Streaming Latency

[yoloshiku]

There is a 5~10 second latency and cause video not syncing with audio. Is there anyway to reduce the latency? (Ver 1.9.0.0)

Really great app and appreciate your time and effort for developing.

[SamDel]

Thanks! It is not really possible to synchronize audio and video with this application.

The only thing you can try is:

• Go to the options tab.
• Put the stream format on wav.
• Check 'Show the lag control (experimental)'.
• Then go to the main tab and drag the lag slider to the left side and keep it there for a while.
• When you start hearing drop-outs the buffers are empty, the lag is minimal and audio quality is poor. You can then experiment to get the slider in the best position.

I don't think there's a good way to get minimal lag.

[GrahamDLL]

I get good sync by putting Chromcast Audio devices in groups in the Google Home app. I don't know whether this is relevant to syncing video and audio.

[stevorr]

Getting 16 second lag on Windows 10, tested with and without groups.

[SamDel]

There are two kinds of lag:

1. Lag between devices when playing the same stream When streaming to a group the lag between devices in the group is zero. When streaming to individual devices you can have lag.

2. Lag between the desktop and the device The lag between the desktop and the device (or group) depends on the stream format that's selected on the options tab. The device takes a ~300k buffer before it starts playing.

For me the lag between the desktop and the devices/groups is:

• When streaming in wav format: < 5 seconds
• When streaming in mp3 320 format: < 10 seconds
• When streaming in mp3 128 format: > 20 seconds

[jeppenodgaard]

Do you think increasing the sampling rate could decrease the latency?

[SamDel]

Yes, I think so.

When the size of the audio gets bigger (less compacted) the latency decreases (only when streaming in WAV format). I'm not sure how much it can be decreased, but you can try!

So, increasing the sample rate, bits per sample, number of channels , etc., should change the latency. You can also have a look at the WAV header if you want to play with it.

Let me know!

[Joshndroid]

I have noticed just with initial playing that while watching live sports i can't seem to get the audio to sync. I have played with the device buffer (in seconds) and realistically it makes no discernible difference. I am not sure how the app actually handles audio data but is it possible to just have it run as a continuous audio stream, as in a direct pipe from audio out (irrespective of if there is actually audio coming out or not)? It seems at present it recognizes that there is audio from a stream source, thinks about it, then sends it off to the device (chrome cast audio). My post may be silly, as i said i am unsure how it handles the audio stream, but it just seems like its almost laughable out of sync.. Cracking program in of itself however, for music, its awesome!

[SamDel]

Thanks!

You can't use this tool to play the audio on your Chromecast Audio device in sync with the video on your desktop.

When live sports is playing, the audio is captured by this tool. Then there's a (very small) buffer in the application, the audio is send to the device which also keeps a (bigger) buffer. Meanwhile the video continues to play. You will always have a lag of at least ~2 seconds.

[oucil]

I'm also getting a >15s lag on Windows 10 when set to WAV, and it was closer to 60s on mp3 320. I'm on a 6core/12thread i7 with 32GB ddr4 ram, so it's not the machine. The device buffer in options is set to 0, but I've tried it at 5s as well, no difference it seems. I'm only streaming to a single device (Vizio TV w. Chromecast support). Both are using wired network connections on the same network.

[SamDel]

There's a fix for the lag in 2.7.1. Can you try that one? Can you post the logs if the lag stays the same? (I noticed myself the built-in Chromecasts are not as responsive, compared to the standalone devces.)

[FA-Bubba]

I am using version 2.7.1 I have an external speaker connected to my PC which is usually off, but when I need to have sync'd audio (e.g.: If I am playing a video), I turn it on. Previously, this always worked for me, but recently, I cannot get any audio from the speaker, or even from the PC's built-in speakers.

What can I change to restore audio to my speaker when DAS is streaming to my Google Devices?

Thanks!

[SamDel]

Should work when DAS is streaming, maybe check the drivers for the speakers?

[FA-Bubba]

FIXED! Some process had slid the speaker volume to zero! Not sure how that happened, but I didn't notice since that didn't affect the DAS volume.

[amergin]

Following. The program is wonderful but I'd really like to reduce the streaming latency as well. Over 10s lag (on mp3 320kbps quality) is kinda annoying when you're mainly using the software to cast music, and sporadically choose to flip through catalog of different songs. I don't expect to able to sync video audio, but a lag of 2-5 seconds should be viable? What is Chrome browser's cast feature or Airparrot or Airfoil doing differently, they usually have a relatively short lag?

