Test suite failure when running with address sanitization
ninja: Entering directory `_build'
[1/21] Generating version.h with a custom command
[1/2] Running all tests.
1/12 Validate desktop file OK 0.01s
2/12 Validate daemon desktop file OK 0.01s
3/12 Validate appstream file OK 0.04s
4/12 Validate schema file OK 0.01s
5/12 provider OK 0.13s
7/12 call OK 0.13s
9/12 plugins OK 0.12s
12/12 util OK 0.12s
The output from the failed tests:
6/12 origin FAIL 0.27s (exit status 1)
--- command ---
07:13:09 G_TEST_BUILDDIR='/home/fortysixandtwo/git/calls/_build/tests' PYTHONDONTWRITEBYTECODE='yes' MALLOC_CHECK_='2' G_TEST_SRCDIR='/home/fortysixandtwo/git/calls/tests' NO_AT_BRIDGE='1' CALLS_AUDIOSINK='fakesink' CALLS_AUDIOSRC='audiotestsrc' GSETTINGS_BACKEND='memory' G_DEBUG='gc-friendly,fatal-warnings' /home/fortysixandtwo/git/calls/_build/tests/origin
--- stdout ---
# random seed: R02Sd2dffecdc3dc14ad22335b0e37ae864b
# GLib-GIO-DEBUG: _g_io_module_get_default: Found default implementation memory (GMemorySettingsBackend) for ?gsettings-backend?
# GLib-DEBUG: unsetenv() is not thread-safe and should not be used after threads are created
# GLib-GIO-DEBUG: _g_io_module_get_default: Found default implementation gvfs (GDaemonVfs) for ?gio-vfs?
1..3
# Start of Calls tests
# Start of Origin tests
ok 1 /Calls/Origin/object
ok 2 /Calls/Origin/get_name
ok 3 /Calls/Origin/calls
# End of Origin tests
# End of Calls tests
--- stderr ---
=================================================================
==1015912==ERROR: LeakSanitizer: detected memory leaks
Direct leak of 13 byte(s) in 1 object(s) allocated from:
#0 0x7f8ba1fa8e8f in __interceptor_malloc ../../../../src/libsanitizer/asan/asan_malloc_linux.cpp:145
#1 0x7f8ba1dcdd48 in g_malloc ../../../glib/gmem.c:106
SUMMARY: AddressSanitizer: 13 byte(s) leaked in 1 allocation(s).
-------
8/12 manager FAIL 0.32s (exit status 1)
--- command ---
07:13:09 G_TEST_BUILDDIR='/home/fortysixandtwo/git/calls/_build/tests' PYTHONDONTWRITEBYTECODE='yes' MALLOC_CHECK_='2' G_TEST_SRCDIR='/home/fortysixandtwo/git/calls/tests' NO_AT_BRIDGE='1' CALLS_AUDIOSINK='fakesink' CALLS_AUDIOSRC='audiotestsrc' GSETTINGS_BACKEND='memory' G_DEBUG='gc-friendly,fatal-warnings' /home/fortysixandtwo/git/calls/_build/tests/manager
--- stdout ---
# random seed: R02S456f2bb777b108103df184bce560d2ba
# GLib-GIO-DEBUG: _g_io_module_get_default: Found default implementation memory (GMemorySettingsBackend) for ?gsettings-backend?
