Hello community,
here is the log from the commit of package webrtc-audio-processing for
openSUSE:Factory checked in at 2016-07-01 09:55:14
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Comparing /work/SRC/openSUSE:Factory/webrtc-audio-processing (Old)
and /work/SRC/openSUSE:Factory/.webrtc-audio-processing.new (New)
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Package is "webrtc-audio-processing"
Changes:
--------
---
/work/SRC/openSUSE:Factory/webrtc-audio-processing/webrtc-audio-processing.changes
2013-03-08 11:20:50.000000000 +0100
+++
/work/SRC/openSUSE:Factory/.webrtc-audio-processing.new/webrtc-audio-processing.changes
2016-07-01 09:55:15.000000000 +0200
@@ -1,0 +2,77 @@
+Sat Jun 25 10:39:08 UTC 2016 - [email protected]
+
+- Remove webrtc-aarch64.patch, no longer needed
+- Adapt the rest of webrtc- patches to new arch naming
+
+-------------------------------------------------------------------
+Thu Jun 23 13:31:14 UTC 2016 - [email protected]
+
+- Remove unneeded explicit version dependency for automake
+
+-------------------------------------------------------------------
+Wed Jun 22 11:55:11 UTC 2016 - [email protected]
+
+- Update to 0.3
+ * build: enforce linking with --no-undefined, add explicit -lpthread
+ * build: Make sure files with SSE2 code are compiled with -msse2
+- Remove no-undefined.patch
+- Remove webrtc-audio-processing-0.2-x86_msse2.patch
+-------------------------------------------------------------------
+Mon Jun 20 13:02:06 UTC 2016 - [email protected]
+
+- Add no-undefined.patch patch
+
https://cgit.freedesktop.org/pulseaudio/webrtc-audio-processing/patch/?id=d58164e4d87854233564b59e76259b72e21507f6
+- Add big_endian_support_2.patch
https://bugs.freedesktop.org/show_bug.cgi?id=95738
+- Adapt webrtc-audio-processing-0.2-x86_msse2.patch to new version
+- Adapt big_endian_support.patch to new version
+
+-------------------------------------------------------------------
+Mon May 30 09:00:51 UTC 2016 - [email protected]
+
+- Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build
+
https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-May/026294.html
+- Add big_endian_support.patch
+ https://bugs.freedesktop.org/show_bug.cgi?id=95738
+- New automake version dependency >= 1.5
+
+-------------------------------------------------------------------
+Thu May 26 21:19:28 UTC 2016 - [email protected]
+
+- Update to 0.2:
+ Contains API breaking changes.
+
+ Upstream changes include:
+ * Rewritten AGC and voice activity detection
+ * Intelligibility enhancer
+ * Extended AEC filter
+ * Beamformer
+ * Transient suppressor
+ * ARM, NEON and MIPS optimisations (MIPS optimisations are not hooked up)
+
+ API changes:
+ * We no longer include a top-level audio_processing.h. The webrtc tree format
+ is used, so use webrtc/modules/audio_processing/include/audio_processing.h
+ * The top-level module_common_types.h has also been moved to
+ webrtc/modules/interface/module_common_types.h
+ * C++11 support is now required while compiling client code
+ * AudioProcessing::Create() does not take any arguments any more
+ * AudioProcessing::Destroy() is gone, use standard C++ "delete" instead
+ * Stream parameters are now configured via StreamConfig and ProcessingConfig
+ rather than set_sample_rate(), set_num_channels(), etc.
+ * AudioFrame field names have changed
+ * Use config API for newer audio processing options
+ * Use ProcessReverseStream() instead of AnalyzeReverseStream(), particularly
+ when using the intelligibility enhancer
+ * GainControl::set_analog_level_limits() is broken. The AGC implementation
+ hard codes 0-255 as the volume range
+
+ Other notes:
+ * The new audio processing parameters are not all tested, and a few are not
+ enabled upstream (in Chromium) either
+ * The rewritten AGC appears to be less sensitive, and it might make sense to
+ initialise the capture volume to something reasonable (33% or 50%, for
+ example) to make sure there is sufficient energy in the stream to trigger
+ the AGC mechanism
+- Adapted all 3 arch patches
+
+-------------------------------------------------------------------
Old:
----
webrtc-aarch64.patch
webrtc-audio-processing-0.1.tar.xz
New:
----
big_endian_support.patch
big_endian_support_2.patch
webrtc-audio-processing-0.3.tar.xz
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Other differences:
------------------
++++++ webrtc-audio-processing.spec ++++++
--- /var/tmp/diff_new_pack.Hwj7JM/_old 2016-07-01 09:55:16.000000000 +0200
+++ /var/tmp/diff_new_pack.Hwj7JM/_new 2016-07-01 09:55:16.000000000 +0200
@@ -2,7 +2,7 @@
#
# spec file for package webrtc-audio-processing
#
-# Copyright (c) 2013 SUSE LINUX Products GmbH, Nuernberg, Germany.
