[libtgvoip] Updated to 2.4.2 (regular release).

commit 27556f41d962dd2084a35f71d0aea88256828f31 Author: Vitaly Zaitsev <vitaly(a)easycoding.org&gt; Date: Sat Jan 12 14:20:15 2019 +0100 Updated to 2.4.2 (regular release). .gitignore | 1 + libtgvoip-build-fixes.patch | 192 +++++--------------------------------------- libtgvoip-sse2.patch | 43 ---------- libtgvoip.spec | 10 ++- sources | 2 +- 5 files changed, 31 insertions(+), 217 deletions(-) --- diff --git a/.gitignore b/.gitignore index 234dcb3..252aff7 100644 --- a/.gitignore +++ b/.gitignore @@ -8,3 +8,4 @@ /libtgvoip-2.2.3.tar.gz /libtgvoip-2.2.4.tar.gz /libtgvoip-2.4.tar.gz +/libtgvoip-2.4.2.tar.gz diff --git a/libtgvoip-build-fixes.patch b/libtgvoip-build-fixes.patch index 62e6c38..a09cbce 100644 --- a/libtgvoip-build-fixes.patch +++ b/libtgvoip-build-fixes.patch @@ -1,18 +1,14 @@ -From a68fc701c8110245f898100228fef2905ef6731e Mon Sep 17 00:00:00 2001 +From e94437a0f3444da9f42022090ba797725998928a Mon Sep 17 00:00:00 2001 From: Vitaly Zaitsev <vitaly(a)easycoding.org&gt; Date: Thu, 6 Jul 2017 17:27:01 +0200 Subject: [PATCH] Fixed build of libtgvoip under Fedora as shared library. --- - libtgvoip.gyp | 7 +++--- - webrtc_dsp/common_audio/wav_file.cc | 20 ++++++++++++---- - webrtc_dsp/common_audio/wav_header.cc | 34 ++++++++++++++++++++++++++- - webrtc_dsp/rtc_base/system/arch.h | 24 ++++++++++++++++++- - webrtc_dsp/typedefs.h | 24 ++++++++++++++++++- - 5 files changed, 98 insertions(+), 11 deletions(-) + libtgvoip.gyp | 13 +++++++------ + 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/libtgvoip.gyp b/libtgvoip.gyp -index 8c62b56..c362667 100644 +index 8c62b56..8fe9c5b 100644 --- a/libtgvoip.gyp +++ b/libtgvoip.gyp @@ -4,7 +4,8 @@ @@ -39,170 +35,26 @@ index 8c62b56..c362667 100644 ], 'direct_dependent_settings': { 'include_dirs': [ -diff --git a/webrtc_dsp/common_audio/wav_file.cc b/webrtc_dsp/common_audio/wav_file.cc -index 1b8bcbd..9ebf635 100644 ---- a/webrtc_dsp/common_audio/wav_file.cc -+++ b/webrtc_dsp/common_audio/wav_file.cc -@@ -93,9 +93,6 @@ size_t WavReader::num_samples() const { - } +@@ -70,8 +71,6 @@ + '<(tgvoip_src_loc)/video/VideoSource.h', + '<(tgvoip_src_loc)/video/VideoRenderer.cpp', + '<(tgvoip_src_loc)/video/VideoRenderer.h', +- '<(tgvoip_src_loc)/json11.cpp', +- '<(tgvoip_src_loc)/json11.hpp', - 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(num_samples, num_samples_remaining_); - const size_t read = -@@ -104,6 +101,12 @@ 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_ -= 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; - } + # Windows + '<(tgvoip_src_loc)/os/windows/NetworkSocketWinsock.cpp', +@@ -722,7 +721,9 @@ -@@ -178,10 +181,17 @@ size_t WavWriter::num_samples() const { - 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_ += written; - RTC_CHECK(num_samples_ >= written); // detect size_t overflow -diff --git a/webrtc_dsp/common_audio/wav_header.cc b/webrtc_dsp/common_audio/wav_header.cc -index e49b748..68a2620 100644 ---- a/webrtc_dsp/common_audio/wav_header.cc -+++ b/webrtc_dsp/common_audio/wav_header.cc -@@ -86,7 +86,39 @@ 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) & 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(size_t bytes_in_payload) { -diff --git a/webrtc_dsp/rtc_base/system/arch.h b/webrtc_dsp/rtc_base/system/arch.h -index 326b83e..ef27c17 100644 ---- a/webrtc_dsp/rtc_base/system/arch.h -+++ b/webrtc_dsp/rtc_base/system/arch.h -@@ -47,8 +47,30 @@ - #elif defined(__pnacl__) - #define WEBRTC_ARCH_32_BITS - #define WEBRTC_ARCH_LITTLE_ENDIAN -+#elif defined(__powerpc64__) -+#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__ -+#define WEBRTC_ARCH_LITTLE_ENDIAN -+#else -+#define WEBRTC_ARCH_BIG_ENDIAN -+#endif -+#define WEBRTC_ARCH_64_BITS -+#elif defined(__powerpc__) -+#define WEBRTC_ARCH_BIG_ENDIAN -+#define WEBRTC_ARCH_32_BITS -+#else -+/* 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 Please add support for your architecture in typedefs.h -+#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)) -diff --git a/webrtc_dsp/typedefs.h b/webrtc_dsp/typedefs.h -index c960d95..d6ce35e 100644 ---- a/webrtc_dsp/typedefs.h -+++ b/webrtc_dsp/typedefs.h -@@ -47,8 +47,30 @@ - #elif defined(__pnacl__) - #define WEBRTC_ARCH_32_BITS - #define WEBRTC_ARCH_LITTLE_ENDIAN -+#elif defined(__powerpc64__) -+#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__ -+#define WEBRTC_ARCH_LITTLE_ENDIAN -+#else -+#define WEBRTC_ARCH_BIG_ENDIAN -+#endif -+#define WEBRTC_ARCH_64_BITS -+#elif defined(__powerpc__) -+#define WEBRTC_ARCH_BIG_ENDIAN -+#define WEBRTC_ARCH_32_BITS -+#else -+/* 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 Please add support for your architecture in typedefs.h -+#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)) + ], +- 'libraries': [], ++ 'libraries': [ ++ 'json11', ++ ], + 'configurations': { + 'Debug': {}, + 'Release': {}, -- -2.19.2 +2.20.1 diff --git a/libtgvoip.spec b/libtgvoip.spec index d63a250..fec2c92 100644 --- a/libtgvoip.spec +++ b/libtgvoip.spec @@ -1,6 +1,6 @@ Name: libtgvoip -Version: 2.4 -Release: 2%{?dist} +Version: 2.4.2 +Release: 1%{?dist} Summary: VoIP library for Telegram clients # Libtgvoip shared library - Public Domain. @@ -10,13 +10,13 @@ URL: https://github.com/grishka/%{name} Source0: %{url}/archive/%{version}.tar.gz#/%{name}-%{version}.tar.gz Patch0: %{name}-build-fixes.patch -Patch100: %{name}-sse2.patch Provides: bundled(webrtc-audio-processing) = 0.3 BuildRequires: pulseaudio-libs-devel BuildRequires: alsa-lib-devel BuildRequires: openssl-devel +BuildRequires: json11-devel BuildRequires: opus-devel BuildRequires: gcc-c++ BuildRequires: cmake @@ -49,6 +49,7 @@ popd # Installing shared library... mkdir -p "%{buildroot}%{_libdir}" install -m 0755 -p out/Release/lib.target/%{name}.so.%{version} "%{buildroot}%{_libdir}/%{name}.so.%{version}" +ln -s %{name}.so.%{version} "%{buildroot}%{_libdir}/%{name}.so.2.4" ln -s %{name}.so.%{version} "%{buildroot}%{_libdir}/%{name}.so.2" ln -s %{name}.so.%{version} "%{buildroot}%{_libdir}/%{name}.so" @@ -69,6 +70,9 @@ find video -maxdepth 1 -type f -name "*.h" -exec install -m 0644 -p '{}' %{build %{_libdir}/%{name}.so %changelog +* Sat Jan 12 2019 Vitaly Zaitsev <vitaly(a)easycoding.org&gt; - 2.4.2-1 +- Updated to 2.4.2 (regular release). + * Tue Dec 11 2018 Vitaly Zaitsev <vitaly(a)easycoding.org&gt; - 2.4-2 - Backported upstream patch with crash and build fixes. diff --git a/sources b/sources index 2d4950e..9b36e00 100644 --- a/sources +++ b/sources @@ -1 +1 @@ -d0a9f1b6fa82e9681346525aa06f9b4e libtgvoip-2.4.tar.gz +34ef6e2ce1679615bf30048c7c323610 libtgvoip-2.4.2.tar.gz