Package init
This commit is contained in:
overweight
commit
ba3c548d39
12
audiofile-0.3.6-CVE-2015-7747.patch
Normal file
12
audiofile-0.3.6-CVE-2015-7747.patch
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@ -0,0 +1,12 @@
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diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/libaudiofile/modules/ModuleState.cpp audiofile-0.3.6/libaudiofile/modules/ModuleState.cpp
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--- audiofile-0.3.6-orig/libaudiofile/modules/ModuleState.cpp 2013-03-06 06:30:03.000000000 +0100
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+++ audiofile-0.3.6/libaudiofile/modules/ModuleState.cpp 2015-10-08 11:29:51.846082162 +0200
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@@ -402,7 +402,7 @@
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addModule(new Transform(outfc, in.pcm, out.pcm));
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if (in.channelCount != out.channelCount)
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- addModule(new ApplyChannelMatrix(infc, isReading,
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+ addModule(new ApplyChannelMatrix(outfc, isReading,
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in.channelCount, out.channelCount,
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in.pcm.minClip, in.pcm.maxClip,
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track->channelMatrix));
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48
audiofile-0.3.6-left-shift-neg.patch
Normal file
48
audiofile-0.3.6-left-shift-neg.patch
Normal file
@ -0,0 +1,48 @@
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diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/libaudiofile/modules/SimpleModule.h audiofile-0.3.6/libaudiofile/modules/SimpleModule.h
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--- audiofile-0.3.6-orig/libaudiofile/modules/SimpleModule.h 2013-03-06 06:30:03.000000000 +0100
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+++ audiofile-0.3.6/libaudiofile/modules/SimpleModule.h 2016-02-03 21:19:43.065454454 +0100
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@@ -123,7 +123,7 @@
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typedef typename IntTypes<Format>::UnsignedType UnsignedType;
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static const int kScaleBits = (Format + 1) * CHAR_BIT - 1;
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- static const int kMinSignedValue = -1 << kScaleBits;
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+ static const int kMinSignedValue = 0-(1U<<kScaleBits);
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struct signedToUnsigned : public std::unary_function<SignedType, UnsignedType>
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{
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diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/test/FloatToInt.cpp audiofile-0.3.6/test/FloatToInt.cpp
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--- audiofile-0.3.6-orig/test/FloatToInt.cpp 2013-02-11 18:23:26.000000000 +0100
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+++ audiofile-0.3.6/test/FloatToInt.cpp 2016-02-03 21:21:14.714510229 +0100
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@@ -115,7 +115,7 @@
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EXPECT_EQ(readData[i], expectedData[i]);
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}
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-static const int32_t kMinInt24 = -1<<23;
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+static const int32_t kMinInt24 = 0-(1U<<23);
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static const int32_t kMaxInt24 = (1<<23) - 1;
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TEST_F(FloatToIntTest, Int24)
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diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/test/IntToFloat.cpp audiofile-0.3.6/test/IntToFloat.cpp
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--- audiofile-0.3.6-orig/test/IntToFloat.cpp 2013-02-11 18:23:26.000000000 +0100
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+++ audiofile-0.3.6/test/IntToFloat.cpp 2016-02-03 21:20:57.380445355 +0100
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@@ -117,7 +117,7 @@
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EXPECT_EQ(readData[i], expectedData[i]);
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}
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-static const int32_t kMinInt24 = -1<<23;
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+static const int32_t kMinInt24 = 0-(1U<<23);
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static const int32_t kMaxInt24 = (1<<23) - 1;
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TEST_F(IntToFloatTest, Int24)
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diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/test/Sign.cpp audiofile-0.3.6/test/Sign.cpp
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--- audiofile-0.3.6-orig/test/Sign.cpp 2013-02-11 18:23:26.000000000 +0100
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+++ audiofile-0.3.6/test/Sign.cpp 2016-02-03 21:20:38.742450826 +0100
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@@ -116,7 +116,7 @@
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|
EXPECT_EQ(readData[i], expectedData[i]);
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}
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-static const int32_t kMinInt24 = -1<<23;
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+static const int32_t kMinInt24 = 0-(1U<<23);
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static const int32_t kMaxInt24 = (1<<23) - 1;
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static const uint32_t kMaxUInt24 = (1<<24) - 1;
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52
audiofile-0.3.6-narrowing.patch
Normal file
52
audiofile-0.3.6-narrowing.patch
Normal file
@ -0,0 +1,52 @@
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diff -Nur audiofile-0.3.6-orig/test/NeXT.cpp audiofile-0.3.6/test/NeXT.cpp
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--- audiofile-0.3.6-orig/test/NeXT.cpp 2013-02-11 18:23:26.000000000 +0100
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+++ audiofile-0.3.6/test/NeXT.cpp 2016-02-04 10:37:32.457140823 +0100
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@@ -37,13 +37,13 @@
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#include "TestUtilities.h"
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-const char kDataUnspecifiedLength[] =
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+const signed char kDataUnspecifiedLength[] =
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{
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'.', 's', 'n', 'd',
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0, 0, 0, 24, // offset of 24 bytes
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- 0xff, 0xff, 0xff, 0xff, // unspecified length
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+ -1, -1, -1, -1, // unspecified length
