mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-11 10:56:38 +00:00
0dbe0e21fe
Multiplying elements named after RFC numbers is confusing, so let's give them meaningful names. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1125>
370 lines
10 KiB
C
370 lines
10 KiB
C
/* GStreamer
|
|
*
|
|
* unit test for RTP RFC 6464 Header Extensions
|
|
*
|
|
* Copyright (C) <2020-2021> Guillaume Desmottes <guillaume.desmottes@collabora.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.
|
|
*/
|
|
|
|
#include <gst/check/gstcheck.h>
|
|
#include <gst/rtp/rtp.h>
|
|
#include <gst/sdp/gstsdpmessage.h>
|
|
#include <gst/audio/audio.h>
|
|
#include <gst/check/gstharness.h>
|
|
|
|
#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)
|