Complete Yocto mirror with license table for TQMa6UL (2038-compliance)
- 264 license table entries with exact download URLs (224/264 resolved) - Complete sources/ directory with all BitBake recipes - Build configuration: tqma6ul-multi-mba6ulx, spaetzle (musl) - Full traceability for Softwarefreigabeantrag - GCC 13.4.0, Linux 6.6.102, U-Boot 2023.04, musl 1.2.4 - License distribution: GPL-2.0 (24), MIT (23), GPL-2.0+ (18), BSD-3 (16)
This commit is contained in:
Siggi (OpenClaw Agent)
@@ -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() {
|
||||
@@ -0,0 +1,33 @@
|
||||
From 141fb5cbdaa6a883a61b5bf390c849159fc77605 Mon Sep 17 00:00:00 2001
|
||||
From: Markus Volk <f_l_k@t-online.de>
|
||||
Date: Thu, 14 Sep 2023 16:12:32 +0200
|
||||
Subject: [PATCH] file_wrapper.h: add missing include for musl
|
||||
|
||||
this fixes:
|
||||
| In file included from ../webrtc-audio-processing-1.3/webrtc/rtc_base/system/file_wrapper.cc:11:
|
||||
| ../webrtc-audio-processing-1.3/webrtc/rtc_base/system/file_wrapper.h:86:21: error: 'int64_t' has not been declared
|
||||
|
||||
if built with musl libc
|
||||
|
||||
Upstream-Status: Submitted [https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/37]
|
||||
|
||||
Signed-off-by: Markus Volk <f_l_k@t-online.de>
|
||||
---
|
||||
webrtc/rtc_base/system/file_wrapper.h | 1 +
|
||||
1 file changed, 1 insertion(+)
|
||||
|
||||
diff --git a/webrtc/rtc_base/system/file_wrapper.h b/webrtc/rtc_base/system/file_wrapper.h
|
||||
index 42c463c..c34d366 100644
|
||||
--- a/webrtc/rtc_base/system/file_wrapper.h
|
||||
+++ b/webrtc/rtc_base/system/file_wrapper.h
|
||||
@@ -13,6 +13,7 @@
|
||||
|
||||
#include <stddef.h>
|
||||
#include <stdio.h>
|
||||
+#include <cstdint>
|
||||
|
||||
#include <string>
|
||||
|
||||
--
|
||||
2.41.0
|
||||
|
||||
@@ -0,0 +1,21 @@
|
||||
DESCRIPTION = "Audio processing bits of the WebRTC reference implementation"
|
||||
HOMEPAGE = "https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/"
|
||||
SECTION = "audio"
|
||||
|
||||
DEPENDS = "abseil-cpp"
|
||||
DEPENDS:append:libc-musl = " libexecinfo"
|
||||
|
||||
LICENSE = "BSD-3-Clause"
|
||||
LIC_FILES_CHKSUM = "file://COPYING;md5=da08a38a32a340c5d91e13ee86a118f2"
|
||||
|
||||
SRC_URI = " \
|
||||
http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-${PV}.tar.xz \
|
||||
file://0001-add-missing-header-for-musl.patch \
|
||||
file://0001-Fix-return-type-errors.patch \
|
||||
"
|
||||
SRC_URI[sha256sum] = "2365e93e778d7b61b5d6e02d21c47d97222e9c7deff9e1d0838ad6ec2e86f1b9"
|
||||
S = "${WORKDIR}/webrtc-audio-processing-${PV}"
|
||||
|
||||
LDFLAGS:append:libc-musl = " -lexecinfo"
|
||||
|
||||
inherit meson pkgconfig
|
||||
@@ -0,0 +1,41 @@
|
||||
From 0c332b7d94f8425c4f33344ddf406b6eea458861 Mon Sep 17 00:00:00 2001
|
||||
From: Tanu Kaskinen <tanuk@iki.fi>
|
||||
Date: Mon, 8 May 2017 17:01:49 +0300
|
||||
Subject: [PATCH 4/6] typedefs.h: add support for 64-bit and big endian MIPS
|
||||
|
||||
The 64-bit check is taken from the upstream webrtc project, the big
|
||||
endian check is my own addition.