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0, 0, 0, 3, // 16-bit linear
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- 0, 0, 172, 68, // 44100 Hz
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+ 0, 0, -84, 68, // 44100 Hz (0xAC44)
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0, 0, 0, 1, // 1 channel
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0, 1,
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0, 1,
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@@ -57,13 +57,13 @@
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0, 55
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};
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-const char kDataTruncated[] =
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+const signed char kDataTruncated[] =
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{
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'.', 's', 'n', 'd',
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0, 0, 0, 24, // offset of 24 bytes
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0, 0, 0, 20, // length of 20 bytes
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0, 0, 0, 3, // 16-bit linear
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- 0, 0, 172, 68, // 44100 Hz
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+ 0, 0, -84, 68, // 44100 Hz (0xAC44)
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0, 0, 0, 1, // 1 channel
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0, 1,
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0, 1,
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@@ -152,13 +152,13 @@
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|
ASSERT_EQ(::unlink(testFileName.c_str()), 0);
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}
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-const char kDataZeroChannels[] =
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+const signed char kDataZeroChannels[] =
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{
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'.', 's', 'n', 'd',
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0, 0, 0, 24, // offset of 24 bytes
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0, 0, 0, 2, // 2 bytes
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0, 0, 0, 3, // 16-bit linear
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- 0, 0, 172, 68, // 44100 Hz
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+ 0, 0, -84, 68, // 44100 Hz (0xAC44)
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0, 0, 0, 0, // 0 channels
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0, 1
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};
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176
audiofile-0.3.6-pull42.patch
Normal file
176
audiofile-0.3.6-pull42.patch
Normal file
@ -0,0 +1,176 @@
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diff -Nur audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp audiofile-0.3.6-pull42/libaudiofile/modules/BlockCodec.cpp
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|
--- audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp 2013-03-06 06:30:03.000000000 +0100
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+++ audiofile-0.3.6-pull42/libaudiofile/modules/BlockCodec.cpp 2017-03-10 15:40:02.000000000 +0100
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@@ -52,8 +52,9 @@
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// Decompress into m_outChunk.
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|
for (int i=0; i<blocksRead; i++)
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|
{
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- decodeBlock(static_cast<const uint8_t *>(m_inChunk->buffer) + i * m_bytesPerPacket,
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- static_cast<int16_t *>(m_outChunk->buffer) + i * m_framesPerPacket * m_track->f.channelCount);
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+ if (decodeBlock(static_cast<const uint8_t *>(m_inChunk->buffer) + i * m_bytesPerPacket,
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+ static_cast<int16_t *>(m_outChunk->buffer) + i * m_framesPerPacket * m_track->f.channelCount)==0)
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+ break;
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framesRead += m_framesPerPacket;
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}
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diff -Nur audiofile-0.3.6/libaudiofile/modules/MSADPCM.cpp audiofile-0.3.6-pull42/libaudiofile/modules/MSADPCM.cpp
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|
--- audiofile-0.3.6/libaudiofile/modules/MSADPCM.cpp 2013-03-06 06:30:03.000000000 +0100
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+++ audiofile-0.3.6-pull42/libaudiofile/modules/MSADPCM.cpp 2017-03-10 15:40:02.000000000 +0100
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@@ -101,24 +101,60 @@
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768, 614, 512, 409, 307, 230, 230, 230
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|
};
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||||||
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+int firstBitSet(int x)
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+{
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|
+ int position=0;
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|
+ while (x!=0)
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|
+ {
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|
+ x>>=1;
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+ ++position;
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|
+ }
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|
+ return position;
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|
+}
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+
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|
+#ifndef __has_builtin
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|
+#define __has_builtin(x) 0
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|
+#endif
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|
+
|
||||||
|
+bool multiplyCheckOverflow(int a, int b, int *result)
|
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|
+{
|
||||||
|
+#if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow))
|
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|
+ return __builtin_mul_overflow(a, b, result);
|
||||||
|
+#else
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|
+ if (firstBitSet(a)+firstBitSet(b)>31) // int is signed, so we can't use 32 bits
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|
+ return true;
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|
+ *result = a * b;
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||||||
|
+ return false;
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||||||
|
+#endif
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|
+}
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|
+
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|
+
|
||||||
|
// Compute a linear PCM value from the given differential coded value.
