new file mode 100644
@@ -0,0 +1,95 @@
+From b6ad4b7086a6487b36d626248322f4c9d5bf420a Mon Sep 17 00:00:00 2001
+From: "thomas.georgec" <thomas.georgec@lge.com>
+Date: Sun, 12 Mar 2023 14:28:50 +0530
+Subject: [PATCH] Fix return-type errors
+
+Fix "control reaches end of non-void function" in code when -Werror=return-type
+is used.
+
+ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc: In function 'float webrtc::{anonymous}::GetLevel(const webrtc::VadLevelAnalyzer::Result&, LevelEstimatorType)':
+ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc:45:1: error: control reaches end of non-void function [-Werror=return-type]
+ 45 | }
+ | ^
+ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc: In function 'webrtc::GainControl::Mode webrtc::{anonymous}::Agc1ConfigModeToInterfaceMode(webrtc::AudioProcessing::Config::GainController1::Mode)':
+ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc:117:1: error: control reaches end of non-void function [-Werror=return-type]
+ 117 | }
+ | ^
+ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc: In lambda function:
+ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc:1853:13: error: control reaches end of non-void function -Werror=return-type]
+ 1853 | default:
+ | ^~~~~~~
+
+Signed-off-by: Martin Jansa <martin.jansa@gmail.com>
+---
+Upstream-Status: Submitted [https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/28]
+
+ .../audio_processing/agc2/adaptive_mode_level_estimator.cc | 2 ++
+ webrtc/modules/audio_processing/audio_processing_impl.cc | 3 +++
+ webrtc/modules/audio_processing/include/audio_processing.cc | 6 ++++++
+ 3 files changed, 11 insertions(+)
+
+diff --git a/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc b/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
+index f09f63b..9cdf6ca 100644
+--- a/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
++++ b/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
+@@ -42,6 +42,8 @@ float GetLevel(const VadLevelAnalyzer::Result& vad_level,
+ return vad_level.peak_dbfs;
+ break;
+ }
++ RTC_NOTREACHED();
++ __builtin_unreachable ();
+ }
+
+ } // namespace
+diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
+index 67208df..3b8262a 100644
+--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
++++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
+@@ -114,6 +114,8 @@ GainControl::Mode Agc1ConfigModeToInterfaceMode(
+ case Agc1Config::kFixedDigital:
+ return GainControl::kFixedDigital;
+ }
++ RTC_NOTREACHED();
++ __builtin_unreachable ();
+ }
+
+ // Maximum lengths that frame of samples being passed from the render side to
+@@ -1852,6 +1854,7 @@ void AudioProcessingImpl::InitializeNoiseSuppressor() {
+ return NsConfig::SuppressionLevel::k21dB;
+ default:
+ RTC_NOTREACHED();
++ __builtin_unreachable ();
+ }
+ };
+
+diff --git a/webrtc/modules/audio_processing/include/audio_processing.cc b/webrtc/modules/audio_processing/include/audio_processing.cc
+index 8854415..cc8752b 100644
+--- a/webrtc/modules/audio_processing/include/audio_processing.cc
++++ b/webrtc/modules/audio_processing/include/audio_processing.cc
+@@ -28,6 +28,8 @@ std::string NoiseSuppressionLevelToString(
+ case AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh:
+ return "VeryHigh";
+ }
++ RTC_NOTREACHED();
++ __builtin_unreachable ();
+ }
+
+ std::string GainController1ModeToString(
+@@ -40,6 +42,8 @@ std::string GainController1ModeToString(
+ case AudioProcessing::Config::GainController1::Mode::kFixedDigital:
+ return "FixedDigital";
+ }
++ RTC_NOTREACHED();
++ __builtin_unreachable ();
+ }
+
+ std::string GainController2LevelEstimatorToString(
+@@ -50,6 +54,8 @@ std::string GainController2LevelEstimatorToString(
+ case AudioProcessing::Config::GainController2::LevelEstimator::kPeak:
+ return "Peak";
+ }
++ RTC_NOTREACHED();
++ __builtin_unreachable ();
+ }
+
+ int GetDefaultMaxInternalRate() {
@@ -9,8 +9,9 @@ LICENSE = "BSD-3-Clause"
LIC_FILES_CHKSUM = "file://COPYING;md5=da08a38a32a340c5d91e13ee86a118f2"
SRC_URI = " \
- http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-${PV}.tar.xz \
- file://0001-add-missing-header-for-musl.patch \
+ http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-${PV}.tar.xz \
+ file://0001-add-missing-header-for-musl.patch \
+ file://0001-Fix-return-type-errors.patch \
"
SRC_URI[sha256sum] = "2365e93e778d7b61b5d6e02d21c47d97222e9c7deff9e1d0838ad6ec2e86f1b9"
S = "${WORKDIR}/webrtc-audio-processing-${PV}"
* apply the work around from: https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/28 which is still needed for 1.3 version: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc: In function 'float webrtc::{anonymous}::GetLevel(const webrtc::VadLevelAnalyzer::Result&, LevelEstimatorType)': webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc:45:1: error: control reaches end of non-void function [-Werror=return-type] 45 | } | ^ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc: In function 'webrtc::GainControl::Mode webrtc::{anonymous}::Agc1ConfigModeToInterfaceMode(webrtc::AudioProcessing::Config::GainController1::Mode)': webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc:117:1: error: control reaches end of non-void function [-Werror=return-type] 117 | } | ^ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc: In lambda function: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc:1853:13: error: control reaches end of non-void function -Werror=return-type] 1853 | default: | ^~~~~~~ * this can now be dropped from meta-webosose, where it was since: https://github.com/webosose/meta-webosose/commit/c2a24372e9f792bc900939a6642e64c459073b36#diff-c367f86896c48beb338dac912d3bf03e07d92c8345fa08dc3dcd93093a949c16 Signed-off-by: Martin Jansa <martin.jansa@gmail.com> --- .../0001-Fix-return-type-errors.patch | 95 +++++++++++++++++++ .../webrtc-audio-processing-1_1.3.bb | 5 +- 2 files changed, 98 insertions(+), 2 deletions(-) create mode 100644 meta-multimedia/recipes-multimedia/webrtc-audio-processing/webrtc-audio-processing-1/0001-Fix-return-type-errors.patch