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No sound on specific files


Masterz

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Hi, have you checked the volume in the video player?

Sure. Checked and changed up/down.

 

IE 11.0.9600.19236 totally denies to playback - just tried.

Chrome - still no sound, as yesterday :)) switching between EN/RU tracks - silence on both.

embyserver.txt

ffmpeg-remux-61a16161-2948-43c1-b8e3-bdce46426361_1.txt

ffmpeg-remux-ea21047f-8e6f-47e4-8f62-d55e0efb4577_1.txt

ffmpeg-remux-8fa646b5-f71d-4d94-b150-a1ce2b916c3c_1.txt

ffmpeg-remux-63fd7aea-c0de-4c44-b0fa-f12269e7416f_1.txt

ffmpeg-remux-758b290d-ff19-4903-8533-c87964bd6aa5_1.txt

ffmpeg-remux-c3727121-3ce9-4761-b6d3-68554a13419c_1.txt

Edited by Masterz
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Happy2Play

Only thing that stands out is refframe=16 via mediainfo, Emby shows refframe=1.  And large audio delay "Delay relative to video : 8 s 250 ms" (pretty sure we saw this issue before, but might have been a subtitle issue).

 

No sound in Chrome, Firefox, or Edge all browsers converted audio to mp3 or aac.  Edge shouldn't have converted the ac3 track though.

 

Lowering bitrate to force transcoding provided audio.

 

Reencoded file provided Refframe=4 and Audio Delay relative to video : -83 ms with no playback issues.  

Edited by Happy2Play
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Only thing that stands out is refframe=16 via mediainfo, Emby shows refframe=1.  And large audio delay "Delay relative to video : 8 s 250 ms" (pretty sure we saw this issue before, but might have been a subtitle issue).

 

No sound in Chrome, Firefox, or Edge all browsers converted audio to mp3 or aac.  Edge shouldn't have converted the ac3 track though.

 

Lowering bitrate to force transcoding provided audio.

 

Reencoded file provided Refframe=4 and Audio Delay relative to video : -83 ms with no playback issues.  

 

Thank You very much.

 

Pls say about "Lowering bitrate to force transcoding provided audio." - is it achievable by Emby server/client UI settings ?

 

Thanx.

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Happy2Play

Thank You very much.

 

Pls say about "Lowering bitrate to force transcoding provided audio." - is it achievable by Emby server/client UI settings ?

 

Thanx.

 

Click the gear on the on screen display when you hit play and change the quality to 2mb or lower.  Or you can set playback home network quality in your user settings (user icon-playback).

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Nope does not help. Changing Settings->Quality to whatever - anyway no sound.

 

Set to 480p - 1Mbps - see attached "stats for nerds".

Some time it show "codec not supported, exceeds limit" (bitrate 1000kbps),

then "container not supported" (bitrate 3.9Mbps, higher than original media info  transcoding progress >100%).

 

is it OK ?

post-215087-0-46459000-1550223650_thumb.jpg

post-215087-0-51690000-1550223879_thumb.jpg

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Happy2Play

Nope does not help. Changing Settings->Quality to whatever - anyway no sound.

 

Set to 480p - 1Mbps - see attached "stats for nerds".

Some time it show "codec not supported, exceeds limit" (bitrate 1000kbps),

then "container not supported" (bitrate 3.9Mbps, higher than original media info  transcoding progress >100%).

 

is it OK ?

 

The first image looks right but the second one doesn't.  The bitrate of the media you provided was just over 2mb, so I set a bitrate of 2mb and I get audio in Chrome, Firefox and Edge went transcoded.  

 

Chrome

5c668cfb51c6f_nerd.jpg

Edited by Happy2Play
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Happy2Play

Hmm looks like Hardware Acceleration has a play in this also.  When transcoding with HWA encoder enabled I get audio.  If I disable HWA I don't get audio when transcoding.

 

Providing both logs

 

@@Luke

ffmpeg-transcode-15f74266-85da-4f1e-a212-5aba8b29401c_1 (hwa sound.txt

ffmpeg-transcode-8ba74878-755c-4d91-a48c-01bbdbd5f0a2_1-disabled hwa no sound.txt

Edited by Happy2Play
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When the audio stream is delayed by more than 8s, then that's a very specific case with an irregular video file.

Even while that may not necessarily mean that the video file is violating the specs, it's an edge case or at least and not a typical media file.

 

We cannot tweak Emby for every special or faulty media file on the planet.

At some point there's a line that needs to be drawn - specifically in situations where the required fix for a single case can cause regressions for many users.

 

I'm afraid, but we cannot safely compensate a stream delay which is that large.

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Happy2Play

When the audio stream is delayed by more than 8s, then that's a very specific case with an irregular video file.

Even while that may not necessarily mean that the video file is violating the specs, it's an edge case or at least and not a typical media file.

 

We cannot tweak Emby for every special or faulty media file on the planet.

At some point there's a line that needs to be drawn - specifically in situations where the required fix for a single case can cause regressions for many users.

 

I'm afraid, but we cannot safely compensate a stream delay which is that large.

 

Is there a reason Hardware transcode will play sound and Software transcode will now?  Shouldn't transcoding in general resolve the issue?

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Is there a reason Hardware transcode will play sound and Software transcode will now?  Shouldn't transcoding in general resolve the issue?

 

This does not seem to be caused by hw vs sw transcoding directly, it's rather a matter of how the data is running through ffmpeg:

  1. Input stream is de-multiplexed into one or more streams

    (let's assume one video and one audio stream for example)

  2. Each stream is going through its own processing pipeline
  3. The demuxer is spitting out frames and adding them to the individual pipeline

    (video frames into the video pipeline and audio frames into the audio pipeline)

  4. Those pipelines could look like this:
    • Video:  BUFFER >> HEVC Decoder >> BUFFER >> H264 Encoder >> BUFFER
    • Audio:  BUFFER >> Stream Copier >> BUFFER
  5. As a last step, there's the multiplexer (muxer) that needs to read frames from the two output buffers and mux them into the output stream

    Typically the muxer takes care that audio and video frames that are supposed to play at the same time are aligned in the output stream.

    When there's an offset between the timestamps in both buffers, it needs to wait until it finds some matching frames in both buffers.

    But the buffer sizes are limited and at some point, the muxer cannot wait anymore. Specifically in case of the ts muxer, this can lead to the muxer dropping a stream altogether when he doesn't receive any frames for it.

That's just a very rough explanation. There are a lot more details and different variants for the way frames are forwarded and buffered in the transcoding process, just think of different buffer sizes.

Those things can differ depending on which encoders and decoders are used. That's why using different video decoders and encoders can even affect the audio stream processing.

 

From experience, all these things are usually compensated well when there are track delays of like 2 or 3 seconds.

 

But 8 seconds is just too much. That may work in one case but not in other cases (= ffmpeg setups).

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I think it would be great to support this in the future, but for now you'd have to re-encode them without the audio delay. Thanks.

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