/* GStreamer * Copyright (C) <2018> Havard Graff * Copyright (C) <2020-2021> Guillaume Desmottes * * 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 */ /** * SECTION:element-rtphdrextclientaudiolevel * @title: rtphdrextclientaudiolevel * @short_description: Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension * * Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension. * The extension should be automatically created by payloader and depayloaders, * if their `auto-header-extension` property is enabled, if the extension * is part of the RTP caps. * * ## Example pipeline * |[ * gst-launch-1.0 pulsesrc ! level audio-level-meta=true ! audiconvert ! * rtpL16pay ! application/x-rtp, * extmap-1=(string)\< \"\", urn:ietf:params:rtp-hdrext:ssrc-audio-level, * \"vad=on\" \> ! udpsink * ]| * * Since: 1.20 * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstrtphdrext-clientaudiolevel.h" #include #define CLIENT_AUDIO_LEVEL_HDR_EXT_URI GST_RTP_HDREXT_BASE"ssrc-audio-level" GST_DEBUG_CATEGORY_STATIC (rtphdrclient_audio_level_debug); #define GST_CAT_DEFAULT (rtphdrclient_audio_level_debug) #define DEFAULT_VAD TRUE enum { PROP_0, PROP_VAD, }; struct _GstRTPHeaderExtensionClientAudioLevel { GstRTPHeaderExtension parent; gboolean vad; }; G_DEFINE_TYPE_WITH_CODE (GstRTPHeaderExtensionClientAudioLevel, gst_rtp_header_extension_client_audio_level, GST_TYPE_RTP_HEADER_EXTENSION, GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "rtphdrextclientaudiolevel", 0, "RTP RFC 6464 Header Extensions");); GST_ELEMENT_REGISTER_DEFINE (rtphdrextclientaudiolevel, "rtphdrextclientaudiolevel", GST_RANK_MARGINAL, GST_TYPE_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL); static void gst_rtp_header_extension_client_audio_level_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRTPHeaderExtensionClientAudioLevel *self = GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (object); switch (prop_id) { case PROP_VAD: g_value_set_boolean (value, self->vad); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstRTPHeaderExtensionFlags gst_rtp_header_extension_client_audio_level_get_supported_flags (GstRTPHeaderExtension * ext) { return GST_RTP_HEADER_EXTENSION_ONE_BYTE | GST_RTP_HEADER_EXTENSION_TWO_BYTE; } static gsize gst_rtp_header_extension_client_audio_level_get_max_size (GstRTPHeaderExtension * ext, const GstBuffer * input_meta) { return 2; } static void set_vad (GstRTPHeaderExtension * ext, gboolean vad) { GstRTPHeaderExtensionClientAudioLevel *self = GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (ext); if (self->vad == vad) return; GST_DEBUG_OBJECT (ext, "vad: %d", vad); self->vad = vad; g_object_notify (G_OBJECT (self), "vad"); } static gboolean gst_rtp_header_extension_client_audio_level_set_attributes (GstRTPHeaderExtension * ext, GstRTPHeaderExtensionDirection direction, const gchar * attributes) { if (g_str_equal (attributes, "vad=on") || g_str_equal (attributes, "")) { set_vad (ext, TRUE); } else if (g_str_equal (attributes, "vad=off")) { set_vad (ext, FALSE); } else { GST_WARNING_OBJECT (ext, "Invalid attribute: %s", attributes); return FALSE; } return TRUE; } static gboolean gst_rtp_header_extension_client_audio_level_set_caps_from_attributes (GstRTPHeaderExtension * ext, GstCaps * caps) { GstRTPHeaderExtensionClientAudioLevel *self = GST_RTP_HEADER_EXTENSION_CLIENT_AUDIO_LEVEL (ext); const gchar *vad; if (self->vad) vad = "vad=on"; else vad = "vad=off"; return gst_rtp_header_extension_set_caps_from_attributes_helper (ext, caps, vad); } static gssize