|
||||
|
||||
Upstream-Status: Pending
|
||||
Signed-off-by: Tanu Kaskinen <tanuk@iki.fi>
|
||||
---
|
||||
webrtc/typedefs.h | 11 ++++++++++-
|
||||
1 file changed, 10 insertions(+), 1 deletion(-)
|
||||
|
||||
diff --git a/webrtc/typedefs.h b/webrtc/typedefs.h
|
||||
index d875490..d1b2f54 100644
|
||||
--- a/webrtc/typedefs.h
|
||||
+++ b/webrtc/typedefs.h
|
||||
@@ -41,9 +41,18 @@
|
||||
//#define WEBRTC_ARCH_ARMEL
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
-#elif defined(__MIPSEL__)
|
||||
+#elif defined(__MIPSEL__) || defined(__MIPSEB__)
|
||||
+#define WEBRTC_ARCH_MIPS_FAMILY
|
||||
+#if defined(__LP64__)
|
||||
+#define WEBRTC_ARCH_64_BITS
|
||||
+#else
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
+#endif
|
||||
+#if defined(__MIPSEL__)
|
||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
+#else
|
||||
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||
+#endif
|
||||
#elif defined(__pnacl__)
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
--
|
||||
2.11.0
|
||||
|
||||
@@ -0,0 +1,28 @@
|
||||
From db5f570e928c8ca5b0b8dc702e1af0a57277f092 Mon Sep 17 00:00:00 2001
|
||||
From: Tanu Kaskinen <tanuk@iki.fi>
|
||||
Date: Mon, 3 Jul 2017 16:20:08 +0300
|
||||
Subject: [PATCH 5/6] typedefs.h: add support for PowerPC
|
||||
|
||||
Upstream-Status: Pending
|
||||
Signed-off-by: Tanu Kaskinen <tanuk@iki.fi>
|
||||
---
|
||||
webrtc/typedefs.h | 3 +++
|
||||
1 file changed, 3 insertions(+)
|
||||
|
||||
diff --git a/webrtc/typedefs.h b/webrtc/typedefs.h
|
||||
index d1b2f54..6e34d9e 100644
|
||||
--- a/webrtc/typedefs.h
|
||||
+++ b/webrtc/typedefs.h
|
||||
@@ -53,6 +53,9 @@
|
||||
#else
|
||||
#define WEBRTC_ARCH_BIG_ENDIAN
|
||||
#endif
|
||||
+#elif defined(__powerpc__)
|
||||
+#define WEBRTC_ARCH_32_BITS
|
||||
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||
#elif defined(__pnacl__)
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
--
|
||||
2.11.0
|
||||
|
||||
@@ -0,0 +1,116 @@
|
||||
From 7d31da8ef93987000f297d435dbacaf7d436107b Mon Sep 17 00:00:00 2001
|
||||
From: Tanu Kaskinen <tanuk@iki.fi>
|
||||
Date: Thu, 15 Jun 2017 18:38:30 +0300
|
||||
Subject: [PATCH 6/6] common_audio: implement endianness conversion in wav file
|
||||
handling
|
||||
|
||||
The code didn't build for big endian machines due to the missing
|
||||
endianness conversions.
|
||||
|
||||
Upstream-Status: Pending
|
||||
Signed-off-by: Tanu Kaskinen <tanuk@iki.fi>
|
||||
---
|
||||
webrtc/common_audio/wav_file.cc | 27 +++++++++++++++++++++------
|
||||
webrtc/common_audio/wav_header.cc | 32 +++++++++++++++++++++++++++++++-
|
||||
2 files changed, 52 insertions(+), 7 deletions(-)
|
||||
|
||||
diff --git a/webrtc/common_audio/wav_file.cc b/webrtc/common_audio/wav_file.cc
|
||||
index b14b620..e2f7738 100644
|
||||
--- a/webrtc/common_audio/wav_file.cc
|
||||
+++ b/webrtc/common_audio/wav_file.cc
|
||||
@@ -64,9 +64,6 @@ WavReader::~WavReader() {
|
||||
}
|
||||
|
||||
size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
|
||||
-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
-#error "Need to convert samples to big-endian when reading from WAV file"
|
||||
-#endif
|
||||
// There could be metadata after the audio; ensure we don't read it.
