[Bug] [Solved] [Confirmed] Audio Bitrate for transcodes

Server Version#: 1.22.3.4523
Player Version#: Plex Web Version 4.54.5

Hello,
I noticed that if I play one of my movies in a browser the sound quality is not good. After looking at to what it gets converted I would like to increase the bitrate to get better quality. Plex converts 5.1 AC3 640 kbit/s to 2.0 aac-lc 96 kbit/s stereo and 5.1 DTS to 2.0 aac-lc 128 kbit/s.
My movies are all remuxes i made from the bluray and I’m in no bitrate limited scenario.
How can i increase the bitrate for the transcoded audio stream?

What is your target video bitrate?
Or the maximum overall bitrate that is allowed for remote connections?

Take a look at the Plex Dashboard during such a playback (simply open a second browser). See if it says “local”, “remote” or “indirect”.

max. Also video doesn’t get transcoded, only audio.

Take a look at the Plex Dashboard during such a playback (simply open a second browser). See if it says “local”, “remote” or “indirect”.

it says remote

And is the web browser actually remote from the server?

What is set on the page Settings - Server - Remote for “Upload Speed” and “Limit remote stream bitrate”?

Yes the server is remote but client and server can reach easily more that 100 mbit/s between them. I can stream uhd remuxes with bitrates of 100 mbit/s with the plex desktop app.
Also i have set no upload limit and limit stream bitrate over the internet is set to original (no limit). This is only affecting browser streaming. If there is no transcoding because the audio is already aac or flac it works all fine. Plex does chose to transcode to such a low bitrate and i want to change that but I don’t find a way to do that.

What is the bitrate of the source file? I seem to remember that the audio bitrate is scaled up if the video bitrate is larger.

Another way to avoid this is to use a different client software than the web app. Depending on your operating system, you might get superior results by using Plex for Windows/Mac, Plex HTPC or Plex Media Player.

Format                                   : Matroska
Format version                           : Version 4
File size                                : 24.2 GiB
Duration                                 : 1 h 39 min
Overall bit rate mode                    : Variable
Overall bit rate                         : 34.7 Mb/s
Encoded date                             : UTC 2019-01-19 08:49:56
Writing application                      : mkvmerge v19.0.0 ('Brave Captain') 64-bit
Writing library                          : libebml v1.3.5 + libmatroska v1.4.8

Video
ID                                       : 1
Format                                   : AVC
Format/Info                              : Advanced Video Codec
Format profile                           : High@L4.1
Format settings                          : CABAC / 4 Ref Frames
Format settings, CABAC                   : Yes
Format settings, ReFrames                : 4 frames
Codec ID                                 : V_MPEG4/ISO/AVC
Duration                                 : 1 h 39 min
Bit rate mode                            : Variable
Bit rate                                 : 30.2 Mb/s
Maximum bit rate                         : 33.7 Mb/s
Width                                    : 1 920 pixels
Height                                   : 1 080 pixels
Display aspect ratio                     : 16:9
Frame rate mode                          : Constant
Frame rate                               : 23.976 (24000/1001) FPS
Color space                              : YUV
Chroma subsampling                       : 4:2:0
Bit depth                                : 8 bits
Scan type                                : Progressive
Bits/(Pixel*Frame)                       : 0.608
Stream size                              : 21.1 GiB (87%)
Title                                    : upgrade.2018
Default                                  : Yes
Forced                                   : No

Audio #1
ID                                       : 2
Format                                   : DTS
Format/Info                              : Digital Theater Systems
Codec ID                                 : A_DTS
Duration                                 : 1 h 39 min
Bit rate mode                            : Constant
Bit rate                                 : 768 kb/s
Channel(s)                               : 6 channels
Channel layout                           : C L R Ls Rs LFE
Sampling rate                            : 48.0 kHz
Frame rate                               : 93.750 FPS (512 SPF)
Bit depth                                : 24 bits
Compression mode                         : Lossy
Stream size                              : 549 MiB (2%)
Title                                    : ger
Language                                 : German
Default                                  : Yes
Forced                                   : No