# GLib-DEBUG: unsetenv() is not thread-safe and should not be used after threads are created
# GLib-GIO-DEBUG: _g_io_module_get_default: Found default implementation gvfs (GDaemonVfs) for ?gio-vfs?
1..4
# Start of Calls tests
# Start of Manager tests
# CallsManager-DEBUG: Scanning for plugins in `/home/fortysixandtwo/install/lib/x86_64-linux-gnu/calls/plugins'
ok 1 /Calls/Manager/without_provider
# CallsManager-DEBUG: Scanning for plugins in `/home/fortysixandtwo/install/lib/x86_64-linux-gnu/calls/plugins'
# CallsProvider-DEBUG: Loaded plugin `dummy'
# CallsProvider-DEBUG: Created provider from plugin `dummy'
# CallsManager-DEBUG: origins changed: pos=0 rem=0 added=1 total=1
# CallsManager-DEBUG: before adding: 0
# CallsManager-DEBUG: Adding origin Dummy origin (0x619000370390)
# CallsManager-DEBUG: after adding: 1
# CallsManager-DEBUG: Remove provider: dummy
# CallsManager-DEBUG: Removing origin Dummy origin (0x619000370390)
ok 2 /Calls/Manager/dummy_provider
# CallsManager-DEBUG: Scanning for plugins in `/home/fortysixandtwo/install/lib/x86_64-linux-gnu/calls/plugins'
# CallsProvider-DEBUG: Loaded plugin `mm'
# CallsMMProvider-DEBUG: Watching for ModemManager
# CallsProvider-DEBUG: Created provider from plugin `mm'
# CallsManager-DEBUG: origins changed: pos=0 rem=0 added=0 total=0
# CallsManager-DEBUG: Remove provider: mm
ok 3 /Calls/Manager/mm_provider
# CallsManager-DEBUG: Scanning for plugins in `/home/fortysixandtwo/install/lib/x86_64-linux-gnu/calls/plugins'
# CallsProvider-DEBUG: Loaded plugin `sip'
# CallsProvider-DEBUG: Created provider from plugin `sip'
# CallsManager-DEBUG: origins changed: pos=0 rem=0 added=0 total=0
# CallsAccountProvider-DEBUG: Trying to add account for Alice
# CallsSipOrigin-DEBUG: Updating credentials
# CallsSipOrigin-DEBUG: Protocol not set, falling back to 'UDP'
# CallsSipMediaManager-DEBUG: Creating CallsSipMediaManager
# CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMA is available
# CallsGstRfc3551-DEBUG: Adding PCMA to the codec candidates
# CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMU is available
# CallsGstRfc3551-DEBUG: Adding PCMU to the codec candidates
# CallsGstRfc3551-DEBUG: Gstreamer plugin for GSM is available
# CallsGstRfc3551-DEBUG: Adding GSM to the codec candidates
# CallsGstRfc3551-DEBUG: Gstreamer plugin for G722 is available
# CallsGstRfc3551-DEBUG: Adding G722 to the codec candidates
# CallsGstRfc3551-DEBUG: Gstreamer plugin for G723 is available
# CallsGstRfc3551-DEBUG: Adding G723 to the codec candidates
# CallsManager-DEBUG: origins changed: pos=0 rem=0 added=1 total=1
# CallsManager-DEBUG: before adding: 0
# CallsManager-DEBUG: Adding origin Alice (0x61d0002f5dc0)
# CallsManager-DEBUG: after adding: 1
# CallsAccountProvider-DEBUG: Trying to add account for Alice
# CallsSipProvider-DEBUG: Cannot add credentials with name 'Alice' multiple times
# CallsAccountProvider-DEBUG: Trying to add account for Bob
# CallsSipOrigin-DEBUG: Updating credentials
# CallsSipOrigin-DEBUG: Protocol not set, falling back to 'UDP'
# CallsManager-DEBUG: origins changed: pos=1 rem=0 added=1 total=2
# CallsManager-DEBUG: before adding: 1
# CallsManager-DEBUG: Adding origin Bob (0x61d0002f66b0)
# CallsManager-DEBUG: after adding: 2
# CallsProvider-DEBUG: Loaded plugin `mm'
# CallsMMProvider-DEBUG: Watching for ModemManager
# CallsProvider-DEBUG: Created provider from plugin `mm'
# CallsManager-DEBUG: origins changed: pos=0 rem=0 added=0 total=0
# CallsAccountProvider-DEBUG: Trying to remove account from Alice
# CallsManager-DEBUG: origins changed: pos=0 rem=1 added=0 total=1
# CallsManager-DEBUG: Removing origin Alice (0x61d0002f5dc0)
# CallsAccountProvider-DEBUG: Trying to remove account from Alice
# CallsManager-DEBUG: Remove provider: mm
# CallsAccountProvider-DEBUG: Trying to remove account from Bob
# CallsManager-DEBUG: origins changed: pos=0 rem=1 added=0 total=0
# CallsManager-DEBUG: Removing origin Bob (0x61d0002f66b0)
ok 4 /Calls/Manager/multiple_provider_mm_sip
# End of Manager tests
# End of Calls tests
--- stderr ---
su_source_port_create() returns 0x61400002bea0
=================================================================
==1015923==ERROR: LeakSanitizer: detected memory leaks
Direct leak of 16384 byte(s) in 1 object(s) allocated from:
#0 0x7fcbbd323e8f in __interceptor_malloc ../../../../src/libsanitizer/asan/asan_malloc_linux.cpp:145
#1 0x7fcbbd148d48 in g_malloc ../../../glib/gmem.c:106
SUMMARY: AddressSanitizer: 16384 byte(s) leaked in 1 allocation(s).
