commit webrtc-audio-processing for openSUSE:Factory

1 Jul 2016

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 - oholecek@suse.com
+
+- 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 - oholecek@suse.com
+
+- Remove unneeded explicit version dependency for automake
+
+-------------------------------------------------------------------
+Wed Jun 22 11:55:11 UTC 2016 - oholecek@suse.com
+
+- 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 - oholecek@suse.com
+
+- Add no-undefined.patch patch
+  https://cgit.freedesktop.org/pulseaudio/webrtc-audio-processing/patch/?id=d5...
+- 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 - oholecek@suse.com
+
+- Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build
+  https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-May/026294.ht...
+- 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 - oholecek@suse.com
+
+- 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 <pascal.bleser@opensuse.org>
 #
 # 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__

    

root＠hilbert.suse.de

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