This isn't complete still fails with:
http://errors.yoctoproject.org/Errors/Details/698131/
and after fixing this include (by dropping webrtc prefix), the other 2 includes don't exist anymore as well and fails with:
http://errors.yoctoproject.org/Errors/Details/698135/
FAILED: ext/webrtcdsp/libgstwebrtcdsp.so.p/gstwebrtcechoprobe.cpp.o
x86_64-oe-linux-g++ -m64 -march=core2 -mtune=core2 -msse3 -mfpmath=sse -fstack-protector-strong -O2 -D_FORTIFY_SOURCE=2 -Wformat -Wformat-security -Werror=format-security --sysroot=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot -Iext/webrtcdsp/libgstwebrtcdsp.so.p -Iext/webrtcdsp -I../gst-plugins-bad-1.22.0/ext/webrtcdsp -I. -I../gst-plugins-bad-1.22.0 -Igst-libs -I../gst-plugins-bad-1.22.0/gst-libs -Igst-libs/gst/audio -ITOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/gstreamer-1.0 -ITOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/glib-2.0 -ITOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/lib/glib-2.0/include -ITOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/orc-0.4 -ITOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/webrtc-audio-processing-1 -fdiagnostics-color=always -D_FILE_OFFSET_BITS=64 -Wall -Winvalid-pch -std=c++11 -Wno-non-virtual-dtor -fvisibility=hidden -fno-strict-aliasing -Wformat-nonliteral -Wmissing-declarations -Wredundant-decls -Wwrite-strings -Wformat -Wformat-security -Winit-self -Wmissing-include-dirs -Waddress -Wno-multichar -Wvla -Wpointer-arith -O2 -pipe -g -feliminate-unused-debug-types -fmacro-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/gst-plugins-bad-1.22.0=/usr/src/debug/gstreamer1.0-plugins-bad/1.22.0-r0 -fdebug-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/gst-plugins-bad-1.22.0=/usr/src/debug/gstreamer1.0-plugins-bad/1.22.0-r0 -fmacro-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/build=/usr/src/debug/gstreamer1.0-plugins-bad/1.22.0-r0 -fdebug-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/build=/usr/src/debug/gstreamer1.0-plugins-bad/1.22.0-r0 -fdebug-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot= -fmacro-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot= -fdebug-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot-native= -fvisibility-inlines-hidden -fPIC -DWEBRTC_LIBRARY_IMPL -DWEBRTC_POSIX -DNOMINMAX -pthread -DHAVE_CONFIG_H -MD -MQ ext/webrtcdsp/libgstwebrtcdsp.so.p/gstwebrtcechoprobe.cpp.o -MF ext/webrtcdsp/libgstwebrtcdsp.so.p/gstwebrtcechoprobe.cpp.o.d -o ext/webrtcdsp/libgstwebrtcdsp.so.p/gstwebrtcechoprobe.cpp.o -c ../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.cpp
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.cpp: In function 'gboolean gst_webrtc_echo_probe_setup(GstAudioFilter*, const GstAudioInfo*)':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.cpp:105:8: error: 'webrtc' has not been declared
105 | (webrtc::AudioFrame::kMaxDataSizeSamples * 2) < self->period_size)
| ^~~~~~
In file included from TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/gstreamer-1.0/gst/gst.h:55,
from ../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.h:26,
from ../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.cpp:34:
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.cpp:117:7: error: 'webrtc' has not been declared
117 | webrtc::AudioFrame::kMaxDataSizeSamples, self->period_size / 2);
| ^~~~~~
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/gstreamer-1.0/gst/gstinfo.h:727:31: note: in definition of macro 'GST_CAT_LEVEL_LOG'
727 | (GObject *) (object), __VA_ARGS__); \
| ^~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.cpp:114:3: note: in expansion of macro 'GST_WARNING_OBJECT'
114 | GST_WARNING_OBJECT (self, "webrtcdsp format produce too big period "
| ^~~~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.cpp: In function 'gint gst_webrtc_echo_probe_read(GstWebrtcEchoProbe*, GstClockTime, gpointer, GstBuffer**)':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.cpp:308:3: error: 'webrtc' has not been declared
308 | webrtc::AudioFrame * frame = (webrtc::AudioFrame *) _frame;
| ^~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.cpp:308:24: error: 'frame' was not declared in this scope; did you mean '_frame'?
