/* * Opus Depayloader 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 #include "gstrtpelements.h" #include "gstrtpopusdepay.h" #include "gstrtputils.h" GST_DEBUG_CATEGORY_STATIC (rtpopusdepay_debug); #define GST_CAT_DEFAULT (rtpopusdepay_debug) static GstStaticPadTemplate gst_rtp_opus_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) 48000, " "encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\", \"MULTIOPUS\" }") ); static GstStaticPadTemplate gst_rtp_opus_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-opus, channel-mapping-family = (int) [ 0, 1 ]") ); static GstBuffer *gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp_buffer); static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps); G_DEFINE_TYPE (GstRTPOpusDepay, gst_rtp_opus_depay, GST_TYPE_RTP_BASE_DEPAYLOAD); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpopusdepay, "rtpopusdepay", GST_RANK_PRIMARY, GST_TYPE_RTP_OPUS_DEPAY, rtp_element_init (plugin)); static void gst_rtp_opus_depay_class_init (GstRTPOpusDepayClass * klass) { GstRTPBaseDepayloadClass *gstbasertpdepayload_class; GstElementClass *element_class; element_class = GST_ELEMENT_CLASS (klass); gstbasertpdepayload_class = (GstRTPBaseDepayloadClass *) klass; gst_element_class_add_static_pad_template (element_class, &gst_rtp_opus_depay_src_template); gst_element_class_add_static_pad_template (element_class, &gst_rtp_opus_depay_sink_template); gst_element_class_set_static_metadata (element_class, "RTP Opus packet depayloader", "Codec/Depayloader/Network/RTP", "Extracts Opus audio from RTP packets", "Danilo Cesar Lemes de Paula "); gstbasertpdepayload_class->process_rtp_packet = gst_rtp_opus_depay_process; gstbasertpdepayload_class->set_caps = gst_rtp_opus_depay_setcaps; GST_DEBUG_CATEGORY_INIT (rtpopusdepay_debug, "rtpopusdepay", 0, "Opus RTP Depayloader"); } static void gst_rtp_opus_depay_init (GstRTPOpusDepay * rtpopusdepay) { } static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) { GstCaps *srccaps; GstStructure *s; gboolean ret; const gchar *sprop_maxcapturerate; srccaps = gst_caps_new_empty_simple ("audio/x-opus"); s = gst_caps_get_structure (caps, 0); if (g_str_equal (gst_structure_get_string (s, "encoding-name"), "MULTIOPUS")) { gint channels; gint stream_count; gint coupled_count; const gchar *encoding_params; const gchar *num_streams; const gchar *coupled_streams; const gchar *channel_mapping; gchar *endptr; if (!gst_structure_has_field_typed (s, "encoding-params", G_TYPE_STRING) || !gst_structure_has_field_typed (s, "num_streams", G_TYPE_STRING) || !gst_structure_has_field_typed (s, "coupled_streams", G_TYPE_STRING) || !gst_structure_has_field_typed (s, "channel_mapping", G_TYPE_STRING)) { GST_WARNING_OBJECT (depayload, "Encoding name 'MULTIOPUS' requires " "encoding-params, num_streams, coupled_streams and channel_mapping " "as string fields in caps."); goto reject_caps; } gst_caps_set_simple (srccaps, "channel-mapping-family", G_TYPE_INT, 1, NULL); encoding_params = gst_structure_get_string (s, "encoding-params"); channels = g_ascii_strtoull (encoding_params, &endptr, 10); if (*endptr != '\0' || channels > 255) { GST_WARNING_OBJECT (depayload, "Invalid encoding-params value '%s'", encoding_params); goto reject_caps; } gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, channels, NULL); num_streams = gst_structure_get_string (s, "num_streams"); stream_count = g_ascii_strtoull (num_streams, &endptr, 10); if (*endptr != '\0' || stream_count > channels) { GST_WARNING_OBJECT (depayload, "Invalid num_streams value '%s'", num_streams); goto reject_caps; } gst_caps_set_simple (srccaps, "stream-count", G_TYPE_INT, stream_count, NULL); coupled_streams = gst_structure_get_string (s, "coupled_streams"); coupled_count = g_ascii_strtoull (coupled_streams, &endptr, 10); if (*endptr != '\0' || coupled_count > stream_count) { GST_WARNING_OBJECT (depayload, "Invalid coupled_streams value '%s'", coupled_streams); goto reject_caps; } gst_caps_set_simple (srccaps, "coupled-count", G_TYPE_INT, coupled_count, NULL); channel_mapping = gst_structure_get_string (s, "channel_mapping"); { gchar **split; gchar **ptr; GValue mapping = G_VALUE_INIT; GValue v = G_VALUE_INIT; split = g_strsplit (channel_mapping, ",", -1); g_value_init (&mapping, GST_TYPE_ARRAY); g_value_init (&v, G_TYPE_INT); for (ptr = split; *ptr; ++ptr) { gint channel = g_ascii_strtoull (*ptr, &endptr, 10); if (*endptr != '\0' || channel > channels) { GST_WARNING_OBJECT (depayload, "Invalid channel_mapping value '%s'", channel_mapping); g_value_unset (&mapping); break; } g_value_set_int (&v, channel); gst_value_array_append_value (&mapping, &v); } g_value_unset (&v); g_strfreev (split); if (G_IS_VALUE (&mapping)) { gst_caps_set_value (srccaps, "channel-mapping", &mapping); g_value_unset (&mapping); } else { goto reject_caps; } } } else { const gchar *sprop_stereo; gst_caps_set_simple (srccaps, "channel-mapping-family", G_TYPE_INT, 0, NULL); if ((sprop_stereo = gst_structure_get_string (s, "sprop-stereo"))) { if (strcmp (sprop_stereo, "0") == 0) gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 1, NULL); else if (strcmp (sprop_stereo, "1") == 0) gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 2, NULL); else GST_WARNING_OBJECT (depayload, "Unknown sprop-stereo value '%s'", sprop_stereo); } else { /* Although sprop-stereo defaults to mono as per RFC 7587, this just means that the signal is likely mono and can be safely downmixed, it may still be stereo at times. */ gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 2, NULL); } } if ((sprop_maxcapturerate = gst_structure_get_string (s, "sprop-maxcapturerate"))) { gulong rate; gchar *tailptr; rate = strtoul (sprop_maxcapturerate, &tailptr, 10); if (rate > INT_MAX || *tailptr != '\0') { GST_WARNING_OBJECT (depayload, "Failed to parse sprop-maxcapturerate value '%s'", sprop_maxcapturerate); } else { gst_caps_set_simple (srccaps, "rate", G_TYPE_INT, rate, NULL); } } ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps); GST_DEBUG_OBJECT (depayload, "set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret); gst_caps_unref (srccaps); depayload->clock_rate = 48000; return ret; reject_caps: gst_caps_unref (srccaps); return FALSE; } static GstBuffer * gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp_buffer) { GstBuffer *outbuf; outbuf = gst_rtp_buffer_get_payload_buffer (rtp_buffer); gst_rtp_drop_non_audio_meta (depayload, outbuf); return outbuf; }