Audio #2
ID                                       : 3
Format                                   : DTS XLL
Format/Info                              : Digital Theater Systems
Commercial name                          : DTS-HD Master Audio
Codec ID                                 : A_DTS
Duration                                 : 1 h 39 min
Bit rate mode                            : Variable
Bit rate                                 : 3 689 kb/s
Channel(s)                               : 6 channels
Channel layout                           : C L R Ls Rs LFE
Sampling rate                            : 48.0 kHz
Frame rate                               : 93.750 FPS (512 SPF)
Bit depth                                : 24 bits
Compression mode                         : Lossless
Stream size                              : 2.57 GiB (11%)
Title                                    : eng_avc.dts
Language                                 : English
Default                                  : No
Forced                                   : No

I don’t always have the choice to install different software on clients, sometimes a browser is all i can use, so fixing this on server side is the only way for me.

I will have to ask the developers what the rules are here and whether they can be adapted for larger bitrates.
I agree that 96 kbps is a bit low. (Personally, I’d consider even 128 kbps “low” in AAC.)

1 Like

Since I’m not bitrate limited even converting to flac would be fine for me.
Testing with AAC Multichannel Playback Test and flac-test-files/surround71.flac at master · sfiera/flac-test-files · GitHub chrome does playback up to 7.1 in flac and aac just fine, so I’m not sure why there is this low limit to 96 kbit/s stereo anyways.

@OttoKerner have you got any response from the devs?

@OttoKerner any feedback?

The developers say this is not intended, so they like to investigate why it happens for you.

Could you repeat your test, while your server has “debug” logging enabled?
(Do NOT enable “verbose” logging).
Server logs
Please stop and restart your server after enabling the logs!

Before starting the test, please do also enable logs in your client
Plex Web App Logs

Then play one affected video for about 1 minute.
Fetch both the server and the web app logs.
Put them into a ZIP file and send them to me per PM.
Notice of the time when you started playback of which media title would be appreciated.

The sample log you’ve sent me doesn’t exhibit the issue. The audio bitrate is 256 kbps stereo.
Audio=(id=7375 decision=transcode bitrate=256 encoder=aac channels=2 rate=48000

You won’t get surround audio in the web app.

I’m sorry and confused.
I did check with mediainfo the file and it told me this

General
Complete name                            : /mnt/d/Downloads/2.m4s
Format                                   : MPEG-4
Format profile                           : Base Media
Codec ID                                 : iso5 (iso6/mp41)
File size                                : 71.9 KiB
Duration                                 : 4 s 992 ms
Overall bit rate                         : 118 kb/s
Writing application                      : Lavf58.27.104
FileExtension_Invalid                    : mov mp4 m4v m4a m4b m4p m4r 3ga 3gpa 3gpp 3gp 3gpp2 3g2 k3g jpm jpx mqv ismv isma ismt f4a f4b f4v

Audio
ID                                       : 1
Format                                   : AAC LC
Format/Info                              : Advanced Audio Codec Low Complexity
Codec ID                                 : mp4a-40-2
Duration                                 : 4 s 992 ms
Bit rate                                 : 115 kb/s
Channel(s)                               : 2 channels
Channel layout                           : L R
Sampling rate                            : 48.0 kHz
Frame rate                               : 46.875 FPS (1024 SPF)
Compression mode                         : Lossy
Stream size                              : 70.1 KiB (97%)
Default                                  : Yes
Alternate group                          : 1

Which file is this? The playback transcoding produces a volatile chunk of your movie which exists only on the server for a limited time and in the memory of your web browser.

If I play a movie it generates two $upcountingnumber.m4s files one contains video and one audio (You can press F12 to get the download link while playing a movie). I downloaded the smaller one which contained audio and checked that with mediainfo.

You should check a chunk of a part of the movie, where there is a “dense” audio track – like loud music or a big explosion or the like.
If this is a typical movie scene with just dialog and no loud ambient noises, the actual bitrate can be lower than the “target” bitrate which has been set for the transcoder.

Could you repeat the test, but do this server configuration change first:
Settings - Server - Transcoder - ‘Transcoder quality’ = “Make my CPU hurt”