308 | webrtc::AudioFrame * frame = (webrtc::AudioFrame *) _frame;
| ^~~~~
| _frame
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.cpp:308:33: error: 'webrtc' has not been declared
308 | webrtc::AudioFrame * frame = (webrtc::AudioFrame *) _frame;
| ^~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcechoprobe.cpp:308:53: error: expected primary-expression before ')' token
308 | webrtc::AudioFrame * frame = (webrtc::AudioFrame *) _frame;
| ^
FAILED: ext/webrtcdsp/libgstwebrtcdsp.so.p/gstwebrtcdsp.cpp.o
x86_64-oe-linux-g++ -m64 -march=core2 -mtune=core2 -msse3 -mfpmath=sse -fstack-protector-strong -O2 -D_FORTIFY_SOURCE=2 -Wformat -Wformat-security -Werror=format-security --sysroot=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot -Iext/webrtcdsp/libgstwebrtcdsp.so.p -Iext/webrtcdsp -I../gst-plugins-bad-1.22.0/ext/webrtcdsp -I. -I../gst-plugins-bad-1.22.0 -Igst-libs -I../gst-plugins-bad-1.22.0/gst-libs -Igst-libs/gst/audio -ITOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/gstreamer-1.0 -ITOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/glib-2.0 -ITOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/lib/glib-2.0/include -ITOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/orc-0.4 -ITOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/webrtc-audio-processing-1 -fdiagnostics-color=always -D_FILE_OFFSET_BITS=64 -Wall -Winvalid-pch -std=c++11 -Wno-non-virtual-dtor -fvisibility=hidden -fno-strict-aliasing -Wformat-nonliteral -Wmissing-declarations -Wredundant-decls -Wwrite-strings -Wformat -Wformat-security -Winit-self -Wmissing-include-dirs -Waddress -Wno-multichar -Wvla -Wpointer-arith -O2 -pipe -g -feliminate-unused-debug-types -fmacro-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/gst-plugins-bad-1.22.0=/usr/src/debug/gstreamer1.0-plugins-bad/1.22.0-r0 -fdebug-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/gst-plugins-bad-1.22.0=/usr/src/debug/gstreamer1.0-plugins-bad/1.22.0-r0 -fmacro-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/build=/usr/src/debug/gstreamer1.0-plugins-bad/1.22.0-r0 -fdebug-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/build=/usr/src/debug/gstreamer1.0-plugins-bad/1.22.0-r0 -fdebug-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot= -fmacro-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot= -fdebug-prefix-map=TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot-native= -fvisibility-inlines-hidden -fPIC -DWEBRTC_LIBRARY_IMPL -DWEBRTC_POSIX -DNOMINMAX -pthread -DHAVE_CONFIG_H -MD -MQ ext/webrtcdsp/libgstwebrtcdsp.so.p/gstwebrtcdsp.cpp.o -MF ext/webrtcdsp/libgstwebrtcdsp.so.p/gstwebrtcdsp.cpp.o.d -o ext/webrtcdsp/libgstwebrtcdsp.so.p/gstwebrtcdsp.cpp.o -c ../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp
In file included from TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/absl/base/config.h:86,
from TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/absl/types/optional.h:38,
from TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/webrtc-audio-processing-1/modules/audio_processing/include/audio_processing.h:26,
from ../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:74:
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/absl/base/policy_checks.h:79:2: error: #error "C++ versions less than C++14 are not supported."