[SamDel]

The difference with the tools you mention is that they don't play live streams but recorded audio or internet streams. That way the (initial) device buffer (~350k) can be filled right away.

This tool captures the audio live from your desktop. When you use mp3 320kbps (40000 bytes/second) it takes 350k/40k=8.75 seconds before the initial device buffer is filled. So the audio that's captured now is played over (about) 9 seconds on the device. With WAV & mp3 128kbps you can do a similar calculation.

I don't know of a way to reduce the buffersize the device uses.

[oucil]

What about a logarithmic increase in buffering packet sizes until it reaches the target ~400KB, but start playing after the first packet? I haven't examined the course at all, just brainstorming.

[amergin]

The difference with the tools you mention is that they don't play live streams but recorded audio or internet streams. That way the (initial) device buffer (~350k) can be filled right away.

This tool captures the audio live from your desktop. When you use mp3 320kbps (40000 bytes/second) it takes 350k/40k=8.75 seconds before the initial device buffer is filled. So the audio that's captured now is played over (about) 9 seconds on the device. With WAV & mp3 128kbps you can do a similar calculation.

I don't know of a way to reduce the buffersize the device uses.

That is not entirely true for Airparrot. I can capture the desktop audio with that software, and the playback starts with a less than 2 seconds delay. So it's not a predefined Internet stream or prerecorded audio, but an unknown continous stream from desktop.

With Airfoil and Airparrot, I feel as though their audio quality is inferior to this software.

[starfishpatkhoo]

I have used Airfoil, Airparrot, Bubble UPnP and even various frankenstein combos of VLC + batch scripts + Python + NodeJS etc etc - all in order to cast my MP3 library (mostly in MusicBee) to my Google Home devices around my house.

ALL of them have some delay of some kind, because buffering is unavoidable. The way Chromecast works is the cast device will open a stream from the source. In order to ensure there is no drops in playback, some kind of buffering is required. This is more obvious when you need to synchronise video to audio. The "2 second delay" is a 2 second buffer (at least). In Desktop Audio Streamer, you can choose how much of a buffer you want. YMMV the number of devices you have, your network performance, your PC performance, etc etc etc.

Conclusion after my extensive testing of all the other software products, Chromecast Desktop Audio Streamer works the best, and is the most stable. In fact, I sometimes even combine it with Voicemeeter to playback audio in other rooms in my house AND my PC speaker at the same time (although it is not fully synchronised - I need to muck around with Voicemeeter's delay :)

[SamDel]

What about a logarithmic increase in buffering packet sizes until it reaches the target ~400KB, but start playing after the first packet? I haven't examined the course at all, just brainstorming.

It's the buffer on the Chromecast device. I don't think you can change it's behaviour. Odd thing is that it looks like the buffer has the same size (in bytes, not seconds) for all file formats, maybe I'm doing something wrong there.

[SamDel]

That is not entirely true for Airparrot. I can capture the desktop audio with that software, and the playback starts with a less than 2 seconds delay. So it's not a predefined Internet stream or prerecorded audio, but an unknown continous stream from desktop.

With Airfoil and Airparrot, I feel as though their audio quality is inferior to this software.

I forgot to mention that I didn't try Airparrot myself. Is it also possible to capture audio, and stream to a Chromecast device with Airparrot? Their site says it's only for AirPlay-enabled devices.

[starfishpatkhoo]

I forgot to mention that I didn't try Airparrot myself. Is it also possible to capture audio, and stream to a Chromecast device with Airparrot? Their site says it's only for AirPlay-enabled devices.

Yep, Airparrot does support Chromecast devices from Windows.

https://www.airsquirrels.com/airparrot/features/win

[elibroftw]

I recently added live audio support on my music player. To reduce the lag there were a couple of things I needed to do.

