtp_L16_pay_parent_class parent_class G_DEFINE_TYPE (GstRtpL16Pay, gst_rtp_L16_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpL16pay, "rtpL16pay", GST_RANK_SECONDARY, GST_TYPE_RTP_L16_PAY, rtp_element_init (plugin)); static void gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass) { GstElementClass *gstelement_class; GstRTPBasePayloadClass *gstrtpbasepayload_class; gstelement_class = (GstElementClass *) klass; gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; gstrtpbasepayload_class->set_caps = gst_rtp_L16_pay_setcaps; gstrtpbasepayload_class->get_caps = gst_rtp_L16_pay_getcaps; gstrtpbasepayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer; gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_L16_pay_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_L16_pay_sink_template); gst_element_class_set_static_metadata (gstelement_class, "RTP audio payloader", "Codec/Payloader/Network/RTP", "Payload-encode Raw audio into RTP packets (RFC 3551)", "Wim Taymans <wim.taymans@gmail.com>"); GST_DEBUG_CATEGORY_INIT (rtpL16pay_debug, "rtpL16pay", 0, "L16 RTP Payloader"); } static void gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay) { GstRTPBaseAudioPayload *rtpbaseaudiopayload; rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL16pay); /* tell rtpbaseaudiopayload that this is a sample based codec */ gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); } static gboolean gst_rtp_L16_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps) { GstRtpL16Pay *rtpL16pay; gboolean res; gchar *params; GstAudioInfo *info; const GstRTPChannelOrder *order; GstRTPBaseAudioPayload *rtpbaseaudiopayload; rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload); rtpL16pay = GST_RTP_L16_PAY (basepayload); info = &rtpL16pay->info; gst_audio_info_init (info); if (!gst_audio_info_from_caps (info, caps)) goto invalid_caps; order = gst_rtp_channels_get_by_pos (info->channels, info->position); rtpL16pay->order = order; gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L16", info->rate); params = g_strdup_printf ("%d", info->channels); if (!order && info->channels > 2) { GST_ELEMENT_WARNING (rtpL16pay, STREAM, DECODE, (NULL), ("Unknown channel order for %d channels", info->channels)); } if (order && order->name) { res = gst_rtp_base_payload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, info->channels, "channel-order", G_TYPE_STRING, order->name, NULL); } else { res = gst_rtp_base_payload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, info->channels, NULL); } g_free (params); /* octet-per-sample is 2 * channels for L16 */ gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, 2 * info->channels); return res; /* ERRORS */ invalid_caps: { GST_DEBUG_OBJECT (rtpL16pay, "invalid caps"); return FALSE; } } static GstCaps * gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad, GstCaps * filter) { GstCaps *otherpadcaps; GstCaps *caps; caps = gst_pad_get_pad_template_caps (pad); otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad); if (otherpadcaps) { if (!gst_caps_is_empty (otherpadcaps)) { GstStructure *structure; gint channels; gint pt; gint rate; structure = gst_caps_get_structure (otherpadcaps, 0); caps = gst_caps_make_writable (caps); 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 == GST_RTP_PAYLOAD_L16_STEREO) gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL); else if (pt == GST_RTP_PAYLOAD_L16_MONO) 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 == GST_RTP_PAYLOAD_L16_STEREO || pt == GST_RTP_PAYLOAD_L16_MONO) 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; } static GstFlowReturn gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload, GstBuffer * buffer) { GstRtpL16Pay *rtpL16pay; rtpL16pay = GST_RTP_L16_PAY (basepayload); buffer = gst_buffer_make_writable (buffer); if (rtpL16pay->order && !gst_audio_buffer_reorder_channels (buffer, rtpL16pay->info.finfo->format, rtpL16pay->info.channels, rtpL16pay->info.position, rtpL16pay->order->pos)) { return GST_FLOW_ERROR; } return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload, buffer); }