FFMPEG - 5.1 EAC3 encode output quality

Question

I have PCM Wave source files that are 5.1 surround. My goal is to convert this to EAC3 at 192k per Apple recommendations.

Here is the MediaInfo of my source media:

General
Complete name                            : X:\******WAVE\5.1_PCM.wav
Format                                   : Wave
File size                                : 2.38 GiB
Duration                                 : 49 min 3 s
Overall bit rate mode                    : Constant
Overall bit rate                         : 6 949 kb/s
Producer                                 : Pro Tools
Encoded date                             : 2019-03-00 17:03:25
Producer_Reference                       : 15uZdJ9Ng2naaaGk

Audio
Format                                   : PCM
Format settings                          : Little / Signed
Codec ID                                 : 00000001-0000-0010-8000-00AA00389B71
Duration                                 : 49 min 3 s
Bit rate mode                            : Constant
Bit rate                                 : 6 912 kb/s
Channel(s)                               : 6 channels
Channel layout                           : L R C LFE Lb Rb
Sampling rate                            : 48.0 kHz
Bit depth                                : 24 bits
Stream size                              : 2.37 GiB (99%)

I'm encoding this with ffmpeg by using very simply:

ffmpeg -i IN.wav -c:a eac3 -b:a 192k -ar 48k OUT.mp4

Here is the MediaInfo of the output:

General
Complete name                            : X:\******\6ch_eac3.mp4
Format                                   : MPEG-4
Format profile                           : Base Media
Codec ID                                 : isom (isom/iso2/mp41)
File size                                : 67.4 MiB
Duration                                 : 49 min 3 s
Overall bit rate mode                    : Constant
Overall bit rate                         : 192 kb/s
Recorded date                            : 2019-03-0
Writing application                      : Lavf58.20.100

Audio
ID                                       : 1
Format                                   : E-AC-3
Format/Info                              : Enhanced AC-3
Commercial name                          : Dolby Digital Plus
Codec ID                                 : ec-3
Duration                                 : 49 min 3 s
Bit rate mode                            : Constant
Bit rate                                 : 192 kb/s
Channel(s)                               : 6 channels
Channel layout                           : L R C LFE Ls Rs
Sampling rate                            : 48.0 kHz
Frame rate                               : 31.250 FPS (1536 SPF)
Bit depth                                : 16 bits
Compression mode                         : Lossy
Stream size                              : 67.4 MiB (100%)
Service kind                             : Complete Main
Default                                  : Yes
Alternate group                          : 1

The thing is the output quality is really bad. It sounds extremely compressed, judging by my ear it sounds like about 32kbps. My theory is that the output is for some reason 192k divided by 6 channels, each channel being 32kbps and the whole audio track is 192k overall. To my knowledge, that's not how its supposed to work and it should sound like the equivalent of 192k for each channel in one bitstream.

I also encoded the same source file using Telestream Vantage, here is the MediaInfo which is extremely similar to the ffmpeg output above:

General
Complete name                            : X:\******\WAVE\Internet_192k_6ch_telestream.mp4
Format                                   : MPEG-4
Format profile                           : Base Media / Version 2
Codec ID                                 : mp42 (mp42/isom)
File size                                : 69.0 MiB
Duration                                 : 49 min 3 s
Overall bit rate mode                    : Constant
Overall bit rate                         : 197 kb/s
Encoded date                             : UTC 2019-08-01 15:30:58
Tagged date                              : UTC 2019-08-01 15:30:58

Audio
ID                                       : 1
Format                                   : E-AC-3
Format/Info                              : Enhanced AC-3
Commercial name                          : Dolby Digital Plus
Codec ID                                 : ec-3
Duration                                 : 49 min 3 s
Source duration                          : 49 min 3 s
Bit rate mode                            : Constant
Bit rate                                 : 192 kb/s
Channel(s)                               : 6 channels
Channel layout                           : L R C LFE Ls Rs
Sampling rate                            : 48.0 kHz
Frame rate                               : 31.250 FPS (1536 SPF)
Bit depth                                : 16 bits
Compression mode                         : Lossy
Stream size                              : 67.4 MiB (98%)
Source stream size                       : 67.4 MiB (98%)
Service kind                             : Complete Main
Encoded date                             : UTC 2019-08-01 15:30:58
Tagged date                              : UTC 2019-08-01 15:30:58

This file sounds fantastic, like true 192k with about the same filesize as the ffmpeg encode.

