audiofile/audiofile-0.3.6-pull42.patch
2017-03-12 11:49:48 +01:00

177 lines
5.4 KiB
Diff

diff -Nur audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp audiofile-0.3.6-pull42/libaudiofile/modules/BlockCodec.cpp
--- audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp 2013-03-06 06:30:03.000000000 +0100
+++ audiofile-0.3.6-pull42/libaudiofile/modules/BlockCodec.cpp 2017-03-10 15:40:02.000000000 +0100
@@ -52,8 +52,9 @@
// Decompress into m_outChunk.
for (int i=0; i<blocksRead; i++)
{
- decodeBlock(static_cast<const uint8_t *>(m_inChunk->buffer) + i * m_bytesPerPacket,
- static_cast<int16_t *>(m_outChunk->buffer) + i * m_framesPerPacket * m_track->f.channelCount);
+ if (decodeBlock(static_cast<const uint8_t *>(m_inChunk->buffer) + i * m_bytesPerPacket,
+ static_cast<int16_t *>(m_outChunk->buffer) + i * m_framesPerPacket * m_track->f.channelCount)==0)
+ break;
framesRead += m_framesPerPacket;
}
diff -Nur audiofile-0.3.6/libaudiofile/modules/MSADPCM.cpp audiofile-0.3.6-pull42/libaudiofile/modules/MSADPCM.cpp
--- audiofile-0.3.6/libaudiofile/modules/MSADPCM.cpp 2013-03-06 06:30:03.000000000 +0100
+++ audiofile-0.3.6-pull42/libaudiofile/modules/MSADPCM.cpp 2017-03-10 15:40:02.000000000 +0100
@@ -101,24 +101,60 @@
768, 614, 512, 409, 307, 230, 230, 230
};
+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
+}
+
+
// Compute a linear PCM value from the given differential coded value.
static int16_t decodeSample(ms_adpcm_state &state,
- uint8_t code, const int16_t *coefficient)
+ uint8_t code, const int16_t *coefficient, bool *ok=NULL)
{
int linearSample = (state.sample1 * coefficient[0] +
state.sample2 * coefficient[1]) >> 8;
+ int delta;
linearSample += ((code & 0x08) ? (code - 0x10) : code) * state.delta;
linearSample = clamp(linearSample, MIN_INT16, MAX_INT16);
- int delta = (state.delta * adaptationTable[code]) >> 8;
+ 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;