/* * Opus Payloader Gst Element * * @author: Danilo Cesar Lemes de Paula * * 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. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include "gstrtpopuspay.h" GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug); #define GST_CAT_DEFAULT (rtpopuspay_debug) static GstStaticPadTemplate gst_rtp_opus_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-opus, channels = (int) [1, 2], channel-mapping-family = (int) 0") ); static GstStaticPadTemplate gst_rtp_opus_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) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 48000, " "encoding-params = (string) \"2\", " "encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\" }") ); static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps); static GstCaps *gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad, GstCaps * filter); static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer); G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD); static void gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass) { GstRTPBasePayloadClass *gstbasertppayload_class; GstElementClass *element_class; gstbasertppayload_class = (GstRTPBasePayloadClass *) klass; element_class = GST_ELEMENT_CLASS (klass); gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps; gstbasertppayload_class->get_caps = gst_rtp_opus_pay_getcaps; gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer; gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_opus_pay_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_opus_pay_sink_template)); gst_element_class_set_static_metadata (element_class, "RTP Opus payloader", "Codec/Payloader/Network/RTP", "Puts Opus audio in RTP packets", "Danilo Cesar Lemes de Paula "); GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0, "Opus RTP Payloader"); } static void gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay) { } static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps) { gboolean res; GstCaps *src_caps; GstStructure *s; char *encoding_name; gint channels, rate; const char *sprop_stereo = NULL; char *sprop_maxcapturerate = NULL; src_caps = gst_pad_get_allowed_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload)); if (src_caps) { src_caps = gst_caps_make_writable (src_caps); src_caps = gst_caps_truncate (src_caps); s = gst_caps_get_structure (src_caps, 0); gst_structure_fixate_field_string (s, "encoding-name", "OPUS"); encoding_name = g_strdup (gst_structure_get_string (s, "encoding-name")); gst_caps_unref (src_caps); } else { encoding_name = g_strdup ("X-GST-OPUS-DRAFT-SPITTKA-00"); } s = gst_caps_get_structure (caps, 0); if (gst_structure_get_int (s, "channels", &channels)) { if (channels > 2) { GST_ERROR_OBJECT (payload, "More than 2 channels with channel-mapping-family=0 is invalid"); return FALSE; } else if (channels == 2) { sprop_stereo = "1"; } else { sprop_stereo = "0"; } } if (gst_structure_get_int (s, "rate", &rate)) { sprop_maxcapturerate = g_strdup_printf ("%d", rate); } gst_rtp_base_payload_set_options (payload, "audio", FALSE, encoding_name, 48000); g_free (encoding_name); if (sprop_maxcapturerate && sprop_stereo) { res = gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate", G_TYPE_STRING, sprop_maxcapturerate, "sprop-stereo", G_TYPE_STRING, sprop_stereo, NULL); } else if (sprop_maxcapturerate) { res = gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate", G_TYPE_STRING, sprop_maxcapturerate, NULL); } else if (sprop_stereo) { res = gst_rtp_base_payload_set_outcaps (payload, "sprop-stereo", G_TYPE_STRING, sprop_stereo, NULL); } else { res = gst_rtp_base_payload_set_outcaps (payload, NULL); } g_free (sprop_maxcapturerate); return res; } typedef struct { GstRtpOPUSPay *pay; GstBuffer *outbuf; } CopyMetaData; static gboolean foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data) { CopyMetaData *data = user_data; GstRtpOPUSPay *pay = data->pay; GstBuffer *outbuf = data->outbuf; const GstMetaInfo *info = (*meta)->info; const gchar *const *tags = gst_meta_api_type_get_tags (info->api); if (!tags || (g_strv_length ((gchar **) tags) == 1 && gst_meta_api_type_has_tag (info->api, g_quark_from_string (GST_META_TAG_AUDIO_STR)))) { GstMetaTransformCopy copy_data = { FALSE, 0, -1 }; GST_DEBUG_OBJECT (pay, "copy metadata %s", g_type_name (info->api)); /* simply copy then */ info->transform_func (outbuf, *meta, inbuf, _gst_meta_transform_copy, ©_data); } else { GST_DEBUG_OBJECT (pay, "not copying metadata %s", g_type_name (info->api)); } return TRUE; } static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload, GstBuffer * buffer) { GstBuffer *outbuf; GstClockTime pts, dts, duration; CopyMetaData data; pts = GST_BUFFER_PTS (buffer); dts = GST_BUFFER_DTS (buffer); duration = GST_BUFFER_DURATION (buffer); outbuf = gst_rtp_buffer_new_allocate (0, 0, 0); data.pay = GST_RTP_OPUS_PAY (basepayload); data.outbuf = outbuf; gst_buffer_foreach_meta (buffer, foreach_metadata, &data); outbuf = gst_buffer_append (outbuf, buffer); GST_BUFFER_PTS (outbuf) = pts; GST_BUFFER_DTS (outbuf) = dts; GST_BUFFER_DURATION (outbuf) = duration; /* Push out */ return gst_rtp_base_payload_push (basepayload, outbuf); } static GstCaps * gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad, GstCaps * filter) { GstCaps *caps, *peercaps, *tcaps; GstStructure *s; const gchar *stereo; if (pad == GST_RTP_BASE_PAYLOAD_SRCPAD (payload)) return GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps (payload, pad, filter); tcaps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload)); peercaps = gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), tcaps); gst_caps_unref (tcaps); if (!peercaps) return GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps (payload, pad, filter); if (gst_caps_is_empty (peercaps)) return peercaps; caps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SINKPAD (payload)); s = gst_caps_get_structure (peercaps, 0); stereo = gst_structure_get_string (s, "stereo"); if (stereo != NULL) { caps = gst_caps_make_writable (caps); if (!strcmp (stereo, "1")) { GstCaps *caps2 = gst_caps_copy (caps); gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL); gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 1, NULL); caps = gst_caps_merge (caps, caps2); } else if (!strcmp (stereo, "0")) { GstCaps *caps2 = gst_caps_copy (caps); gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL); gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 2, NULL); caps = gst_caps_merge (caps, caps2); } } gst_caps_unref (peercaps); if (filter) { GstCaps *tmp = gst_caps_intersect_full (caps, filter, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); caps = tmp; } GST_DEBUG_OBJECT (payload, "Returning caps: %" GST_PTR_FORMAT, caps); return caps; } gboolean gst_rtp_opus_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpopuspay", GST_RANK_PRIMARY, GST_TYPE_RTP_OPUS_PAY); }