|
||||
num_samples = std::min(rtc::checked_cast<uint32_t>(num_samples),
|
||||
num_samples_remaining_);
|
||||
@@ -76,6 +73,13 @@ size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
|
||||
RTC_CHECK(read == num_samples || feof(file_handle_));
|
||||
RTC_CHECK_LE(read, num_samples_remaining_);
|
||||
num_samples_remaining_ -= rtc::checked_cast<uint32_t>(read);
|
||||
+
|
||||
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
|
||||
+ // Convert the read samples from little-endian to big-endian.
|
||||
+ for (size_t i = 0; i < read; i++)
|
||||
+ samples[i] = ((uint16_t) samples[i] >> 8) | ((uint16_t) samples[i] << 8);
|
||||
+#endif
|
||||
+
|
||||
return read;
|
||||
}
|
||||
|
||||
@@ -119,11 +123,22 @@ WavWriter::~WavWriter() {
|
||||
}
|
||||
|
||||
void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
|
||||
-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
-#error "Need to convert samples to little-endian when writing to WAV file"
|
||||
-#endif
|
||||
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
|
||||
+ // Convert the samples from big-endian samples to little-endian.
|
||||
+ int16_t* converted_samples = static_cast<int16_t*>(malloc(num_samples * sizeof(*samples)));
|
||||
+ RTC_CHECK(converted_samples) << "Out of memory.";
|
||||
+ for (int i = 0; i < num_samples; i++)
|
||||
+ converted_samples[i] =
|
||||
+ ((uint16_t) samples[i] >> 8) | ((uint16_t) samples[i] << 8);
|
||||
+
|
||||
+ const size_t written =
|
||||
+ fwrite(converted_samples, sizeof(*converted_samples), num_samples,
|
||||
+ file_handle_);
|
||||
+ free(converted_samples);
|
||||
+#else
|
||||
const size_t written =
|
||||
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
|
||||
+#endif
|
||||
RTC_CHECK_EQ(num_samples, written);
|
||||
num_samples_ += static_cast<uint32_t>(written);
|
||||
RTC_CHECK(written <= std::numeric_limits<uint32_t>::max() ||
|
||||
diff --git a/webrtc/common_audio/wav_header.cc b/webrtc/common_audio/wav_header.cc
|
||||
index 61cfffe..382bfc7 100644
|
||||
--- a/webrtc/common_audio/wav_header.cc
|
||||
+++ b/webrtc/common_audio/wav_header.cc
|
||||
@@ -129,7 +129,37 @@ static inline std::string ReadFourCC(uint32_t x) {
|
||||
return std::string(reinterpret_cast<char*>(&x), 4);
|
||||
}
|
||||
#else
|
||||
-#error "Write be-to-le conversion functions"
|
||||
+static inline void WriteLE16(uint16_t* f, uint16_t x) {
|
||||
+ *f = x >> 8 | x << 8;
|
||||
+}
|
||||
+static inline void WriteLE32(uint32_t* f, uint32_t x) {
|
||||
+ *f = x >> 24
|
||||
+ | (x && 0xFF0000) >> 8
|
||||
+ | (x && 0xFF00) << 8
|
||||
+ | x << 24;
|
||||
+}
|
||||
+static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
|
||||
+ *f = static_cast<uint32_t>(d)
|
||||
+ | static_cast<uint32_t>(c) << 8
|
||||
+ | static_cast<uint32_t>(b) << 16
