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shogo4405 avatar shogo4405 commented on June 25, 2024 2

There is a difference between the PTS (Presentation Time Stamp) calculated theoretically and the PTS received in practice. I believe this difference is manifesting as silence.
スクリーンショット 2023-10-03 1 46 47

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shogo4405 avatar shogo4405 commented on June 25, 2024 2

Maybe this fix could have improved it. If you don't mind, I would like to ask you to test it. #1310

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shogo4405 avatar shogo4405 commented on June 25, 2024 1

I confirmed that the issue can be reproduced when launching Example.app while having a Discord call.
As indicated by 'Offset 128,' I confirmed that there is a corresponding gap in that area. It seems like using this value should work well.

CMBlockBuffer 0x282eea880 totalDataLength: 7680 retainCount: 1 allocator: 0x1e753bb30 subBlockCapacity: 2
 [0] 7680 bytes @ offset 128 Buffer Reference:
    CMBlockBuffer 0x282eea910 totalDataLength: 7832 retainCount: 1 allocator: 0x1e753bb30 subBlockCapacity: 2
     [0] 7832 bytes @ offset 0 Memory Block 0x116782240, 7832 bytes (allocator 0x2827f2700)
}

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shogo4405 avatar shogo4405 commented on June 25, 2024

The fix made in #1208 was for the case of 2-channel stereo. It might not work correctly for 3 channels or more. It seems like the iPhone 14 Pro is detecting 4 channels.

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CedricEugeni avatar CedricEugeni commented on June 25, 2024

Thanks for your reply @shogo4405 😊

I tried to force the AVAudioSession preferredIntputChannels to 1 or 2 using the setPreferredIntputChannels.
While it seems to work on my iPhone 13mini, it seems it doesn't work on iPhone 14.

So if I have to deal with 4-channels samples. what would be the best way to deal with it inside Haishinkit from your point of view ?
Would I need to "remove" 2 channels from the CMSAmpleBuffer and then give it to the actual Haishinkit code to make it work or have you another idea about something that would make it work ? 🤔

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shogo4405 avatar shogo4405 commented on June 25, 2024

I have no idea. We have recently added support for multi-channels in #1273 and #1280. I have also tested it and it seems to work fine with 4 channels.

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leo150 avatar leo150 commented on June 25, 2024

You can try to discard channels within HK by providing a channel map. Negative value is used to discard an input channel:

myStream.audioSettings.outputChannelsMap = [0: 0, 1: 1, 2: -1, 3: -1]

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leo150 avatar leo150 commented on June 25, 2024

Also the format descriptions that show on the screenshots look like "input" audio format. HK does conversion from PCM to compressed AAC format. The "output" format can be found in AudioCodecSettings. There's hardcoded maximum number of output channels which is 2. Any extra channels are removed, say you convert from 4 to 2 channels, channels 3 and 4 will be removed.

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shogo4405 avatar shogo4405 commented on June 25, 2024

I've made significant changes to the internal audio processing in #1294. There's a possibility that it brings improvements. You can test it on the 'main' branch.

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CedricEugeni avatar CedricEugeni commented on June 25, 2024

I'm currently testing it and will tell you if there is improvements or not.
If this does not improve things, I'll take a look and try some things about what @leo150 said earlier to see if it helps.
Thanks for your replies

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CedricEugeni avatar CedricEugeni commented on June 25, 2024

Hi,

I took some time to make several tests and waited for the iPhone 15 Pro release to confirm what I thought.
My issue is also happening on the iPhone 15 Pro.

As @leo150 mentioned, I tried to set audioSettings as below:

stream.audioSettings.bitRate = <info given by my backend>
stream.audioSettings.channels = 1
stream.audioSettings.channelMap = [0: 0, 1: 0, 2: -1, 3: -1]

But it seems it doesn't have any effect on the audio.

I also tried to update the library to use the latest code but it also doesn't seem to change anything.

One piece of info I may provide you is that my AVAudioSession is using playAndRecord category & videoChat mode because I have possible audio feedback on speaker and I want echo canceling/voice processing.

If I'm using AirPods or so, I use default mode and it works as expected, but as soon as my AVAudioSession change, sound is behaving in this weird way.

I also tried, when AVAudioSession change, to detach audio from rtmpStream and then reattach it, thinking that maybe it'd get new settings from incoming audio and set things up in a different way that'd be able to handle audio. But it doesn't work either.

Are all of these info giving you any piece of idea or advice for me ?

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asticode avatar asticode commented on June 25, 2024

@shogo4405 @leo150 @CedricEugeni unfortunately we're facing the very same issue. What we've found is that the crackling sound is due to recurring silences or missing samples in the audio. Here are Audacity screenshots :

missing_samples

Here you can see the recurring silences or missing samples.

missing_frame_with_samples

What we've found is that there's exactly 1024 samples between each recurring silences or missing samples. Therefore we concluded they are either at the beginning or at the end of each audio frame.

We've tried to dig into the resampler and ring buffer but we couldn't find anything that would help us determine where this could come from.

Do the screenshots and/or explanation make you think of anything ? Any help would be greatly appreciated 👍

Cheers
Quentin

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shogo4405 avatar shogo4405 commented on June 25, 2024
  1. Fork this code and provide logs of the print results showing the occurring symptoms. A log of about 30 seconds should be sufficient.
  2. If possible, please provide a simple code snippet for reproduction along with server access details. You can share the reproduction code via a pull request, and the server credentials can be sent via email.

