Unable to re-encode audio stream with different bitrate using ffmpeg

Question

A similar question has been asked elsewhere without a satisfying answer for me.

The following ffmpeg command results in error. When "-c:a copy" is added to just copy the audio stream, the video stream is nicely re-encoded with the requested bitrate.

I'm running this on OS X Yosemite 10.10.2.

How should this error be resolved?

ffmpeg -i input.mp4 -b:v 8M  -b:a 10k output.mp4
ffmpeg version 1.0.1 Copyright (c) 2000-2012 the FFmpeg developers
  built on Dec  7 2012 09:31:51 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
  configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --enable-libgsm --arch=x86_64 --enable-runtime-cpudetect
  libavutil      51. 73.101 / 51. 73.101
  libavcodec     54. 59.100 / 54. 59.100
  libavformat    54. 29.104 / 54. 29.104
  libavdevice    54.  2.101 / 54.  2.101
  libavfilter     3. 17.100 /  3. 17.100
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 1
    compatible_brands: mp41mp42isom
    creation_time   : 2015-02-06 08:37:52
  Duration: 00:00:48.04, start: 0.000000, bitrate: 12332 kb/s
    Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 1920x960 [SAR 1:1 DAR 2:1], 12201 kb/s, 14.98 fps, 14.99 tbr, 600 tbn, 1200 tbc
    Metadata:
      creation_time   : 2015-02-06 08:37:52
      handler_name    : Core Media Video
    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, s16, 125 kb/s
    Metadata:
      creation_time   : 2015-02-06 08:37:52
      handler_name    : Core Media Audio
[libx264 @ 0x7fbbba009c00] using SAR=1/1
[libx264 @ 0x7fbbba009c00] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64
[libx264 @ 0x7fbbba009c00] profile High, level 4.0
[libx264 @ 0x7fbbba009c00] 264 - core 128 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=14 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=8000 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
[NULL @ 0x7fbbba00aa00] Codec is experimental but experimental codecs are not enabled, try -strict -2
Output #0, mp4, to 'output.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 1
    compatible_brands: mp41mp42isom
    Stream #0:0(und): Video: h264, yuv420p, 1920x960 [SAR 1:1 DAR 2:1], q=-1--1, 8000 kb/s, 90k tbn, 14.99 tbc
    Metadata:
      creation_time   : 2015-02-06 08:37:52
      handler_name    : Core Media Video
    Stream #0:1(und): Audio: none, 44100 Hz, mono, flt, 10 kb/s
    Metadata:
      creation_time   : 2015-02-06 08:37:52
      handler_name    : Core Media Audio
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> libx264)
  Stream #0:1 -> #0:1 (aac -> aac)
Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height

Answer 1

Here's your error:

Codec is experimental but experimental codecs are not enabled, try -strict -2

When converting to MP4, ffmpeg tries to choose an AAC audio encoder. Since your ffmpeg is not compiled with libfaac, libfdk-aac or libvo-aacenc, all of which are third party encoders, it defaults to the internal AAC encoder (called aac).

However, this encoder is considered experimental in your build of ffmpeg and will only run when you add -strict -2 or -strict experimental to the options.

Some notes:

• As of FFmpeg 3.0, the built-in aac encoder is no longer considered experimental.

• You are trying 10 kBit/s as audio bitrate which is much too low for the built-in AAC-LC encoder. This bitrate would only work for HE-AAC v2, which libfdk-aac could do. And it'd still sound bad.

• In general, I would always explicitly specify the encoder you want to use. Saves you a bit of trouble when the defaults are not sane.

• I'd recommend you update your ffmpeg version to something more recent, either by downloading a static build, or compiling it yourself. When you compile, you have the choice of using libfdk-aac, which offers better quality than aac, and also has a proper VBR option.

• Please read the AAC encoding guide for more info.