+# Copyright (c) 2016 SUSE LINUX GmbH, Nuernberg, Germany.
# Copyright (c) 2012 Pascal Bleser <[email protected]>
#
# All modifications and additions to the file contributed by third parties
@@ -18,18 +18,23 @@
#
+%define soname 1
# Please submit bugfixes or comments via http://bugs.opensuse.org/
-
Name: webrtc-audio-processing
-%define soname 0
-Version: 0.1
+Version: 0.3
Release: 0
Summary: Real-Time Communication Library for Web Browsers
License: BSD-3-Clause
Group: System/Libraries
-Source:
http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-%{version}.tar.xz
Url:
http://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
-BuildRoot: %{_tmppath}/%{name}-%{version}-build
+Source:
http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-%{version}.tar.xz
+# PATCH-FIX-UPSTREAN big_endian_support.patch
https://bugs.freedesktop.org/show_bug.cgi?id=95738
+Patch1: big_endian_support.patch
+# PATCH-FIX-UPSTREAN big_endian_support.patch
https://bugs.freedesktop.org/show_bug.cgi?id=95738
+Patch2: big_endian_support_2.patch
+# PATCH-FIX-OPENSUSE webrtc-(ppc64|s390x|aarch64).patch
+Patch100: webrtc-ppc64.patch
+Patch101: webrtc-s390x.patch
BuildRequires: autoconf
BuildRequires: automake
BuildRequires: gcc-c++
@@ -38,9 +43,7 @@
BuildRequires: make
BuildRequires: pkgconfig
BuildRequires: xz
-Patch0: webrtc-ppc64.patch
-Patch1: webrtc-s390x.patch
-Patch2: webrtc-aarch64.patch
+BuildRoot: %{_tmppath}/%{name}-%{version}-build
%description
WebRTC is an open source project that enables web browsers with Real-Time
@@ -86,31 +89,29 @@
%prep
%setup -q -T -c "%{name}-%{version}"
-xz --decompress --stdout "%{SOURCE0}" | %__tar xf - --strip-components=1
-%__sed -i 's/\r$//' AUTHORS
-%patch0 -p1
-%patch1
-%patch2
+xz --decompress --stdout "%{SOURCE0}" | tar xf - --strip-components=1
+sed -i 's/\r$//' AUTHORS
+%patch1 -p1
+%patch2 -p1
+%patch100
+%patch101
%build
%configure
-%__make %{?_smp_mflags} V=1
+make %{?_smp_mflags} V=1
%install
%makeinstall
-%__rm -f "%{buildroot}%{_libdir}"/*.la
+rm -f "%{buildroot}%{_libdir}"/*.la
%post -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig
%postun -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig
-%clean
-%{?buildroot:%__rm -rf "%{buildroot}"}
-
%files -n libwebrtc_audio_processing%{soname}
%defattr(-,root,root)
-%doc AUTHORS COPYING NEWS PATENTS README
+%doc AUTHORS COPYING NEWS README.md UPDATING.md
%{_libdir}/libwebrtc_audio_processing.so.%{soname}
%{_libdir}/libwebrtc_audio_processing.so.%{soname}.*.*
++++++ big_endian_support.patch ++++++
diff -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than
webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc
--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than
2016-05-24 08:28:45.749940095 -0400
+++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc 2016-05-24
08:50:30.361020010 -0400
@@ -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 +73,12 @@ size_t WavReader::ReadSamples(size_t num
RTC_CHECK(read == num_samples || feof(file_handle_));
RTC_CHECK_LE(read, num_samples_remaining_);
num_samples_remaining_ -= rtc::checked_cast<uint32_t>(read);
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+ //convert to big-endian
+ for(size_t idx = 0; idx < num_samples; idx++) {
+ samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
+ }
+#endif
return read;
}
@@ -120,10 +123,17 @@ 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
+ int16_t * le_samples = new int16_t[num_samples];
+ for(size_t idx = 0; idx < num_samples; idx++) {
+ le_samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
+ }
+ const size_t written =
+ fwrite(le_samples, sizeof(*le_samples), num_samples, file_handle_);
+ delete []le_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 -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than
webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc
--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than
2016-05-24 08:50:52.591379263 -0400
