diff --git a/webrtc-audio-processing-0.2-big-endian.patch b/webrtc-audio-processing-0.2-big-endian.patch new file mode 100644 index 0000000..9361725 --- /dev/null +++ b/webrtc-audio-processing-0.2-big-endian.patch @@ -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(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(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(written); + RTC_CHECK(written <= std::numeric_limits::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(&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(a) << 24 ) ++ | (static_cast(b) << 16) ++ | (static_cast(c) << 8) ++ | (static_cast(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(&x), 4); ++} + #endif + + static inline uint32_t RiffChunkSize(uint32_t bytes_in_payload) { diff --git a/webrtc-audio-processing.spec b/webrtc-audio-processing.spec index 9e87123..b925e42 100644 --- a/webrtc-audio-processing.spec +++ b/webrtc-audio-processing.spec @@ -1,20 +1,17 @@ Name: webrtc-audio-processing Version: 0.2 -Release: 3%{?dist} +Release: 4%{?dist} Summary: Library for echo cancellation License: BSD URL: http://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/ Source0: http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/%{name}-%{version}.tar.xz -Patch0: webrtc-fix-typedefs-on-other-arches.patch +Patch0: webrtc-fix-typedefs-on-other-arches.patch Patch1: webrtc-audio-processing-0.2-no_undefined.patch Patch2: webrtc-audio-processing-0.2-x86_msse2.patch Patch3: webrtc-audio-processing-0.2-x86_sse2_runtime_detection.patch - -# could probably support more archs, provided they are little endian -# see also http://bugzilla.redhat.com/1336466 -ExclusiveArch: %{arm} %{ix86} x86_64 +Patch4: webrtc-audio-processing-0.2-big-endian.patch BuildRequires: automake libtool @@ -39,6 +36,7 @@ files for developing applications that use %{name}. %ifarch %{ix86} %patch3 -p1 -b .x86_sse2_runtime_detection %endif +%patch4 -p1 -b .bigendian # for patch1/patch2 autoreconf -fi @@ -83,6 +81,9 @@ rm -fv %{buildroot}%{_libdir}/lib*.la %changelog +* Tue May 24 2016 Than Ngo - 0.2-4 +- add support big endian + * Mon May 16 2016 Rex Dieter - 0.2-3 - ExclusiveArch primary archs, FTBFS on big endian arches (#1336466)