remove unnecessary patch

webrtc-audio-processing-0.2-big-endian.patch

@@ -1,90 +0,0 @@
-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) {

webrtc-audio-processing.spec

@@ -1,14 +1,11 @@
 Name:		webrtc-audio-processing
 Version:	0.3.1
-Release:        2
+Release:        3
 Summary:	Real-Time Communication Library for Web Browsers
 License:	BSD and MIT
 URL:		https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
 Source0:	https://freedesktop.org/software/pulseaudio/webrtc-audio-processing/%{name}-%{version}.tar.xz
 
-Patch0:         webrtc-fix-typedefs-on-other-arches.patch
-Patch1:         webrtc-audio-processing-0.2-big-endian.patch
-
 BuildRequires:	autoconf automake libtool gcc gcc-c++
 
 %description
@@ -64,5 +61,8 @@ autoreconf -vif
 %doc NEWS
 
 %changelog
+* Fri Jan 10 2020 zhangrui <zhangrui182@huawei.com> - 0.3.1-3
+- Remove unnecessary patches
+
 * Fri Sep 6 2019 openEuler Buildteam <buildteam@openeuler.org> - 0.3.1-2
 - Package init

webrtc-fix-typedefs-on-other-arches.patch

@@ -1,24 +0,0 @@
-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))