|
||||
+ | static_cast<uint32_t>(a) << 24;
|
||||
+}
|
||||
+
|
||||
+static inline uint32_t ReadLE16(uint16_t x) {
|
||||
+ return x >> 8 | x << 8;
|
||||
+}
|
||||
+
|
||||
+static inline uint32_t ReadLE32(uint32_t x) {
|
||||
+ return x >> 24
|
||||
+ | (x && 0xFF0000) >> 8
|
||||
+ | (x && 0xFF00) << 8
|
||||
+ | x << 24;
|
||||
+}
|
||||
+
|
||||
+static inline std::string ReadFourCC(uint32_t x) {
|
||||
+ x = ReadLE32(x);
|
||||
+ return std::string(reinterpret_cast<char*>(&x), 4);
|
||||
+}
|
||||
#endif
|
||||
|
||||
static inline uint32_t RiffChunkSize(uint32_t bytes_in_payload) {
|
||||
--
|
||||
2.11.0
|
||||
|
||||
@@ -0,0 +1,33 @@
|
||||
Add support for RISC-V
|
||||
|
||||
Upstream-Status: Pending
|
||||
Signed-off-by: Khem Raj <raj.khem@gmail.com>
|
||||
--- a/webrtc/base/basictypes.h
|
||||
+++ b/webrtc/base/basictypes.h
|
||||
@@ -29,6 +29,10 @@
|
||||
#define CPU_ARM 1
|
||||
#endif
|
||||
|
||||
+#if defined(__riscv) || defined(_M_RISCV)
|
||||
+#define CPU_RISCV 1
|
||||
+#endif
|
||||
+
|
||||
#if defined(CPU_X86) && defined(CPU_ARM)
|
||||
#error CPU_X86 and CPU_ARM both defined.
|
||||
#endif
|
||||
--- a/webrtc/typedefs.h
|
||||
+++ b/webrtc/typedefs.h
|
||||
@@ -56,6 +56,13 @@
|
||||
#elif defined(__powerpc__)
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
#define WEBRTC_ARCH_BIG_ENDIAN
|
||||
+#elif defined(__riscv)
|
||||
+#if __riscv_xlen == 64
|
||||
+# define WEBRTC_ARCH_64_BITS
|
||||
+#else
|
||||
+# define WEBRTC_ARCH_32_BITS
|
||||
+#endif
|
||||
+#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
#elif defined(__pnacl__)
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
@@ -0,0 +1,23 @@
|
||||
DESCRIPTION = "Audio processing bits of the WebRTC reference implementation"
|
||||
HOMEPAGE = "https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/"
|
||||
SECTION = "audio"
|
||||
|
||||
DEPENDS:append:libc-musl = " libexecinfo"
|
||||
|
||||
LICENSE = "BSD-3-Clause"
|
||||
LIC_FILES_CHKSUM = "file://COPYING;md5=da08a38a32a340c5d91e13ee86a118f2 \
|
||||
file://webrtc/common.h;beginline=1;endline=9;md5=41f7322d91deabaf0acbbd0b8d0bc548 \
|
||||
"
|
||||
|
||||
SRC_URI = "http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/${BP}.tar.xz \
|
||||
file://0004-typedefs.h-add-support-for-64-bit-and-big-endian-MIP.patch \
|
||||
file://0005-typedefs.h-add-support-for-PowerPC.patch \
|
||||
file://0006-common_audio-implement-endianness-conversion-in-wav-.patch \
|
||||
file://riscv_support.patch \
|
||||
"
|
||||
|
||||
SRC_URI[md5sum] = "6e10724ca34bcbc715a4c208273acb0c"
|
||||
SRC_URI[sha256sum] = "a0fdd938fd85272d67e81572c5a4d9e200a0c104753cb3c209ded175ce3c5dbf"
|
||||
|
||||
LDFLAGS:append:libc-musl = " -lexecinfo"
|
||||
inherit autotools pkgconfig
|
||||
Reference in New Issue
Block a user