Sample accurate audio slicing in ffmpeg?

I need to slice an audio file in .wav format into 10 second chunks. These chunks need to be exactly 10 seconds, not 10.04799988232 seconds.

the current code I am using is

ffmpeg -i test.wav -ss 0 -to 10 -c:a libfdk_aac -b:a 80k aac/test.aac

ffmpeg version 3.2.2 Copyright (c) 2000-2016 the FFmpeg developers
  built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-opencl --disable-lzma --enable-nonfree --enable-vda
  libavutil      55. 34.100 / 55. 34.100
  libavcodec     57. 64.101 / 57. 64.101
  libavformat    57. 56.100 / 57. 56.100
  libavdevice    57.  1.100 / 57.  1.100
  libavfilter     6. 65.100 /  6. 65.100
  libavresample   3.  1.  0 /  3.  1.  0
  libswscale      4.  2.100 /  4.  2.100
  libswresample   2.  3.100 /  2.  3.100
  libpostproc    54.  1.100 / 54.  1.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from '/Users/chris/Repos/mithc/client/assets/audio/wav/test.wav':
  Duration: 00:04:37.62, bitrate: 2307 kb/s
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s32 (24 bit), 2304 kb/s
Output #0, adts, to '/Users/chris/Repos/mithc/client/assets/audio/aac/test.aac':
  Metadata:
    encoder         : Lavf57.56.100
    Stream #0:0: Audio: aac (libfdk_aac), 48000 Hz, stereo, s16, 80 kb/s
    Metadata:
      encoder         : Lavc57.64.101 libfdk_aac
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s24le (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
size=     148kB time=00:00:15.01 bitrate=  80.6kbits/s speed=40.9x    
video:0kB audio:148kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%

This code does not produce exact slices, any ideas how can this be accomplished?

Not possible*. AAC audio is stored in frames which decode to 1024 samples. So, for a 48000 Hz feed, each frame has a duration of 0.02133 seconds.

If you store the audio in a container like M4A which indicates duration per-packet, the duration of the last frame is adjusted to satisfy the specified t/ss-to. But the last frame still contains the full 1024 samples. See the readout below of the last 3 frames of a silent stream specified to be 10 seconds in a M4A. Compare the packet size(s) vis-a-vis the duration.

stream #0:
  keyframe=1
  duration=0.021
  dts=9.941  pts=9.941
  size=213
stream #0:
  keyframe=1
  duration=0.021
  dts=9.963  pts=9.963
  size=213
stream #0:
  keyframe=1
  duration=0.016
  dts=9.984  pts=9.984
  size=214

If this stream were originally stored in .aac, total duration would not be 10.00 seconds. Now whether M4A does the trick for you will depend on your player.

*there is a variant of AAC which decodes to 960 samples. So, a 48 kHz audio could be encoded to a stream exactly 10 seconds long. FFmpeg does not sport such an AAC encoder. AFAIK, many apps including itunes will not play such a file correctly. If you want to encode to this spec, there's an encoder available at https://github.com/Opendigitalradio/ODR-AudioEnc