79 | #error "C++ versions less than C++14 are not supported."
| ^~~~~
In file included from TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/webrtc-audio-processing-1/rtc_base/checks.h:58,
from TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/webrtc-audio-processing-1/api/array_view.h:18,
from TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/webrtc-audio-processing-1/modules/audio_processing/include/audio_processing.h:27:
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/absl/strings/string_view.h: In member function 'constexpr void absl::string_view::remove_prefix(size_type) const':
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/absl/strings/string_view.h:340:10: error: assignment of member 'absl::string_view::ptr_' in read-only object
340 | ptr_ += n;
| ~~~~~^~~~
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/absl/strings/string_view.h:341:13: error: assignment of member 'absl::string_view::length_' in read-only object
341 | length_ -= n;
| ~~~~~~~~^~~~
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/absl/strings/string_view.h: In member function 'constexpr void absl::string_view::remove_suffix(size_type) const':
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/absl/strings/string_view.h:350:13: error: assignment of member 'absl::string_view::length_' in read-only object
350 | length_ -= n;
| ~~~~~~~~^~~~
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/absl/strings/string_view.h: In member function 'constexpr void absl::string_view::swap(absl::string_view&) const':
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/absl/strings/string_view.h:358:13: error: passing 'const absl::string_view' as 'this' argument discards qualifiers [-fpermissive]
358 | *this = s;
| ^
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/absl/strings/string_view.h:161:7: note: in call to 'absl::string_view& absl::string_view::operator=(const absl::string_view&)'
161 | class string_view {
| ^~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: At global scope:
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:122:17: error: 'EchoCancellation' in namespace 'webrtc' does not name a type; did you mean 'EchoCanceller3Config'?
122 | typedef webrtc::EchoCancellation::SuppressionLevel GstWebrtcEchoSuppressionLevel;
| ^~~~~~~~~~~~~~~~
| EchoCanceller3Config
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: In function 'GType gst_webrtc_echo_suppression_level_get_type()':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:130:14: error: 'webrtc::EchoCancellation' has not been declared
130 | {webrtc::EchoCancellation::kLowSuppression, "Low Suppression", "low"},
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:131:14: error: 'webrtc::EchoCancellation' has not been declared
131 | {webrtc::EchoCancellation::kModerateSuppression,
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:133:14: error: 'webrtc::EchoCancellation' has not been declared
133 | {webrtc::EchoCancellation::kHighSuppression, "high Suppression", "high"},
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: At global scope:
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:144:17: error: 'NoiseSuppression' in namespace 'webrtc' does not name a type
144 | typedef webrtc::NoiseSuppression::Level GstWebrtcNoiseSuppressionLevel;
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: In function 'GType gst_webrtc_noise_suppression_level_get_type()':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:152:14: error: 'webrtc::NoiseSuppression' has not been declared
152 | {webrtc::NoiseSuppression::kLow, "Low Suppression", "low"},
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:153:14: error: 'webrtc::NoiseSuppression' has not been declared
153 | {webrtc::NoiseSuppression::kModerate, "Moderate Suppression", "moderate"},
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:154:14: error: 'webrtc::NoiseSuppression' has not been declared
154 | {webrtc::NoiseSuppression::kHigh, "High Suppression", "high"},
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:155:14: error: 'webrtc::NoiseSuppression' has not been declared
155 | {webrtc::NoiseSuppression::kVeryHigh, "Very High Suppression",
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: At global scope:
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:167:17: error: 'GainControl' in namespace 'webrtc' does not name a type
167 | typedef webrtc::GainControl::Mode GstWebrtcGainControlMode;
| ^~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: In function 'GType gst_webrtc_gain_control_mode_get_type()':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:175:14: error: 'webrtc::GainControl' has not been declared
175 | {webrtc::GainControl::kAdaptiveDigital, "Adaptive Digital", "adaptive-digital"},
| ^~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:176:14: error: 'webrtc::GainControl' has not been declared
176 | {webrtc::GainControl::kFixedDigital, "Fixed Digital", "fixed-digital"},
| ^~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: At global scope:
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:187:17: error: 'VoiceDetection' in namespace 'webrtc' does not name a type
187 | typedef webrtc::VoiceDetection::Likelihood GstWebrtcVoiceDetectionLikelihood;
| ^~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: In function 'GType gst_webrtc_voice_detection_likelihood_get_type()':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:195:14: error: 'webrtc::VoiceDetection' has not been declared
195 | {webrtc::VoiceDetection::kVeryLowLikelihood, "Very Low Likelihood", "very-low"},
| ^~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:196:14: error: 'webrtc::VoiceDetection' has not been declared
196 | {webrtc::VoiceDetection::kLowLikelihood, "Low Likelihood", "low"},
| ^~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:197:14: error: 'webrtc::VoiceDetection' has not been declared
197 | {webrtc::VoiceDetection::kModerateLikelihood, "Moderate Likelihood", "moderate"},
| ^~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:198:14: error: 'webrtc::VoiceDetection' has not been declared
198 | {webrtc::VoiceDetection::kHighLikelihood, "High Likelihood", "high"},
| ^~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: At global scope:
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:260:11: error: 'EchoCancellation' in namespace 'webrtc' does not name a type; did you mean 'EchoCanceller3Config'?