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|
static int16_t decodeSample(ms_adpcm_state &state,
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|
- uint8_t code, const int16_t *coefficient)
|
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|
+ uint8_t code, const int16_t *coefficient, bool *ok=NULL)
|
||||||
|
{
|
||||||
|
int linearSample = (state.sample1 * coefficient[0] +
|
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|
state.sample2 * coefficient[1]) >> 8;
|
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|
+ int delta;
|
||||||
|
|
||||||
|
linearSample += ((code & 0x08) ? (code - 0x10) : code) * state.delta;
|
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|
|
||||||
|
linearSample = clamp(linearSample, MIN_INT16, MAX_INT16);
|
||||||
|
|
||||||
|
- int delta = (state.delta * adaptationTable[code]) >> 8;
|
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|
+ if (multiplyCheckOverflow(state.delta, adaptationTable[code], &delta))
|
||||||
|
+ {
|
||||||
|
+ if (ok) *ok=false;
|
||||||
|
+ _af_error(AF_BAD_COMPRESSION, "Error decoding sample");
|
||||||
|
+ return 0;
|
||||||
|
+ }
|
||||||
|
+ delta >>= 8;
|
||||||
|
if (delta < 16)
|
||||||
|
delta = 16;
|
||||||
|
|
||||||
|
state.delta = delta;
|
||||||
|
state.sample2 = state.sample1;
|
||||||
|
state.sample1 = linearSample;
|
||||||
|
+ if (ok) *ok=true;
|
||||||
|
|
||||||
|
return static_cast<int16_t>(linearSample);
|
||||||
|
}
|
||||||
|
@@ -212,13 +248,16 @@
|
||||||
|
{
|
||||||
|
uint8_t code;
|
||||||
|
int16_t newSample;
|
||||||
|
+ bool ok;
|
||||||
|
|
||||||
|
code = *encoded >> 4;
|
||||||
|
- newSample = decodeSample(*state[0], code, coefficient[0]);
|
||||||
|
+ newSample = decodeSample(*state[0], code, coefficient[0], &ok);
|
||||||
|
+ if (!ok) return 0;
|
||||||
|
*decoded++ = newSample;
|
||||||
|
|
||||||
|
code = *encoded & 0x0f;
|
||||||
|
- newSample = decodeSample(*state[1], code, coefficient[1]);
|
||||||
|
+ newSample = decodeSample(*state[1], code, coefficient[1], &ok);
|
||||||
|
+ if (!ok) return 0;
|
||||||
|
*decoded++ = newSample;
|
||||||
|
|
||||||
|
encoded++;
|
||||||
|
diff -Nur audiofile-0.3.6/libaudiofile/WAVE.cpp audiofile-0.3.6-pull42/libaudiofile/WAVE.cpp
|
||||||
|
--- audiofile-0.3.6/libaudiofile/WAVE.cpp 2013-03-06 06:30:03.000000000 +0100
|
||||||
|
+++ audiofile-0.3.6-pull42/libaudiofile/WAVE.cpp 2017-03-10 15:40:02.000000000 +0100
|
||||||
|
@@ -281,6 +281,12 @@
|
||||||
|
|
||||||
|
/* numCoefficients should be at least 7. */
|
||||||
|
assert(numCoefficients >= 7 && numCoefficients <= 255);
|
||||||
|
+ if (numCoefficients < 7 || numCoefficients > 255)
|
||||||
|
+ {
|
