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Accepting request 404777 from home:oholecek:branches:multimedia:libs

Browse Source 
- Remove webrtc-aarch64.patch, no longer needed
- Adapt the rest of webrtc- patches to new arch naming 

- Remove unneeded explicit version dependency for automake

- 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

- 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

- 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

- 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

OBS-URL: https://build.opensuse.org/request/show/404777
OBS-URL: https://build.opensuse.org/package/show/multimedia:libs/webrtc-audio-processing?expand=0&rev=11
This commit is contained in:
Takashi Iwai 2016-06-25 16:50:49 +00:00 committed by Git OBS Bridge
parent 989d463d9f
commit d47a474aa6
9 changed files with 232 additions and 52 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

90
big_endian_support.patch Normal file
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@ -0,0 +1,90 @@
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) {

24
big_endian_support_2.patch Normal file
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@ -0,0 +1,24 @@
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))

12
webrtc-aarch64.patch
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@ -1,12 +0,0 @@
--- src/typedefs.h
+++ src/typedefs.h
@@ -82,6 +82,9 @@
#elif defined(__s390__)
#define WEBRTC_BIG_ENDIAN
#define WEBRTC_ARCH_32_BITS
+#elif defined(__aarch64__)
+#define WEBRTC_LITTLE_ENDIAN
+#define WEBRTC_ARCH_64_BITS
#else
#error Please add support for your architecture in typedefs.h
#endif

3
webrtc-audio-processing-0.1.tar.xz
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@ -1,3 +0,0 @@
version https://git-lfs.github.com/spec/v1
oid sha256:ed4b52f9c2688b97628035a5565377d74704d7c04de4254a768df3342c7afedc
size 392540

3
webrtc-audio-processing-0.3.tar.xz Normal file
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@ -0,0 +1,3 @@
version https://git-lfs.github.com/spec/v1
oid sha256:756e291d4f557d88cd50c4fe3b8454ec238362d22cedb3e6173240d90f0a80fa
size 688096

77
webrtc-audio-processing.changes
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@ -1,3 +1,80 @@
-------------------------------------------------------------------
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=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 - 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.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 - 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
-------------------------------------------------------------------
Thu Mar 7 13:51:31 UTC 2013 - idonmez@suse.com

41
webrtc-audio-processing.spec
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@ -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: libtool
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 @@ WebRTC implements the W3C's proposal for video conferencing on the web.
%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}.*.*

18
webrtc-ppc64.patch
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@ -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__

16
webrtc-s390x.patch
View File

@ -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__