260 | webrtc::EchoCancellation::SuppressionLevel echo_suppression_level;
| ^~~~~~~~~~~~~~~~
| EchoCanceller3Config
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:262:11: error: 'NoiseSuppression' in namespace 'webrtc' does not name a type
262 | webrtc::NoiseSuppression::Level noise_suppression_level;
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:271:11: error: 'GainControl' in namespace 'webrtc' does not name a type
271 | webrtc::GainControl::Mode gain_control_mode;
| ^~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:274:11: error: 'VoiceDetection' in namespace 'webrtc' does not name a type
274 | webrtc::VoiceDetection::Likelihood voice_detection_likelihood;
| ^~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: In function 'GstFlowReturn gst_webrtc_dsp_analyze_reverse_stream(GstWebrtcDsp*, GstClockTime)':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:380:11: error: 'AudioFrame' is not a member of 'webrtc'
380 | webrtc::AudioFrame frame;
| ^~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:400:68: error: 'frame' was not declared in this scope
400 | delay = gst_webrtc_echo_probe_read (probe, rec_time, (gpointer) &frame, &buf);
| ^~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: In function 'void gst_webrtc_vad_post_activity(GstWebrtcDsp*, GstBuffer*, gboolean)':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:455:22: error: 'class webrtc::AudioProcessing' has no member named 'level_estimator'
455 | level = self->apm->level_estimator ()->RMS ();
| ^~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: In function 'GstFlowReturn gst_webrtc_dsp_process_stream(GstWebrtcDsp*, GstBuffer*)':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:493:13: error: 'AudioFrame' is not a member of 'webrtc'
493 | webrtc::AudioFrame frame;
| ^~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:494:5: error: 'frame' was not declared in this scope
494 | frame.num_channels_ = self->info.channels;
| ^~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:514:40: error: 'class webrtc::AudioProcessing' has no member named 'voice_detection'
514 | gboolean stream_has_voice = apm->voice_detection ()->stream_has_voice ();
| ^~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: In function 'gboolean gst_webrtc_dsp_start(GstBaseTransform*)':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:590:24: error: 'ExtendedFilter' is not a member of 'webrtc'
590 | config.Set < webrtc::ExtendedFilter >
| ^~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:590:24: error: 'ExtendedFilter' is not a member of 'webrtc'
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:591:12: error: expected type-specifier
591 | (new webrtc::ExtendedFilter (self->extended_filter));
| ^~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:594:24: error: 'DelayAgnostic' is not a member of 'webrtc'
594 | config.Set < webrtc::DelayAgnostic >
| ^~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:594:24: error: 'DelayAgnostic' is not a member of 'webrtc'
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:595:12: error: expected type-specifier
595 | (new webrtc::DelayAgnostic (self->delay_agnostic));
| ^~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:599:40: error: 'Create' is not a member of 'webrtc::AudioProcessing'
599 | self->apm = webrtc::AudioProcessing::Create (config);
| ^~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: In function 'gboolean gst_webrtc_dsp_setup(GstAudioFilter*, const GstAudioInfo*)':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:646:16: error: 'webrtc::AudioFrame' has not been declared
646 | (webrtc::AudioFrame::kMaxDataSizeSamples * 2) < self->period_size)
| ^~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:680:10: error: 'class webrtc::AudioProcessing' has no member named 'high_pass_filter'
680 | apm->high_pass_filter ()->Enable (true);
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:685:10: error: 'class webrtc::AudioProcessing' has no member named 'echo_cancellation'
685 | apm->echo_cancellation ()->enable_drift_compensation (false);
| ^~~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:686:10: error: 'class webrtc::AudioProcessing' has no member named 'echo_cancellation'
686 | apm->echo_cancellation ()
| ^~~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:687:40: error: 'GstWebrtcDsp' {aka 'struct _GstWebrtcDsp'} has no member named 'echo_suppression_level'
687 | ->set_suppression_level (self->echo_suppression_level);
| ^~~~~~~~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:688:10: error: 'class webrtc::AudioProcessing' has no member named 'echo_cancellation'
688 | apm->echo_cancellation ()->Enable (true);
| ^~~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:693:10: error: 'class webrtc::AudioProcessing' has no member named 'noise_suppression'
693 | apm->noise_suppression ()->set_level (self->noise_suppression_level);
| ^~~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:693:49: error: 'GstWebrtcDsp' {aka 'struct _GstWebrtcDsp'} has no member named 'noise_suppression_level'; did you mean 'noise_suppression'?