||||||
|
+ _af_error(AF_BAD_HEADER,
|
||||||
|
+ "Bad number of coefficients");
|
||||||
|
+ return AF_FAIL;
|
||||||
|
+ }
|
||||||
|
|
||||||
|
m_msadpcmNumCoefficients = numCoefficients;
|
||||||
|
|
||||||
|
@@ -834,6 +840,8 @@
|
||||||
|
}
|
||||||
|
|
||||||
|
TrackSetup *track = setup->getTrack();
|
||||||
|
+ if (!track)
|
||||||
|
+ return AF_NULL_FILESETUP;
|
||||||
|
|
||||||
|
if (track->f.isCompressed())
|
||||||
|
{
|
||||||
|
diff -Nur audiofile-0.3.6/sfcommands/sfconvert.c audiofile-0.3.6-pull42/sfcommands/sfconvert.c
|
||||||
|
--- audiofile-0.3.6/sfcommands/sfconvert.c 2013-03-06 06:30:03.000000000 +0100
|
||||||
|
+++ audiofile-0.3.6-pull42/sfcommands/sfconvert.c 2017-03-10 15:40:02.000000000 +0100
|
||||||
|
@@ -45,6 +45,33 @@
|
||||||
|
void usageerror (void);
|
||||||
|
bool copyaudiodata (AFfilehandle infile, AFfilehandle outfile, int trackid);
|
||||||
|
|
||||||
|
+int firstBitSet(int x)
|
||||||
|
+{
|
||||||
|
+ int position=0;
|
||||||
|
+ while (x!=0)
|
||||||
|
+ {
|
||||||
|
+ x>>=1;
|
||||||
|
+ ++position;
|
||||||
|
+ }
|
||||||
|
+ return position;
|
||||||
|
+}
|
||||||
|
+
|
||||||
|
+#ifndef __has_builtin
|
||||||
|
+#define __has_builtin(x) 0
|
||||||
|
+#endif
|
||||||
|
+
|
||||||
|
+bool multiplyCheckOverflow(int a, int b, int *result)
|
||||||
|
+{
|
||||||
|
+#if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow))
|
||||||
|
+ return __builtin_mul_overflow(a, b, result);
|
||||||
|
+#else
|
||||||
|
+ if (firstBitSet(a)+firstBitSet(b)>31) // int is signed, so we can't use 32 bits
|
||||||
|
+ return true;
|
||||||
|
+ *result = a * b;
|
||||||
|
+ return false;
|
||||||
|
+#endif
|
||||||
|
+}
|
||||||
|
+
|
||||||
|
int main (int argc, char **argv)
|
||||||
|
{
|
||||||
|
if (argc == 2)
|
||||||
|
@@ -323,8 +350,11 @@
|
||||||
|
{
|
||||||
|
int frameSize = afGetVirtualFrameSize(infile, trackid, 1);
|
||||||
|
|
||||||
|
- const int kBufferFrameCount = 65536;
|
||||||
|
- void *buffer = malloc(kBufferFrameCount * frameSize);
|
||||||
|
+ int kBufferFrameCount = 65536;
|
||||||
|
+ int bufferSize;
|
||||||
|
+ while (multiplyCheckOverflow(kBufferFrameCount, frameSize, &bufferSize))
|
||||||
|
+ kBufferFrameCount /= 2;
|
||||||
|
+ void *buffer = malloc(bufferSize);
|
||||||
|
|
||||||
|
AFframecount totalFrames = afGetFrameCount(infile, AF_DEFAULT_TRACK);
|
||||||
|
AFframecount totalFramesWritten = 0;
|
||||||
21
audiofile-0.3.6-pull43.patch
Normal file
21
audiofile-0.3.6-pull43.patch
Normal file
@ -0,0 +1,21 @@
|
|||||||
|
diff -Nur audiofile-0.3.6/libaudiofile/modules/IMA.cpp audiofile-0.3.6-pull43/libaudiofile/modules/IMA.cpp
|
||||||
|