-------
10/12 sip FAIL 0.42s (exit status 1)
--- command ---
07:13:09 G_TEST_BUILDDIR='/home/fortysixandtwo/git/calls/_build/tests' PYTHONDONTWRITEBYTECODE='yes' MALLOC_CHECK_='2' G_TEST_SRCDIR='/home/fortysixandtwo/git/calls/tests' NO_AT_BRIDGE='1' CALLS_AUDIOSINK='fakesink' CALLS_AUDIOSRC='audiotestsrc' GSETTINGS_BACKEND='memory' G_DEBUG='gc-friendly,fatal-warnings' /home/fortysixandtwo/git/calls/_build/tests/sip
--- Listing only the last 100 lines from a long log. ---
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-send-src has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element encoder has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element encoder has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-recv-src has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-recv-src has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element source has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element depayloader has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element source has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element depayloader has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element rtp-send-pipeline has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element rtp-recv-pipeline has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element rtp-send-pipeline has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element rtp-recv-pipeline has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-send-sink has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-recv-sink has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-send-sink has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-recv-sink has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpstorage9 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpssrcdemux8 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpstorage7 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpssrcdemux6 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpssrcdemux9 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpstorage8 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpssrcdemux7 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpstorage6 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpsession9 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpsession8 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpsession7 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtpsession6 has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element send-rtpbin has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element recv-rtpbin has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element send-rtpbin has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element recv-rtpbin has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element payloader has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element decoder has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element payloader has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element decoder has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-send-src has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-send-src has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-src has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element encoder has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-recv-src has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element encoder has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtcp-udp-recv-src has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element source has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element depayloader has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element source has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element depayloader has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-sink has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-sink has changed state from READY to PAUSED
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-sink has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtp-udp-sink has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtp-send-pipeline has changed state from PAUSED to PLAYING
# CallsSipMediaPipeline-DEBUG: Element rtp-send-pipeline has changed state from PAUSED to PLAYING
# DEBUG: Hanging up call
# CallsSipCall-DEBUG: Hanging up ongoing call
# CallsSipOrigin-DEBUG: incoming BYE: 200 Session Terminated
# CallsSipOrigin-DEBUG: The call state has changed: 200 Session Terminated
# CallsSipOrigin-DEBUG: Call terminated. Deactivating media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# CallsSipOrigin-DEBUG: response to BYE: 200 OK
# CallsSipOrigin-DEBUG: The call state has changed: 200 to BYE
# CallsSipOrigin-DEBUG: Call terminated. Deactivating media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# CallsSipMediaPipeline-DEBUG: Stopping media pipeline
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipOrigin-DEBUG: nua_shutdown () complete. Destroying nua handle
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipOrigin-DEBUG: nua_shutdown () complete. Destroying nua handle
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipOrigin-DEBUG: nua_shutdown () complete. Destroying nua handle
ok 5 /Calls/SIP/calls_direct_call
ok 6 /Calls/SIP/media_manager
# End of SIP tests
# End of Calls tests
--- stderr ---
su_source_port_create() returns 0x61400001aaa0
su_source_port_create() returns 0x61400001aca0
su_source_port_create() returns 0x61400001aea0
su_source_port_create() returns 0x6140000222a0
su_source_port_create() returns 0x6140000224a0
=================================================================
==1015926==ERROR: LeakSanitizer: detected memory leaks
Direct leak of 16384 byte(s) in 1 object(s) allocated from:
#0 0x7f0c6bd4ce8f in __interceptor_malloc ../../../../src/libsanitizer/asan/asan_malloc_linux.cpp:145
#1 0x7f0c6ba94d48 in g_malloc ../../../glib/gmem.c:106
SUMMARY: AddressSanitizer: 16384 byte(s) leaked in 1 allocation(s).
-------
11/12 account FAIL 0.32s (exit status 1)
--- command ---
07:13:09 G_TEST_BUILDDIR='/home/fortysixandtwo/git/calls/_build/tests' PYTHONDONTWRITEBYTECODE='yes' MALLOC_CHECK_='2' G_TEST_SRCDIR='/home/fortysixandtwo/git/calls/tests' NO_AT_BRIDGE='1' CALLS_AUDIOSINK='fakesink' CALLS_AUDIOSRC='audiotestsrc' GSETTINGS_BACKEND='memory' G_DEBUG='gc-friendly,fatal-warnings' /home/fortysixandtwo/git/calls/_build/tests/account
--- stdout ---
# random seed: R02S48720ac004ffaf9503885acb47b8effb
# GLib-GIO-DEBUG: _g_io_module_get_default: Found default implementation memory (GMemorySettingsBackend) for ?gsettings-backend?
