/* GStreamer * * unit test for RTP RFC 6464 Header Extensions * * 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 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. */ #include #include #include #include #include #define URN "urn:ietf:params:rtp-hdrext:ssrc-audio-level" #define SDP "v=0\r\n" \ "o=- 123456 2 IN IP4 127.0.0.1 \r\n" \ "s=-\r\n" \ "t=0 0\r\n" \ "a=maxptime:60\r\n" \ "a=sendrecv\r\n" \ "m=audio 55815 RTP/SAVPF 100\r\n" \ "c=IN IP4 1.1.1.1\r\n" \ "a=rtpmap:100 opus/48000/2\r\n" #define SDP_NO_VAD SDP \ "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\n" #define SDP_VAD_ON SDP \ "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=on\r\n" #define SDP_VAD_OFF SDP \ "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=off\r\n" #define SDP_VAD_WRONG SDP \ "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=badger\r\n" static GstCaps * create_caps (const gchar * sdp) { GstSDPMessage *message; glong length = -1; const GstSDPMedia *media; GstCaps *caps; gst_sdp_message_new (&message); gst_sdp_message_parse_buffer ((guint8 *) sdp, length, message); media = gst_sdp_message_get_media (message, 0); fail_unless (media != NULL); caps = gst_sdp_media_get_caps_from_media (media, 100); gst_sdp_media_attributes_to_caps (media, caps); gst_sdp_message_free (message); return caps; } static void check_caps (GstRTPHeaderExtension * ext, gboolean vad) { GstCaps *caps; GstStructure *s; const GValue *arr, *val; caps = gst_caps_new_empty_simple ("application/x-rtp"); fail_unless (gst_rtp_header_extension_set_caps_from_attributes (ext, caps)); s = gst_caps_get_structure (caps, 0); arr = gst_structure_get_value (s, "extmap-1"); fail_unless (arr != NULL); fail_unless (GST_VALUE_HOLDS_ARRAY (arr)); fail_unless (gst_value_array_get_size (arr) == 3); val = gst_value_array_get_value (arr, 0); fail_unless_equals_string (g_value_get_string (val), ""); val = gst_value_array_get_value (arr, 1); fail_unless_equals_string (g_value_get_string (val), URN); val = gst_value_array_get_value (arr, 2); if (vad) { fail_unless_equals_string (g_value_get_string (val), "vad=on"); } else { fail_unless_equals_string (g_value_get_string (val), "vad=off"); } gst_caps_unref (caps); } GST_START_TEST (rtphdrext_client_audio_level_sdp) { GstRTPHeaderExtension *ext; GstCaps *caps; gboolean vad = FALSE; ext = gst_rtp_header_extension_create_from_uri (URN); fail_unless (ext != NULL); gst_rtp_header_extension_set_id (ext, 1); /* vad default to on */ caps = create_caps (SDP_NO_VAD); fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps)); gst_caps_unref (caps); g_object_get (ext, "vad", &vad, NULL); fail_unless (vad); check_caps (ext, TRUE); /* vad is disabled */ caps = create_caps (SDP_VAD_OFF); fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps)); gst_caps_unref (caps); g_object_get (ext, "vad", &vad, NULL); fail_if (vad); /* vad is enabled */ caps = create_caps (SDP_VAD_ON); fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps)); gst_caps_unref (caps); g_object_get (ext, "vad", &vad, NULL); fail_unless (vad); /* invalid vad */ caps = create_caps (SDP_VAD_WRONG); fail_if (gst_rtp_header_extension_set_attributes_from_caps (ext, caps)); gst_caps_unref (caps); gst_object_unref (ext); } GST_END_TEST; GST_START_TEST (rtphdrext_client_audio_level_one_byte) { GstRTPHeaderExtension *ext; GstRTPHeaderExtensionFlags flags; GstBuffer *buffer; guint8 *data; gsize size, written; GstAudioLevelMeta *meta; guint8 level = 12; gboolean voice = TRUE; ext = gst_rtp_header_extension_create_from_uri (URN); fail_unless (ext != NULL); gst_rtp_header_extension_set_id (ext, 1); flags = gst_rtp_header_extension_get_supported_flags (ext); fail_unless (flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE); buffer = gst_buffer_new (); meta = gst_buffer_add_audio_level_meta (buffer, level, voice); size = gst_rtp_header_extension_get_max_size (ext, buffer); fail_unless (size > 0); data = g_malloc0 (size); fail_unless (data != NULL); /* Write extension */ written = gst_rtp_header_extension_write (ext, buffer, GST_RTP_HEADER_EXTENSION_ONE_BYTE, buffer, data, size); fail_unless (written == 1); /* Read it back */ fail_unless (gst_buffer_remove_meta (buffer, (GstMeta *) meta)); fail_unless (gst_rtp_header_extension_read (ext, GST_RTP_HEADER_EXTENSION_ONE_BYTE, data, size, buffer)); meta = gst_buffer_get_audio_level_meta (buffer); fail_unless (meta != NULL); fail_unless_equals_int (meta->level, level); fail_unless (meta->voice_activity == voice); g_free (data); gst_buffer_unref (buffer); gst_object_unref (ext); } GST_END_TEST; GST_START_TEST (rtphdrext_client_audio_level_two_bytes) { GstRTPHeaderExtension *ext; GstRTPHeaderExtensionFlags flags; GstBuffer *buffer; guint8 *data; gsize size, written; GstAudioLevelMeta *meta; guint8 level = 12; gboolean voice = TRUE; ext = gst_rtp_header_extension_create_from_uri (URN); fail_unless (ext != NULL); gst_rtp_header_extension_set_id (ext, 1); flags = gst_rtp_header_extension_get_supported_flags (ext); fail_unless (flags & GST_RTP_HEADER_EXTENSION_TWO_BYTE); buffer = gst_buffer_new (); meta = gst_buffer_add_audio_level_meta (buffer, level, voice); size = gst_rtp_header_extension_get_max_size (ext, buffer); fail_unless (size > 0); data = g_malloc0 (size); fail_unless (data != NULL); /* Write extension */ written = gst_rtp_header_extension_write (ext, buffer, GST_RTP_HEADER_EXTENSION_TWO_BYTE, buffer, data, size); fail_unless (written == 2); /* Read it back */ fail_unless (gst_buffer_remove_meta (buffer, (GstMeta *) meta)); fail_unless (gst_rtp_header_extension_read (ext, GST_RTP_HEADER_EXTENSION_TWO_BYTE, data, size, buffer)); meta = gst_buffer_get_audio_level_meta (buffer); fail_unless (meta != NULL); fail_unless_equals_int (meta->level, level); fail_unless (meta->voice_activity == voice); g_free (data); gst_buffer_unref (buffer); gst_object_unref (ext); } GST_END_TEST; GST_START_TEST (rtphdrext_client_audio_level_no_meta) { GstRTPHeaderExtension *ext; GstBuffer *buffer; guint8 *data; gsize size, written; ext = gst_rtp_header_extension_create_from_uri (URN); fail_unless (ext != NULL); gst_rtp_header_extension_set_id (ext, 1); buffer = gst_buffer_new (); size = gst_rtp_header_extension_get_max_size (ext, buffer); fail_unless (size > 0); data = g_malloc0 (size); fail_unless (data != NULL); written = gst_rtp_header_extension_write (ext, buffer, GST_RTP_HEADER_EXTENSION_ONE_BYTE, buffer, data, size); fail_unless (written == 0); written = gst_rtp_header_extension_write (ext, buffer, GST_RTP_HEADER_EXTENSION_TWO_BYTE, buffer, data, size); fail_unless (written == 0); g_free (data); gst_buffer_unref (buffer); gst_object_unref (ext); } GST_END_TEST; GST_START_TEST (rtphdrext_client_audio_level_payloader_depayloader) { GstHarness *h; GstBuffer *b; GstFlowReturn fret; GstAudioLevelMeta *meta; h = gst_harness_new_parse ("rtpL16pay ! " "application/x-rtp, extmap-1=(string)< \"\", " URN " , \"vad=on\" >" " ! rtpL16depay"); gst_harness_set_src_caps_str (h, "audio/x-raw, rate=44100, channels=1," " layout=interleaved, format=S16BE"); b = gst_buffer_new_allocate (NULL, 100, NULL); gst_buffer_add_audio_level_meta (b, 12, TRUE); fret = gst_harness_push (h, b); fail_unless (fret == GST_FLOW_OK); b = gst_harness_pull (h); meta = gst_buffer_get_audio_level_meta (b); fail_unless (meta != NULL); fail_unless (meta->level == 12); fail_unless (meta->voice_activity == TRUE); gst_buffer_unref (b); gst_harness_teardown (h); } GST_END_TEST; GST_START_TEST (rtphdrext_client_audio_level_payloader_api) { GstHarness *h; GstRTPHeaderExtension *ext; GstBuffer *b; GstFlowReturn fret; GstRTPBuffer rtp = GST_RTP_BUFFER_INIT; guint8 *data; guint size; guint8 level; gboolean voice_activity; h = gst_harness_new ("rtpL16pay"); gst_harness_set_src_caps_str (h, "audio/x-raw, rate=44100, channels=1," " layout=interleaved, format=S16BE"); ext = gst_rtp_header_extension_create_from_uri (URN); gst_rtp_header_extension_set_id (ext, 2); fail_unless (ext); g_signal_emit_by_name (h->element, "add-extension", ext); b = gst_buffer_new_allocate (NULL, 100, NULL); gst_buffer_add_audio_level_meta (b, 12, TRUE); fret = gst_harness_push (h, b); fail_unless (fret == GST_FLOW_OK); b = gst_harness_pull (h); fail_unless (gst_rtp_buffer_map (b, GST_MAP_READ, &rtp)); fail_unless (gst_rtp_buffer_get_extension_onebyte_header (&rtp, 2, 0, (gpointer *) & data, &size)); fail_unless (size == 1); level = data[0] & 0x7F; voice_activity = (data[0] & 0x80) >> 7; fail_unless (level == 12); fail_unless (voice_activity == TRUE); gst_rtp_buffer_unmap (&rtp); gst_buffer_unref (b); gst_object_unref (ext); gst_harness_teardown (h); } GST_END_TEST; static Suite * rtphdrext_client_audio_level_suite (void) { Suite *s = suite_create ("rtphdrext_client_audio_level"); TCase *tc_chain = tcase_create ("general"); suite_add_tcase (s, tc_chain); tcase_add_test (tc_chain, rtphdrext_client_audio_level_sdp); tcase_add_test (tc_chain, rtphdrext_client_audio_level_one_byte); tcase_add_test (tc_chain, rtphdrext_client_audio_level_two_bytes); tcase_add_test (tc_chain, rtphdrext_client_audio_level_no_meta); tcase_add_test (tc_chain, rtphdrext_client_audio_level_payloader_depayloader); tcase_add_test (tc_chain, rtphdrext_client_audio_level_payloader_api); return s; } GST_CHECK_MAIN (rtphdrext_client_audio_level)