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00602537ba
The "(((audio_sample << 1) | 1) << frame->scale_factor[ch][sb])"
part of expression
"frame->sb_sample[blk][ch][sb] =
(((audio_sample << 1) | 1) << frame->scale_factor[ch][sb]) /
levels[ch][sb] - (1 << frame->scale_factor[ch][sb])"
in "sbc_unpack_frame" function can sometimes overflow 32-bit signed int.
This problem can be reproduced by first using bitpool 128 and encoding
some random noise data, and then feeding it to sbc decoder. The obvious
thing to do would be to change "audio_sample" variable type to uint32_t.
However the problem is a little bit more complicated. According
to the section "12.6.2 Scale Factors" of A2DP spec:
scalefactor[ch][sb] = pow(2.0, (scale_factor[ch][sb] + 1))
And according to "12.6.4 Reconstruction of the Subband Samples":
sb_sample[blk][ch][sb] = scalefactor[ch][sb] *
((audio_sample[blk][ch][sb]*2.0+1.0) / levels[ch][sb]-1.0);
Hence the current code for calculating "sb_sample[blk][ch][sb]" is
not quite correct, because it loses one least significant bit of
sample data and passes twice smaller sample values to the synthesis
filter (the filter also deviates from the spec to compensate this).
This all has quite a noticeable impact on audio quality. Moreover,
it makes sense to keep a few extra bits of precision here in order
to minimize rounding errors. So the proposed patch introduces a new
SBCDEC_FIXED_EXTRA_BITS constant and uses uint64_t data type
for intermediate calculations in order to safeguard against
overflows. This patch intentionally addresses only the quality
issue, but performance can be also improved later (like replacing
division with multiplication by reciprocal).
Test for the difference of sbc encoding/decoding roundtrip vs.
the original audio file for joint stereo, bitpool 128, 8 subbands
and http://media.xiph.org/sintel/sintel-master-st.flac sample
demonstrates some quality improvement:
=== before ===
--- comparing original / sbc_encoder.exe + sbcdec ---
stddev: 4.64 PSNR: 82.97 bytes:170495708/170496000
=== after ===
--- comparing original / sbc_encoder.exe + sbcdec ---
stddev: 1.95 PSNR: 90.50 bytes:170495708/170496000
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| doxygen | ||
| m4 | ||
| man | ||
| po | ||
| src | ||
| vala | ||
| .gitignore | ||
| .mailmap | ||
| autogen.sh | ||
| bootstrap.sh | ||
| configure.ac | ||
| git-version-gen | ||
| GPL | ||
| LGPL | ||
| libpulse-mainloop-glib.pc.in | ||
| libpulse-simple.pc.in | ||
| libpulse.pc.in | ||
| LICENSE | ||
| Makefile.am | ||
| orc.mak | ||
| PROTOCOL | ||
| pulseaudio-text.svg | ||
| pulseaudio.svg | ||
| PulseAudioConfig.cmake.in | ||
| PulseAudioConfigVersion.cmake.in | ||
| README | ||
| todo | ||
PULSEAUDIO SOUND SERVER WEB SITE: http://pulseaudio.org/ GIT: git://anongit.freedesktop.org/pulseaudio/pulseaudio GITWEB/CGIT: http://cgit.freedesktop.org/pulseaudio/pulseaudio/ MAILING LIST: http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss GIT COMMITS MAILING LIST: https://tango.0pointer.de/mailman/listinfo/pulseaudio-commits TRAC/BUGZILLA TICKET CHANGES MAILING LIST: http://lists.freedesktop.org/mailman/listinfo/pulseaudio-bugs IRC: #pulseaudio on irc.freenode.org CIA: http://cia.navi.cx/stats/project/polypaudio FRESHMEAT: http://freshmeat.net/projects/pulseaudio/ OHLOH: http://www.ohloh.net/projects/4038 AUTHORS: Several HACKING: In order to run pulseaudio from the build dir __OPTIMIZE__ should be disabled (look at src/pulsecore/core-util.h::pa_run_from_build_tree()), this can be done by passing "CFLAGS=-O0" to the configure script: ./autogen.sh CFLAGS="-ggdb3 -O0" LDFLAGS="-ggdb3" ./configure make ./src/pulseaudio -n -F src/default.pa -p $(pwd)/src/.libs/