693 | apm->noise_suppression ()->set_level (self->noise_suppression_level);
| ^~~~~~~~~~~~~~~~~~~~~~~
| noise_suppression
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:694:10: error: 'class webrtc::AudioProcessing' has no member named 'noise_suppression'
694 | apm->noise_suppression ()->Enable (true);
| ^~~~~~~~~~~~~~~~~
In file included from TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/gstreamer-1.0/gst/gst.h:55,
from ../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.h:26,
from ../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:71:
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:705:45: error: 'GstWebrtcDsp' {aka 'struct _GstWebrtcDsp'} has no member named 'gain_control_mode'; did you mean 'gain_control'?
705 | g_enum_get_value (mode_class, self->gain_control_mode)->value_name);
| ^~~~~~~~~~~~~~~~~
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/gstreamer-1.0/gst/gstinfo.h:727:31: note: in definition of macro 'GST_CAT_LEVEL_LOG'
727 | (GObject *) (object), __VA_ARGS__); \
| ^~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:701:5: note: in expansion of macro 'GST_DEBUG_OBJECT'
701 | GST_DEBUG_OBJECT (self, "Enabling Digital Gain Control, target level "
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:709:10: error: 'class webrtc::AudioProcessing' has no member named 'gain_control'
709 | apm->gain_control ()->set_mode (self->gain_control_mode);
| ^~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:709:43: error: 'GstWebrtcDsp' {aka 'struct _GstWebrtcDsp'} has no member named 'gain_control_mode'; did you mean 'gain_control'?
709 | apm->gain_control ()->set_mode (self->gain_control_mode);
| ^~~~~~~~~~~~~~~~~
| gain_control
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:710:10: error: 'class webrtc::AudioProcessing' has no member named 'gain_control'
710 | apm->gain_control ()->set_target_level_dbfs (self->target_level_dbfs);
| ^~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:711:10: error: 'class webrtc::AudioProcessing' has no member named 'gain_control'
711 | apm->gain_control ()->set_compression_gain_db (self->compression_gain_db);
| ^~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:712:10: error: 'class webrtc::AudioProcessing' has no member named 'gain_control'
712 | apm->gain_control ()->enable_limiter (self->limiter);
| ^~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:713:10: error: 'class webrtc::AudioProcessing' has no member named 'gain_control'
713 | apm->gain_control ()->Enable (true);
| ^~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:722:17: error: 'GstWebrtcDsp' {aka 'struct _GstWebrtcDsp'} has no member named 'voice_detection_likelihood'
722 | self->voice_detection_likelihood)->value_name);
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/gstreamer-1.0/gst/gstinfo.h:727:31: note: in definition of macro 'GST_CAT_LEVEL_LOG'
727 | (GObject *) (object), __VA_ARGS__); \
| ^~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:719:5: note: in expansion of macro 'GST_DEBUG_OBJECT'
719 | GST_DEBUG_OBJECT (self, "Enabling Voice Activity Detection, frame size "
| ^~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:727:10: error: 'class webrtc::AudioProcessing' has no member named 'voice_detection'
