WIP: gstreamer1.0-plugins-bad: accept webrtc-audio-processing-1jansa/webrtc

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>

2 files changed, 26 insertions, 0 deletions

diff --git a/meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-bad/0001-meson.build-accept-webrtc-audio-processing-1.patch b/meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-bad/0001-meson.build-accept-webrtc-audio-processing-1.patch
new file mode 100644
index 0000000000..4f79c9aaeb
--- /dev/null
+++ b/meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-bad/0001-meson.build-accept-webrtc-audio-processing-1.patch
@@ -0,0 +1,25 @@
+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()
diff --git a/meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-bad_1.22.0.bb b/meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-bad_1.22.0.bb
index b9fc17f3e9..a50f72b06f 100644
--- a/meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-bad_1.22.0.bb
+++ b/meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-bad_1.22.0.bb
@@ -9,6 +9,7 @@ SRC_URI = "https://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad
            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"