1. Specify to the chromecast that the source url is a stream and should not be buffered.
2. Force the chromecast to start playing once it's status says that it is playing even though it isn't
3. Keep track of the time it took between telling the chromecast to play and after you told the chromeast to forcefully play
4. Take this time and in your server code, instead of getting the latest system audio, slowly time.sleep(min(lag, 0.25)) before getting the latest system audio.
5. Make sure that the time between each time.sleep() call is at least 1 second. I got my lag from 10s to <1s this way. I linked my code that slowly removes the lag, you might have to do some scrolling in the future. https://github.com/elibroftw/music-caster/blob/master/music_caster.py#L601 @starfishpatkhoo if you just want to just play music on the chromecast, check out my music player https://github.com/elibroftw/music-caster/releases. BTW I used to use ChromeCast-Desktop-Audio-Streamer myself but I hated the lag so I was inspired to make my own music player.

[SamDel]

Thanks @elibroftw,

So the first x (=device playing - command play) seconds you're 'corrupting' the stream by not sending all the audio that's playing to the device. (which causes the buffer on the device to reduce). Do I understand it right?

I tried something similar with the lag control long time ago, but this is a bit smarter. I'll try to implement it later!

[elibroftw]

Not exactly corrupting. I'm just not sending anything for max 0.25 seconds at a time until the delay time I calculated becomes less than 0.25s or something. The Chromecast then stutters a couple of times since the data it's buffering is now 0.25 seconds newer each time I do the time.sleep

[SamDel]

I guess the stuttering isn't a big problem when it's only at startup for a couple of seconds!? You're only supporting wav isn't it? I remember getting really nasty sounds when doing something similar with mp3.

[elibroftw]

Yes I only support .wav, mostly because I have no idea how to convert wav data to mp3 in Python without using ffmpeg

[hibikijoji]

@SamDel I'd like to contribute to this issue as I seem to be able to replicate it.

When I boot up from Windows and make the first-time connection while no music is playing, DAS seems to negotiate with the speakers. Speakers at this point are connected in a random sequence (important) indicated by their connection sound notification. The most noticable part of the initialisation process is that the 'Buffering...' seems to hang around for a while until it progresses to playing. At this stage, my latency seems to be around 9 seconds, which makes me think that it's connection as mp3 rather than wav.

In reponse to this issue, I re-initialise by cycling between mp3 320 and wav a couple of times until the latency finally drops to 2 seconds, which I find acceptable. Most notable at this stage is: 1) speakers successfully negotiate at the same time , indicated by the sound notification. 2) the intialisation process is relatively quick.

For now, I'm ok with doing this every time I boot. I hope though that this might help resolve the issue that many of us seem to be experiencing.

NB: Thank you @SamDel for developing this software! I currently find no software that comes close - it allows me to equalize all my speaker which makes a whole world of difference (much richer, no irritating frequencies). I would like to donate if there was a facility for that. Please continue with the development 👍 It would be sad if this software lost support 😢

[SamDel]

Thanks @hibikijoji.

I can't really reproduce. I did some tests restarting DAS, Windows and/or the device but only one time (can't reproduce) I had simular behaviour. For you it's every time, right?

When changing the stream format all devices are stopped and started again, and the application buffer is cleared (here and here in code). So doing so can affect the application buffer, it can also be the devices/groups respond faster after a couple of times.

You're streaming to a group, right? (or multiple groups?) How much devices per group? Are you also booting the devices or only Windows?

[hibikijoji]

Thanks for your prompt response @SamDel ! Much appreciated.

I can confirm that I am streaming to a Google Home Group that consists of 4 nest minis and 1 chromecast (hardware HDMI audio split to 3.5 mm then to sizable bluetooth speaker). As per screenshot above, when I boot into windows, DAS starts but does not automatically boot up devices.

I have attached the log of the typical steps it takes for me to setup DAS with a windows session. With my settings set to previous post, I did the following:

1. Click on the play sign of my group (sometimes a couple of times if it's stuck on a grey 'Connected')
2. Test if audio has unacceptable lag (usually 9 seconds).
3. If so (which is almost always), flick to mp3 320 and back to wav; then test until lag becomes acceptable (around 2 seconds, which is typical wav)

Let me know if I can provide more data.

NB: On a side note, the current version doesn't seem to save setting if the minized tray setting is ticked.

EDIT: Forgot to upload the log file Debugging.txt 🤢

[SamDel]

Have you also tried to stop/start a group multiple times? Does that affect the lag?

NB: On a side note, the current version doesn't seem to save setting if the minized tray setting is ticked.

That will be fixed in the next version, see this issue.