727 | apm->voice_detection ()->Enable (true);
| ^~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:728:10: error: 'class webrtc::AudioProcessing' has no member named 'voice_detection'
728 | apm->voice_detection ()->set_likelihood (self->voice_detection_likelihood);
| ^~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:728:52: error: 'GstWebrtcDsp' {aka 'struct _GstWebrtcDsp'} has no member named 'voice_detection_likelihood'
728 | apm->voice_detection ()->set_likelihood (self->voice_detection_likelihood);
| ^~~~~~~~~~~~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:729:10: error: 'class webrtc::AudioProcessing' has no member named 'voice_detection'
729 | apm->voice_detection ()->set_frame_size_ms (
| ^~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:731:10: error: 'class webrtc::AudioProcessing' has no member named 'level_estimator'
731 | apm->level_estimator ()->Enable (true);
| ^~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:743:15: error: 'webrtc::AudioFrame' has not been declared
743 | webrtc::AudioFrame::kMaxDataSizeSamples, self->period_size / 2);
| ^~~~~~~~~~
TOPDIR/tmp-glibc/work/core2-64-oe-linux/gstreamer1.0-plugins-bad/1.22.0-r0/recipe-sysroot/usr/include/gstreamer-1.0/gst/gstinfo.h:727:31: note: in definition of macro 'GST_CAT_LEVEL_LOG'
727 | (GObject *) (object), __VA_ARGS__); \
| ^~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:740:3: note: in expansion of macro 'GST_WARNING_OBJECT'
740 | GST_WARNING_OBJECT (self, "webrtcdsp format produce too big period "
| ^~~~~~~~~~~~~~~~~~
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp: In function 'void gst_webrtc_dsp_set_property(GObject*, guint, const GValue*, GParamSpec*)':
../gst-plugins-bad-1.22.0/ext/webrtcdsp/gstwebrtcdsp.cpp:806:13: error: 'GstWebrtcDsp' {aka 'struct _GstWebrtcDsp'} has no member named 'echo_suppression_level'
806 | self->echo_suppression_level =
|
Signed-off-by: Martin Jansa <Martin.Jansa@gmail.com>
--- /dev/null
+From 7be7faee71a476c16c6a1456ea4f80ccfdeaad56 Mon Sep 17 00:00:00 2001
+From: Martin Jansa <Martin.Jansa@gmail.com>
+Date: Fri, 17 Mar 2023 13:28:30 +0100
+Subject: [PATCH] meson.build: accept webrtc-audio-processing-1
+
+Upstream-Status: Pending
+
+Signed-off-by: Martin Jansa <Martin.Jansa@gmail.com>
+---
+ ext/webrtcdsp/meson.build | 2 +-
+ 1 file changed, 1 insertion(+), 1 deletion(-)
+
+diff --git a/ext/webrtcdsp/meson.build b/ext/webrtcdsp/meson.build
+index 5aeae69..7bf2fc0 100644
+--- a/ext/webrtcdsp/meson.build
++++ b/ext/webrtcdsp/meson.build
+@@ -4,7 +4,7 @@ webrtc_sources = [
+ 'gstwebrtcdspplugin.cpp'
+ ]
+
+-webrtc_dep = dependency('webrtc-audio-processing', version : ['>= 0.2', '< 0.4'],
++webrtc_dep = dependency('webrtc-audio-processing-1',
+ required : get_option('webrtcdsp'))
+
+ if not gnustl_dep.found() and get_option('webrtcdsp').enabled()
file://0001-fix-maybe-uninitialized-warnings-when-compiling-with.patch \
file://0002-avoid-including-sys-poll.h-directly.patch \
file://0004-opencv-resolve-missing-opencv-data-dir-in-yocto-buil.patch \
+ file://0001-meson.build-accept-webrtc-audio-processing-1.patch \
"
SRC_URI[sha256sum] = "3c9d9300f5f4fb3e3d36009379d1fb6d9ecd79c1a135df742b8a68417dd663a1"
projects / thirdparty / openembedded / openembedded-core-contrib.git / commitdiff