So clearly I am doing something wrong with mapping or bitstream or something.

Here is the full ffmpeg log:

N:\_TOOLS\ffmpeg\ffmpeg-4.1.3-win64-static\bin>ffmpeg -i X:\02_BROADCAST\5.1_PCM
.wav -c:a eac3 -b:a 192k -ar 48k X:\02_BROADCAST\6ch_eac3.mp4
ffmpeg version 4.1.3 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 8.3.1 (GCC) 20190414
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfi
g --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-lib
freetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amr
wb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --
enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-l
ibwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --
enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --en
able-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --en
able-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --e
nable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enab
le-avisynth
  libavutil      56. 22.100 / 56. 22.100
  libavcodec     58. 35.100 / 58. 35.100
  libavformat    58. 20.100 / 58. 20.100
  libavdevice    58.  5.100 / 58.  5.100
  libavfilter     7. 40.101 /  7. 40.101
  libswscale      5.  3.100 /  5.  3.100
  libswresample   3.  3.100 /  3.  3.100
  libpostproc    55.  3.100 / 55.  3.100
Input #0, wav, from 'X:\02_BROADCAST\5.1_PCM.wav':
  Metadata:
    encoded_by      : Pro Tools
    originator_reference: 15uZdJ9Ng2naaaGk
    date            : 2019-03-0
    creation_time   : 17:03:25
    time_reference  : 172828656
  Duration: 00:49:03.23, bitrate: 6948 kb/s
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1, s32 (2
4 bit), 6912 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s24le (native) -> eac3 (native))
Press [q] to stop, [?] for help
Output #0, mp4, to 'X:\02_BROADCAST\6ch_eac3.mp4':
  Metadata:
    encoded_by      : Pro Tools
    originator_reference: 15uZdJ9Ng2naaaGk
    date            : 2019-03-0
    time_reference  : 172828656
    encoder         : Lavf58.20.100
    Stream #0:0: Audio: eac3 (ec-3 / 0x332D6365), 48000 Hz, 5.1, fltp (24 bit),
192 kb/s
    Metadata:
      encoder         : Lavc58.35.100 eac3
size=   68983kB time=00:49:03.22 bitrate= 192.0kbits/s speed=73.3x
video:0kB audio:68982kB subtitle:0kB other streams:0kB global headers:0kB muxing
 overhead: 0.001441%

N:\_TOOLS\ffmpeg\ffmpeg-4.1.3-win64-static\bin>

Anyone know what I am running into?

As a note this is concerned with video production because this EAC3 is going into a DASH wrapper with video immediately after this encode.

Edit: Something I just realized on this and tested is that the audio quality that I hear when using ffmpeg to encode to ac3 and then to encode to eac3 is the same. But as eac3 is almost twice as efficient as ac3, I would expect the eac3 encode to be much higher quality and sound less compressed. Case in point is that the Telestream Vantage encode is much higher quality than an ac3 at the same bitrate. Is this a fault of of the ffmpeg encoder that is used for eac3?

Answer 1

Rather old question, but responding FBO whomever stumbles upon this in the future (besides me; lol)....

Your issue was the fact you set total average bit rate to 192kbps, not per channel. Presuming ffmpeg's default AC-3 encoder, if you simply set the sample rate to 48kbps (which you did in your example above), you'd be encoding at 1,536kbps for all channels. That is because the only bit depth that codec supports is 32 bits per sample.

32 bit depth x 48,000 sampling frequency = 1,536,000 bits per second

If you had simply omitted this portion of your command line you should have gotten that bit rate:

-b:a 192k

That is because what you did there was limit the average bit rate to 192kbps, which is why you ended up with poor quality sound in your target file. At 6 channels (5+1), you're looking at 32 kbps per channel (192 / 6 = 32).

To attain your goal of 192 kbps per channel, you should have limited the output to 6 x 192 or 1,152 kbps, like this:

-b:a 1152k

Cheers. :-)