--- audiofile-0.3.6/libaudiofile/modules/IMA.cpp 2013-03-06 06:30:03.000000000 +0100
|
||||||
|
+++ audiofile-0.3.6-pull43/libaudiofile/modules/IMA.cpp 2017-03-06 18:06:35.000000000 +0100
|
||||||
|
@@ -169,7 +169,7 @@
|
||||||
|
if (encoded[1] & 0x80)
|
||||||
|
m_adpcmState[c].previousValue -= 0x10000;
|
||||||
|
|
||||||
|
- m_adpcmState[c].index = encoded[2];
|
||||||
|
+ m_adpcmState[c].index = clamp(encoded[2], 0, 88);
|
||||||
|
|
||||||
|
*decoded++ = m_adpcmState[c].previousValue;
|
||||||
|
|
||||||
|
@@ -210,7 +210,7 @@
|
||||||
|
predictor -= 0x10000;
|
||||||
|
|
||||||
|
state.previousValue = clamp(predictor, MIN_INT16, MAX_INT16);
|
||||||
|
- state.index = encoded[1] & 0x7f;
|
||||||
|
+ state.index = clamp(encoded[1] & 0x7f, 0, 88);
|
||||||
|
encoded += 2;
|
||||||
|
|
||||||
|
for (int n=0; n<m_framesPerPacket; n+=2)
|
||||||
31
audiofile-0.3.6-pull44.patch
Normal file
31
audiofile-0.3.6-pull44.patch
Normal file
@ -0,0 +1,31 @@
|
|||||||
|
diff -Nur audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp audiofile-0.3.6-pull44/libaudiofile/modules/BlockCodec.cpp
|
||||||
|
--- audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp 2013-03-06 06:30:03.000000000 +0100
|
||||||
|
+++ audiofile-0.3.6-pull44/libaudiofile/modules/BlockCodec.cpp 2017-03-09 10:21:18.000000000 +0100
|
||||||
|
@@ -47,7 +47,7 @@
|
||||||
|
|
||||||
|
// Read the compressed data.
|
||||||
|
ssize_t bytesRead = read(m_inChunk->buffer, m_bytesPerPacket * blockCount);
|
||||||
|
- int blocksRead = bytesRead >= 0 ? bytesRead / m_bytesPerPacket : 0;
|
||||||
|
+ int blocksRead = (bytesRead >= 0 && m_bytesPerPacket > 0) ? bytesRead / m_bytesPerPacket : 0;
|
||||||
|
|
||||||
|
// Decompress into m_outChunk.
|
||||||
|
for (int i=0; i<blocksRead; i++)
|
||||||
|
diff -Nur audiofile-0.3.6/libaudiofile/WAVE.cpp audiofile-0.3.6-pull44/libaudiofile/WAVE.cpp
|
||||||
|
--- audiofile-0.3.6/libaudiofile/WAVE.cpp 2013-03-06 06:30:03.000000000 +0100
|
||||||
|
+++ audiofile-0.3.6-pull44/libaudiofile/WAVE.cpp 2017-03-09 10:21:18.000000000 +0100
|
||||||
|
@@ -326,6 +326,7 @@
|
||||||
|
{
|
||||||
|
_af_error(AF_BAD_NOT_IMPLEMENTED,
|
||||||
|
"IMA ADPCM compression supports only 4 bits per sample");
|
||||||
|
+ return AF_FAIL;
|
||||||
|
}
|
||||||
|
|
||||||
|
int bytesPerBlock = (samplesPerBlock + 14) / 8 * 4 * channelCount;
|
||||||
|
@@ -333,6 +334,7 @@
|
||||||
|
{
|
||||||
|
_af_error(AF_BAD_CODEC_CONFIG,
|
||||||
|
"Invalid samples per block for IMA ADPCM compression");
|
||||||
|
+ return AF_FAIL;
|
||||||
|
}
|
||||||
|
|
||||||
|
track->f.sampleWidth = 16;
|
||||||
BIN
audiofile-0.3.6.tar.gz
Normal file
BIN
audiofile-0.3.6.tar.gz
Normal file
Binary file not shown.
