commit e8905b1ce2a58ef91234a7bed0b197d2e65b9dc7 Author: Vitaly Zaitsev <vitaly(a)easycoding.org&gt; Date: Fri Aug 4 16:54:47 2017 +0500 Added some patches to fix build under other architectures. libtgvoip-build-fixes.patch | 150 ++++++++++++++++++++++++++++++++++++++++++-- libtgvoip.spec | 5 +- 2 files changed, 150 insertions(+), 5 deletions(-) --- diff --git a/libtgvoip-build-fixes.patch b/libtgvoip-build-fixes.patch index cbd631f..7ae4a62 100644 --- a/libtgvoip-build-fixes.patch +++ b/libtgvoip-build-fixes.patch @@ -1,14 +1,17 @@ -From b53352e8801e65c6a2e7ef1573d6f2bf0696c7cd Mon Sep 17 00:00:00 2001 +From a396724480f232267f2b100d35193204e0df3953 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 ++++--- - 1 file changed, 4 insertions(+), 3 deletions(-) + libtgvoip.gyp | 17 +++++++++----- + webrtc_dsp/webrtc/common_audio/wav_file.cc | 20 ++++++++++++---- + webrtc_dsp/webrtc/common_audio/wav_header.cc | 34 +++++++++++++++++++++++++++- + webrtc_dsp/webrtc/typedefs.h | 12 +++++++++- + 4 files changed, 70 insertions(+), 13 deletions(-) diff --git a/libtgvoip.gyp b/libtgvoip.gyp -index 527708d..c5d2cd5 100644 +index 527708d..f5db56a 100644 --- a/libtgvoip.gyp +++ b/libtgvoip.gyp @@ -4,7 +4,8 @@ @@ -35,6 +38,145 @@ index 527708d..c5d2cd5 100644 ], 'direct_dependent_settings': { 'include_dirs': [ +@@ -381,9 +382,6 @@ + 'defines': [ + 'WEBRTC_POSIX', + ], +- 'cflags_cc': [ +- '-msse2', +- ], + 'direct_dependent_settings': { + 'libraries': [ + +@@ -391,6 +389,13 @@ + }, + }, + ], ++ [ ++ '"<!(uname -p)" == "i686"', { ++ 'cflags_cc': [ ++ '-msse2', ++ ], ++ } ++ ], + ], + }, + ], +diff --git a/webrtc_dsp/webrtc/common_audio/wav_file.cc b/webrtc_dsp/webrtc/common_audio/wav_file.cc +index 2b9098a..4f99da7 100644 +--- a/webrtc_dsp/webrtc/common_audio/wav_file.cc ++++ b/webrtc_dsp/webrtc/common_audio/wav_file.cc +@@ -77,9 +77,6 @@ size_t WavReader::num_samples() const { + } + + 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 = +@@ -88,6 +85,12 @@ 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; + } + +@@ -144,10 +147,17 @@ 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/webrtc/common_audio/wav_header.cc b/webrtc_dsp/webrtc/common_audio/wav_header.cc +index 402ea17..b11ee58 100644 +--- a/webrtc_dsp/webrtc/common_audio/wav_header.cc ++++ b/webrtc_dsp/webrtc/common_audio/wav_header.cc +@@ -127,7 +127,39 @@ 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/webrtc/typedefs.h b/webrtc_dsp/webrtc/typedefs.h +index c960d95..69740b0 100644 +--- a/webrtc_dsp/webrtc/typedefs.h ++++ b/webrtc_dsp/webrtc/typedefs.h +@@ -48,7 +48,17 @@ + #define WEBRTC_ARCH_32_BITS + #define WEBRTC_ARCH_LITTLE_ENDIAN + #else +-#error Please add support for your architecture in typedefs.h ++#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)) -- 2.13.3 diff --git a/libtgvoip.spec b/libtgvoip.spec index ed38009..175bda8 100644 --- a/libtgvoip.spec +++ b/libtgvoip.spec @@ -5,7 +5,7 @@ Summary: VoIP library for Telegram clients Name: libtgvoip Version: 1.0 -Release: 2.%{date}git%{shortcommit0}%{?dist} +Release: 3.%{date}git%{shortcommit0}%{?dist} # Libtgvoip shared library - Public Domain. # Bundled webrtc library - BSD with patented echo cancellation algorithms. @@ -76,6 +76,9 @@ find audio -maxdepth 1 -type f -name "*.h" -exec install -m 0644 -p '{}' %{build %{_libdir}/%{name}.so %changelog +* Fri Aug 04 2017 Vitaly Zaitsev <vitaly(a)easycoding.org&gt; - 1.0-3.20170801gitbfd5cfe +- Fixed build other another architectures. + * Wed Aug 02 2017 Vitaly Zaitsev <vitaly(a)easycoding.org&gt; - 1.0-2.20170801gitbfd5cfe - Updated to latest snapshot. Small SPEC fixes. Added virtual provides.