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8515ea12d2
At least 'audiofile-division-by-zero-BlockCodec-runPull.patch' could end up not being included in a tarball. * gnu/packages/patches/audiofile-Check-the-number-of-coefficients.patch: Rename to... * gnu/packages/patches/audiofile-check-number-of-coefficients.patch: ... this. * gnu/packages/patches/audiofile-division-by-zero-BlockCodec-runPull.patch: Rename to... * gnu/packages/patches/audiofile-division-by-zero.patch: ....this. * gnu/packages/patches/audiofile-signature-of-multiplyCheckOverflow.patch: Rename to... * gnu/packages/patches/audiofile-function-signature.patch: ... this. * gnu/packages/patches/audiofile-Fix-multiply-overflow-sfconvert.patch: Rename to.... * gnu/packages/patches/audiofile-multiply-overflow.patch: ... this. * gnu/packages/patches/audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch: Rename to... * gnu/packages/patches/audiofile-overflow-in-MSADPCM.patch: ... this. * gnu/packages/audio.scm (audiofile)[source]: Adjust accordingly. * gnu/local.mk (dist_patch_DATA): Likewise.
116 lines
3.6 KiB
Diff
116 lines
3.6 KiB
Diff
From: Antonio Larrosa <larrosa@kde.org>
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Date: Mon, 6 Mar 2017 13:43:53 +0100
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Subject: Check for multiplication overflow in MSADPCM decodeSample
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Check for multiplication overflow (using __builtin_mul_overflow
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if available) in MSADPCM.cpp decodeSample and return an empty
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decoded block if an error occurs.
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This fixes the 00193-audiofile-signintoverflow-MSADPCM case of #41
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---
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libaudiofile/modules/BlockCodec.cpp | 5 ++--
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libaudiofile/modules/MSADPCM.cpp | 47 +++++++++++++++++++++++++++++++++----
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2 files changed, 46 insertions(+), 6 deletions(-)
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diff --git a/libaudiofile/modules/BlockCodec.cpp b/libaudiofile/modules/BlockCodec.cpp
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index 45925e8..4731be1 100644
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--- a/libaudiofile/modules/BlockCodec.cpp
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+++ b/libaudiofile/modules/BlockCodec.cpp
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@@ -52,8 +52,9 @@ void BlockCodec::runPull()
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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 --git a/libaudiofile/modules/MSADPCM.cpp b/libaudiofile/modules/MSADPCM.cpp
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index 8ea3c85..ef9c38c 100644
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--- a/libaudiofile/modules/MSADPCM.cpp
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+++ b/libaudiofile/modules/MSADPCM.cpp
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@@ -101,24 +101,60 @@ static const int16_t adaptationTable[] =
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768, 614, 512, 409, 307, 230, 230, 230
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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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+
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+int multiplyCheckOverflow(int a, int b, int *result)
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+{
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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);
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+#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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+
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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)
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{
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int linearSample = (state.sample1 * coefficient[0] +
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state.sample2 * coefficient[1]) >> 8;
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+ int delta;
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linearSample += ((code & 0x08) ? (code - 0x10) : code) * state.delta;
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linearSample = clamp(linearSample, MIN_INT16, MAX_INT16);
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- int delta = (state.delta * adaptationTable[code]) >> 8;
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+ if (multiplyCheckOverflow(state.delta, adaptationTable[code], &delta))
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+ {
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+ if (ok) *ok=false;
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+ _af_error(AF_BAD_COMPRESSION, "Error decoding sample");
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+ return 0;
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+ }
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+ delta >>= 8;
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if (delta < 16)
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delta = 16;
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state.delta = delta;
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state.sample2 = state.sample1;
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state.sample1 = linearSample;
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+ if (ok) *ok=true;
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return static_cast<int16_t>(linearSample);
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}
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@@ -212,13 +248,16 @@ int MSADPCM::decodeBlock(const uint8_t *encoded, int16_t *decoded)
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{
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uint8_t code;
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int16_t newSample;
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+ bool ok;
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code = *encoded >> 4;
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- newSample = decodeSample(*state[0], code, coefficient[0]);
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+ newSample = decodeSample(*state[0], code, coefficient[0], &ok);
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+ if (!ok) return 0;
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*decoded++ = newSample;
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code = *encoded & 0x0f;
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- newSample = decodeSample(*state[1], code, coefficient[1]);
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+ newSample = decodeSample(*state[1], code, coefficient[1], &ok);
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+ if (!ok) return 0;
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*decoded++ = newSample;
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encoded++;
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