95
audiofile.spec
Normal file
95
audiofile.spec
Normal file
@ -0,0 +1,95 @@
|
|||||||
|
Name: audiofile
|
||||||
|
Version: 0.3.6
|
||||||
|
Release: 22
|
||||||
|
Summary: Library for reading and writing audio files in many common formats
|
||||||
|
License: LGPLv2+ and GPLv2+
|
||||||
|
URL: http://audiofile.68k.org/
|
||||||
|
Source0: http://audiofile.68k.org/%{name}-%{version}.tar.gz
|
||||||
|
Patch0: audiofile-0.3.6-CVE-2015-7747.patch
|
||||||
|
Patch1: audiofile-0.3.6-left-shift-neg.patch
|
||||||
|
Patch2: audiofile-0.3.6-narrowing.patch
|
||||||
|
Patch3: audiofile-0.3.6-pull42.patch
|
||||||
|
Patch4: audiofile-0.3.6-pull43.patch
|
||||||
|
Patch5: audiofile-0.3.6-pull44.patch
|
||||||
|
|
||||||
|
BuildRequires: gcc-c++ libtool alsa-lib-devel flac-devel
|
||||||
|
|
||||||
|
%description
|
||||||
|
The Audio File Library is a C-based library for reading and writing audio files in many
|
||||||
|
common formats.
|
||||||
|
|
||||||
|
The Audio File Library provides a uniform API which abstracts away details of file formats
|
||||||
|
and data formats. The same calls for opening a file, accessing and manipulating audio
|
||||||
|
metadata (e.g. sample rate, sample format, textual information, MIDI parameters), and
|
||||||
|
reading and writing sample data will work with any supported audio file format.
|
||||||
|
|
||||||
|
%package devel
|
||||||
|
Summary: Development files for Audio File applications
|
||||||
|
Requires: %{name}%{?_isa} = %{version}-%{release}
|
||||||
|
Provides: audiofile-static
|
||||||
|
Obsoletes: audiofile-static
|
||||||
|
|
||||||
|
%description devel
|
||||||
|
The audiofile-devel package contains libraries, include files, and
|
||||||
|
other resources you can use to develop Audio File applications.
|
||||||
|
|
||||||
|
%package_help
|
||||||
|
|
||||||
|
%prep
|
||||||
|
%autosetup -n %{name}-%{version} -p1
|
||||||
|
|
||||||
|
%build
|
||||||
|
%configure
|
||||||
|
%make_build
|
||||||
|
|
||||||
|
%install
|
||||||
|
rm -rf $RPM_BUILD_ROOT
|
||||||
|
%make_install
|
||||||
|
rm -f $RPM_BUILD_ROOT%{_libdir}/libaudiofile.la
|
||||||
|
|
||||||
|
%check
|
||||||
|
make check
|
||||||
|
|
||||||
|
%pre
|
||||||
|
|
||||||
|
%preun
|
||||||
|
|
||||||
|
%post
|
||||||
|
|
||||||
|
%postun
|
||||||
|
|
||||||
|
%ldconfig_scriptlets
|
||||||
|
|
||||||
|
%files
|
||||||
|
%doc ACKNOWLEDGEMENTS AUTHORS NEWS NOTES README TODO COPYING*
|
||||||
|
%{_bindir}/sfconvert
|
||||||
|
%{_bindir}/sfinfo
|
||||||
|
%{_libdir}/lib*.so.1*
|
||||||
|
|
||||||
|
%files devel
|
||||||
|
%doc ChangeLog docs/*.3.txt
|
||||||
|
%{_libdir}/libaudiofile.so
|
||||||
|
%{_libdir}/pkgconfig/audiofile.pc
|
||||||
|
%{_includedir}/*
|
||||||
|
%{_libdir}/libaudiofile.a
|
||||||
|
|
||||||
|
%files help
|
||||||
|
%{_mandir}/man1/*
|
||||||
|
%{_mandir}/man3/*
|
||||||
|
|
||||||
|
%changelog
|
||||||
|
* Fri Sep 27 2019 chengquan<chengquan3@huawei.com> - 0.3.6-22
|
||||||
|
- Type:bugfix
|
||||||
|
- ID:NA
|
||||||
|
- SUG:NA
|
||||||
|
- DESC:add help package and merge devel package
|
||||||
|
|
||||||
|
* Tue Sep 17 2019 shenyangyang<shenyangyang4@huawei.com> - 0.3.6-21
|
||||||
|
- Type:bugfix
|
||||||
|
- ID:NA
|
||||||
|
- SUG:NA
|
||||||
|
- DESC:and % before isa of devel
|
||||||
|
|
||||||
|
* Tue Aug 13 2019 openEuler Buildteam <buildteam@openeuler.org> - 0.3.6-20
|
||||||
|
- Package init
|
||||||
|
|
||||||
Loading…
x
Reference in New Issue
Block a user