# GLib-DEBUG: unsetenv() is not thread-safe and should not be used after threads are created
# GLib-GIO-DEBUG: _g_io_module_get_default: Found default implementation gvfs (GDaemonVfs) for ?gio-vfs?
1..1
# Start of Calls tests
# Start of Account tests
# CallsProvider-DEBUG: Loaded plugin `sip'
# CallsProvider-DEBUG: Created provider from plugin `sip'
# CallsAccountProvider-DEBUG: Trying to add account for Alice
# CallsSipOrigin-DEBUG: Updating credentials
# CallsSipOrigin-DEBUG: Protocol not set, falling back to 'UDP'
# CallsSipMediaManager-DEBUG: Creating CallsSipMediaManager
# CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMA is available
# CallsGstRfc3551-DEBUG: Adding PCMA to the codec candidates
# CallsGstRfc3551-DEBUG: Gstreamer plugin for PCMU is available
# CallsGstRfc3551-DEBUG: Adding PCMU to the codec candidates
# CallsGstRfc3551-DEBUG: Gstreamer plugin for GSM is available
# CallsGstRfc3551-DEBUG: Adding GSM to the codec candidates
# CallsGstRfc3551-DEBUG: Gstreamer plugin for G722 is available
# CallsGstRfc3551-DEBUG: Adding G722 to the codec candidates
# CallsGstRfc3551-DEBUG: Gstreamer plugin for G723 is available
# CallsGstRfc3551-DEBUG: Adding G723 to the codec candidates
# CallsAccountProvider-DEBUG: Trying to add account for Bob
# CallsSipOrigin-DEBUG: Updating credentials
# CallsSipOrigin-DEBUG: Protocol not set, falling back to 'UDP'
# CallsAccountProvider-DEBUG: Trying to get account from Alice
# CallsAccountProvider-DEBUG: Trying to get account from Alice
# CallsAccountProvider-DEBUG: Trying to get account from Bob
# CallsAccountProvider-DEBUG: Trying to get account from Bob
# CallsAccountProvider-DEBUG: Trying to get account from Alice
# CallsAccountProvider-DEBUG: Trying to get account from Bob
# CallsAccountProvider-DEBUG: Trying to add account for Alice
# CallsSipProvider-DEBUG: Cannot add credentials with name 'Alice' multiple times
# CallsAccountProvider-DEBUG: Trying to remove account from Alice
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipOrigin-DEBUG: nua_shutdown () complete. Destroying nua handle
# CallsAccountProvider-DEBUG: Trying to remove account from Alice
# CallsAccountProvider-DEBUG: Trying to remove account from Bob
# CallsSipOrigin-DEBUG: Requesting nua_shutdown ()
# CallsSipOrigin-DEBUG: response to nua_shutdown: 200 Shutdown successful
# CallsSipOrigin-DEBUG: nua_shutdown () complete. Destroying nua handle
# CallsAccountProvider-DEBUG: Trying to remove account from Bob
ok 1 /Calls/Account/basic
# End of Account tests
# End of Calls tests
--- stderr ---
su_source_port_create() returns 0x61400001aaa0
=================================================================
==1015928==ERROR: LeakSanitizer: detected memory leaks
Direct leak of 16384 byte(s) in 1 object(s) allocated from:
#0 0x7ff4c61c1e8f in __interceptor_malloc ../../../../src/libsanitizer/asan/asan_malloc_linux.cpp:145
#1 0x7ff4c5f09d48 in g_malloc ../../../glib/gmem.c:106
SUMMARY: AddressSanitizer: 16384 byte(s) leaked in 1 allocation(s).
-------
Summary of Failures:
6/12 origin FAIL 0.27s (exit status 1)
8/12 manager FAIL 0.32s (exit status 1)
10/12 sip FAIL 0.42s (exit status 1)
11/12 account FAIL 0.32s (exit status 1)
Ok: 8
Expected Fail: 0
Fail: 4
Unexpected Pass: 0
Skipped: 0
Timeout: 0
Full log written to /home/fortysixandtwo/git/calls/_build/meson-logs/testlog.txt
FAILED: meson-test
/usr/bin/meson test --no-rebuild --print-errorlogs
ninja: build stopped: subcommand failed.