Audio file in shorter in m4a than in PCM

Question

I recorded a voice memo that shows a length of 10:22 in iTunes. I convert it to PCM with

ffmpeg -i input.m4a -c:a pcm_s16le -f s16le temp.pcm

and then back to m4a with

ffmpeg -f s16le -i temp.pcm -c:a aac -b:a 96k -ac 2 output.m4a

and the file now shows a length of 11:18!

I confirm that both input and output files have the same beginning and ending. I tried the same conversion with another voice memo and the duration is the same.

My version is:

ffmpeg version 3.1.1 Copyright (c) 2000-2016 the FFmpeg developers
  built with Apple LLVM version 7.3.0 (clang-703.0.31)

What is going on?

Update: I paste below the full log of both commands. Notice how the first gives Duration: 00:10:22.49 and the second Duration: 00:11:17.56.

First command:

> ffmpeg -i 'input.m4a' -c:a pcm_s16le -f s16le temp.pcm
ffmpeg version 3.1.1 Copyright (c) 2000-2016 the FFmpeg developers
  built with Apple LLVM version 7.3.0 (clang-703.0.31)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/3.1.1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-opencl --enable-libx264 --enable-libmp3lame --enable-libxvid --disable-lzma --enable-vda
  libavutil      55. 28.100 / 55. 28.100
  libavcodec     57. 48.101 / 57. 48.101
  libavformat    57. 41.100 / 57. 41.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 47.100 /  6. 47.100
  libavresample   3.  0.  0 /  3.  0.  0
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.m4a':
  Metadata:
    major_brand     : M4A 
    minor_version   : 0
    compatible_brands: M4A mp42isom
    creation_time   : 2018-09-25 17:09:46
    iTunSMPB        :  00000000 00000840 000003D0 0000000001C7EBF0 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000
    encoder         : com.apple.VoiceMemos (iOS 11.4.1)
    artist          : Louise Le Gat
    title           : input
    grouping        : Short
  Duration: 00:10:22.49, start: 0.044000, bitrate: 66 kb/s
    Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 65 kb/s (default)
    Metadata:
      creation_time   : 2018-09-25 17:09:46
      handler_name    : Core Media Audio
[s16le @ 0x7fb901822c00] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, s16le, to '/Users/miguel/Downloads/temp.pcm':
  Metadata:
    major_brand     : M4A 
    minor_version   : 0
    compatible_brands: M4A mp42isom
    grouping        : Short
    iTunSMPB        :  00000000 00000840 000003D0 0000000001C7EBF0 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000
    title           : input
    artist          : Louise Le Gat
    encoder         : Lavf57.41.100
    Stream #0:0(und): Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s (default)
    Metadata:
      creation_time   : 2018-09-25 17:09:46
      handler_name    : Core Media Audio
      encoder         : Lavc57.48.101 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size=   58360kB time=00:10:22.50 bitrate= 768.0kbits/s speed= 794x    
video:0kB audio:58360kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%

Second command:

> ffmpeg -f s16le -i temp.pcm -c:a aac -b:a 96k -ac 2 output.m4a
ffmpeg version 3.1.1 Copyright (c) 2000-2016 the FFmpeg developers
  built with Apple LLVM version 7.3.0 (clang-703.0.31)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/3.1.1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-opencl --enable-libx264 --enable-libmp3lame --enable-libxvid --disable-lzma --enable-vda
  libavutil      55. 28.100 / 55. 28.100
  libavcodec     57. 48.101 / 57. 48.101
  libavformat    57. 41.100 / 57. 41.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 47.100 /  6. 47.100
  libavresample   3.  0.  0 /  3.  0.  0
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
[s16le @ 0x7fb932000000] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, s16le, from 'temp.pcm':
  Duration: 00:11:17.56, bitrate: 705 kb/s
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, 1 channels, s16, 705 kb/s
[ipod @ 0x7fb931001400] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, ipod, to 'output.m4a':
  Metadata:
    encoder         : Lavf57.41.100
    Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 96 kb/s
    Metadata:
      encoder         : Lavc57.48.101 aac
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
size=    8050kB time=00:11:17.55 bitrate=  97.3kbits/s speed=39.7x    
video:0kB audio:7935kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.447161%
[aac @ 0x7fb931001a00] Qavg: 938.144

Answer 1

Raw PCM does not store stream properties, so when you ingest a raw PCM stream, ffmpeg assumes a sampling rate of 44100 Hz whereas your source is 48000 Hz. So new duration = old duration x (48000/44100).

Export to WAV or AIFF using -c:a pcm_s16le. Alternatively, save as now and ingest using -f s16le -sample_rate 48000 -i temp.pcm. First method is recommended.