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asticode avatar asticode commented on June 25, 2024
  1. We've managed to pin point what is causing the issue : when dealing with WebRTC and 4 channels, CMSampleBuffers sent by the AVCaptureSession sometimes start with 256 "padding/dummy" samples equal to 0 (it happens when the buffer number of samples > 960, there's no problem when the number of samples is exactly 960), the rest of the buffer being fine (here is an example). It leads to 64 silence samples being added per channel, which is exactly the length of silence observed in Audacity.

Unfortunately there doesn't seem to be any information in the CMSampleBuffer informing its user that there's padding to be skipped at the start of the buffer 😱

Do you have any thought on that? Or any idea on how we could go about fixing this issue?

  1. It's not easy to provide reproducible steps, but we'll try it when we exhaust all other possibilities 👍

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shogo4405 avatar shogo4405 commented on June 25, 2024

Fork this code and provide logs of the print results showing the occurring symptoms. A log of about 30 seconds should be sufficient.

Code is here.
https://github.com/shogo4405/HaishinKit.swift/tree/feature/print-samplebuffer

I'm not interested in the byte sequence of the CMSampleBuffer. I want to know the PTS (Presentation Time Stamp). Please provide the dump result.

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asticode avatar asticode commented on June 25, 2024

Fork this code and provide logs of the print results showing the occurring symptoms. A log of about 30 seconds should be sufficient.

Code is here. https://github.com/shogo4405/HaishinKit.swift/tree/feature/print-samplebuffer

I'm not interested in the byte sequence of the CMSampleBuffer. I want to know the PTS (Presentation Time Stamp). Please provide the dump result.

Here's our dump. I had to add a ".zip" extension so that Github let me upload it.

Dump_haishinkit.zip

Is it what you were looking for?

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asticode avatar asticode commented on June 25, 2024

Thanks for clearing that out 👍

Is there anything this lib or we could do to fix the issue? 🤔

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asticode avatar asticode commented on June 25, 2024

First off, thanks for providing a fix to test ❤️

Here's how it went: we had 2 problems

  • a crackling sound
  • audio was getting late compared to video which resulted in linear desynchronization between audio and video

Your patch fixed the second problem but not the first one.

Therefore I think it would be a good idea to merge it, but we still have an issue with the crackling sound.

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asticode avatar asticode commented on June 25, 2024

That looks pretty good indeed! Let us know whether you need any help testing a fix 👍

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shogo4405 avatar shogo4405 commented on June 25, 2024

It's being discussed on the developer forum.
https://developer.apple.com/forums/thread/715177

It seems like there is blank data at the beginning like this, which can disrupt calculations such as PTS (Presentation Time Stamp). When I remove the blank data, it sounds correct.

▫️▫️▫️▫️▪️▪️▪️▪️▪️
▫️▫️▫️▫️▪️▪️▪️▪️▪️
▪️▪️▪️▪️▪️▪️▪️▪️▪️
▫️▫️▫️▫️▪️▪️▪️▪️▪️
▫️▫️▫️▫️▪️▪️▪️▪️▪️
▪️▪️▪️▪️▪️▪️▪️▪️▪️
▫️▫️▫️▫️▪️▪️▪️▪️▪️
▫️▫️▫️▫️▪️▪️▪️▪️▪️
▪️▪️▪️▪️▪️▪️▪️▪️▪️

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asticode avatar asticode commented on June 25, 2024

Did you manage to find a way to automatically remove that annoying blank data based on the CMBlockBuffer information (like the offset you mentioned earlier) or are you missing information?

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shogo4405 avatar shogo4405 commented on June 25, 2024

I haven't found a way to retrieve it from the parameters.

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asticode avatar asticode commented on June 25, 2024

@shogo4405 what are you looking to do regarding this issue? We're still experiencing crackling noise.

Cheers

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shogo4405 avatar shogo4405 commented on June 25, 2024

Are you saying that it's still occurring in the post-fix branch at https://github.com/shogo4405/HaishinKit.swift/pull/1310/files? If so, I won't be able to fix it because I can't reproduce it in my environment.

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asticode avatar asticode commented on June 25, 2024

With your post-fix branch, we're experiencing:

  • way less crackling noise but still random ones every 1 or 2 seconds. they are far more difficult to hear but they are there
  • if we record for a few minutes, we see that audio/video get unsynchronized (video is late compared to audio)

Random crackling noises may be due to audio/video getting unsynchronized 🤔

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shogo4405 avatar shogo4405 commented on June 25, 2024

The direction of the modifications seems promising. I believe that with extensive debugging, improvements can be achieved. On the other hand, I'm currently allocating my time for the 1.7.0 release, so the subsequent fixes will have to wait.

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asticode avatar asticode commented on June 25, 2024

OK thanks for the clarifications 👍 we're available if you need help to debug or test improvements whenever you go back to that issue. Thanks again ❤️

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levs42 avatar levs42 commented on June 25, 2024

As a temporary solution, I disabled voice processing in voice_processing_audio_unit.mm:76 and the issue is gone. In our case the voice processing isn't a requirement.

vpio_unit_description.componentSubType = kAudioUnitSubType_RemoteIO;

I don't have time to create a reproducible example, but ideally we need to add a voice processing unit to the HaishinKit example app so the issue could looked into.

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shogo4405 avatar shogo4405 commented on June 25, 2024

You can reproduce the issue by starting the HaishinKit sample app while on a call in Discord. Since I don't have a good improvement plan at the moment, I will move this to the Discussion section for now.

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