gst_rtp_header_extension_client_audio_level_write (GstRTPHeaderExtension * ext, const GstBuffer * input_meta, GstRTPHeaderExtensionFlags write_flags, GstBuffer * output, guint8 * data, gsize size) { GstAudioLevelMeta *meta; guint level; g_return_val_if_fail (size >= gst_rtp_header_extension_client_audio_level_get_max_size (ext, NULL), -1); g_return_val_if_fail (write_flags & gst_rtp_header_extension_client_audio_level_get_supported_flags (ext), -1); meta = gst_buffer_get_audio_level_meta ((GstBuffer *) input_meta); if (!meta) { GST_LOG_OBJECT (ext, "no meta"); return 0; } level = meta->level; if (level > 127) { GST_LOG_OBJECT (ext, "level from meta is higher than 127: %d, cropping", meta->level); level = 127; } GST_LOG_OBJECT (ext, "writing ext (level: %d voice: %d)", level, meta->voice_activity); /* Both one & two byte use the same format, the second byte being padding */ data[0] = (level & 0x7F) | (meta->voice_activity << 7); if (write_flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE) { return 1; } data[1] = 0; return 2; } static gboolean gst_rtp_header_extension_client_audio_level_read (GstRTPHeaderExtension * ext, GstRTPHeaderExtensionFlags read_flags, const guint8 * data, gsize size, GstBuffer * buffer) { guint8 level; gboolean voice_activity; g_return_val_if_fail (read_flags & gst_rtp_header_extension_client_audio_level_get_supported_flags (ext), -1); /* Both one & two byte use the same format, the second byte being padding */ level = data[0] & 0x7F; voice_activity = (data[0] & 0x80) >> 7; GST_LOG_OBJECT (ext, "reading ext (level: %d voice: %d)", level, voice_activity); gst_buffer_add_audio_level_meta (buffer, level, voice_activity); return TRUE; } static void gst_rtp_header_extension_client_audio_level_class_init (GstRTPHeaderExtensionClientAudioLevelClass * klass) { GstRTPHeaderExtensionClass *rtp_hdr_class; GstElementClass *gstelement_class; GObjectClass *gobject_class; rtp_hdr_class = GST_RTP_HEADER_EXTENSION_CLASS (klass); gobject_class = (GObjectClass *) klass; gstelement_class = GST_ELEMENT_CLASS (klass); gobject_class->get_property = gst_rtp_header_extension_client_audio_level_get_property; /** * rtphdrextclientaudiolevel:vad: * * If the vad extension attribute is enabled or not, default to %FALSE. * * Since: 1.20 */ g_object_class_install_property (gobject_class, PROP_VAD, g_param_spec_boolean ("vad", "vad", "If the vad extension attribute is enabled or not", DEFAULT_VAD, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); rtp_hdr_class->get_supported_flags = gst_rtp_header_extension_client_audio_level_get_supported_flags; rtp_hdr_class->get_max_size = gst_rtp_header_extension_client_audio_level_get_max_size; rtp_hdr_class->set_attributes = gst_rtp_header_extension_client_audio_level_set_attributes; rtp_hdr_class->set_caps_from_attributes = gst_rtp_header_extension_client_audio_level_set_caps_from_attributes; rtp_hdr_class->write = gst_rtp_header_extension_client_audio_level_write; rtp_hdr_class->read = gst_rtp_header_extension_client_audio_level_read; gst_element_class_set_static_metadata (gstelement_class, "Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension", GST_RTP_HDREXT_ELEMENT_CLASS, "Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension", "Guillaume Desmottes "); gst_rtp_header_extension_class_set_uri (rtp_hdr_class, CLIENT_AUDIO_LEVEL_HDR_EXT_URI); } static void gst_rtp_header_extension_client_audio_level_init (GstRTPHeaderExtensionClientAudioLevel * self) { GST_DEBUG_OBJECT (self, "creating element"); self->vad = DEFAULT_VAD; }