+++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc
2016-05-24 08:52:08.552606848 -0400
@@ -129,7 +129,39 @@ static inline std::string ReadFourCC(uin
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) & 0xff00) | ( ( x >> 8) & 0x00ff);
+}
+
+static inline void WriteLE32(uint32_t* f, uint32_t x) {
+ *f = ( (x & 0x000000ff) << 24 )
+ | ((x & 0x0000ff00) << 8)
+ | ((x & 0x00ff0000) >> 8)
+ | ((x & 0xff000000) >> 24 );
+}
+
+static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
+ *f = (static_cast<uint32_t>(a) << 24 )
+ | (static_cast<uint32_t>(b) << 16)
+ | (static_cast<uint32_t>(c) << 8)
+ | (static_cast<uint32_t>(d) );
+}
+
+static inline uint16_t ReadLE16(uint16_t x) {
+ return (( x & 0x00ff) << 8 )| ((x & 0xff00)>>8);
+}
+
+static inline uint32_t ReadLE32(uint32_t x) {
+ return ( (x & 0x000000ff) << 24 )
+ | ( (x & 0x0000ff00) << 8 )
+ | ( (x & 0x00ff0000) >> 8)
+ | ( (x & 0xff000000) >> 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) {
++++++ big_endian_support_2.patch ++++++
diff -up webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef
webrtc-audio-processing-0.2/webrtc/typedefs.h
--- webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef 2016-05-12
09:08:53.885000410 -0500
+++ webrtc-audio-processing-0.2/webrtc/typedefs.h 2016-05-12
09:12:38.006851953 -0500
@@ -48,7 +48,19 @@
#define WEBRTC_ARCH_32_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#else
-#error Please add support for your architecture in typedefs.h
+/* instead of failing, use typical unix defines... */
+#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
+#define WEBRTC_ARCH_BIG_ENDIAN
+#else
+#error __BYTE_ORDER__ is not defined
+#endif
+#if defined(__LP64__)
+#define WEBRTC_ARCH_64_BITS
+#else
+#define WEBRTC_ARCH_32_BITS
+#endif
#endif
#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
++++++ webrtc-audio-processing-0.1.tar.xz -> webrtc-audio-processing-0.3.tar.xz
++++++
++++ 162594 lines of diff (skipped)
++++++ webrtc-ppc64.patch ++++++
--- /var/tmp/diff_new_pack.Hwj7JM/_old 2016-07-01 09:55:17.000000000 +0200
+++ /var/tmp/diff_new_pack.Hwj7JM/_new 2016-07-01 09:55:17.000000000 +0200
@@ -1,17 +1,17 @@
-Index: webrtc-audio-processing-0.1/src/typedefs.h
+Index: webrtc/typedefs.h
===================================================================
---- webrtc-audio-processing-0.1.orig/src/typedefs.h
-+++ webrtc-audio-processing-0.1/src/typedefs.h
-@@ -76,6 +76,12 @@
- //#define WEBRTC_ARCH_ARMEL
+--- webrtc/typedefs.h.org
++++ webrtc/typedefs.h
+@@ -47,6 +47,12 @@
+ #elif defined(__pnacl__)
#define WEBRTC_ARCH_32_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__powerpc64__)
-+#define WEBRTC_BIG_ENDIAN
++#define WEBRTC_ARCH_BIG_ENDIAN
+#define WEBRTC_ARCH_64_BITS
+#elif defined(__powerpc__)
-+#define WEBRTC_BIG_ENDIAN
++#define WEBRTC_ARCH_BIG_ENDIAN
+#define WEBRTC_ARCH_32_BITS
#else
- #error Please add support for your architecture in typedefs.h
- #endif
+ /* instead of failing, use typical unix defines... */
+ #if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
++++++ webrtc-s390x.patch ++++++
--- /var/tmp/diff_new_pack.Hwj7JM/_old 2016-07-01 09:55:17.000000000 +0200
+++ /var/tmp/diff_new_pack.Hwj7JM/_new 2016-07-01 09:55:17.000000000 +0200
@@ -1,15 +1,15 @@
---- src/typedefs.h
-+++ src/typedefs.h
-@@ -82,6 +82,12 @@
+--- webrtc/typedefs.h
++++ webrtc/typedefs.h
+@@ -53,6 +53,12 @@
#elif defined(__powerpc__)
- #define WEBRTC_BIG_ENDIAN
+ #define WEBRTC_ARCH_BIG_ENDIAN
#define WEBRTC_ARCH_32_BITS
+#elif defined(__s390x__)
-+#define WEBRTC_BIG_ENDIAN
++#define WEBRTC_ARCH_BIG_ENDIAN
+#define WEBRTC_ARCH_64_BITS
+#elif defined(__s390__)
-+#define WEBRTC_BIG_ENDIAN
++#define WEBRTC_ARCH_BIG_ENDIAN
+#define WEBRTC_ARCH_32_BITS
#else
- #error Please add support for your architecture in typedefs.h
- #endif
+ /* instead of failing, use typical unix defines... */
+ #if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
