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janbar
Desktop audio from pulseaudio to Playback has High Latency
When I run the application and I am playing a video on youtube for example, the audio is delayed by few seconds. I do not mind a small amount of latency but the time in between skipping from one part of an audio stream to another is far from negligible.
OS: Ubuntu 18.04.3 LTS x86_64 pulseaudio version: pulseaudio 11.1 noson version: noson 4.2.5.snap
That is normal. The audio stream needs to be processed by the app and your sonos device. This feature is designed to cast music, not to watch a video program.
Hello I start using the app last week and have the same problem. Since i have no computer speaker, the function for video is necessary. But i can understand it is not a priority since it works so great for music.
OS : fedora 32
There is no solution to synchronize the video exposed by a video player on your computer and the audio played by a remote device like Sonos player. The two parts cannot be synchronized. A sync of video and audio is built in a single system or a particular system doing for that.
There is most certainly a way to reduce latency dramatically, though. Right now there's around ~8 seconds of latency, thanks to a number of insanely huge buffers in the pulse->flac pipeline. For example, change #define SAMPLES 1024 here:
https://github.com/janbar/noson/blob/c63343f95cece5f462d812ac66141f16cab20610/noson/src/flacencoder.cpp#L24
into #define SAMPLES 128, and you'll already see a dramatic improvement. There are two additional buffers I saw that could probably be dramatically reduced as well.
Aside from the harder question of video synchronization, just getting latency around a second or less would be a significant improvement in terms of basic usability.
@zx2c4 , the value of SAMPLES isn't linked with a buffer size. It is the max size of chunk of data I could send to the Flac encoder. The value of 1024 samples is 1/40 second, so a very short period. Also I checked my code and there is no "blocking" buffer. The only thing that could get a lag should be the chunk size of the HTTP response sent to the Sonos device: for now 16KB, and so ~1/4 of second. Here the culprit seems to be the Sonos device which request many data chunk before starting to play the stream.
If you tried with 128 for SAMPLES, and seen a dramatic improvement, there is probably something relating with the Flac decoder in the Sonos device ... to be checked.
I refactored the code around the FLAC encoder used to pulse. I found a way to reduce the lag and I succeeded to keep less than 1 seconde. The root cause of this issue is there was no buffer ! My code push-pull the data without buffering, and seems that causes a lag from the client (Sonos device) on startup. Probably it needs a minimal set of frame and it computes its lag on the initial timeout to retrieve first frames.
An update of the app including changes for the pulse streamer will be available from my PPA, in couple of hours.
https://launchpad.net/~jlbarriere68/+archive/ubuntu/noson-app
The LAG is very short because I manage 1 seconde of blank on startup to limit output rate and let the device to wake up. Then I disable the mute and send full stream...
Hi I'm currently building this on raspbian to create a media center with some sonos (so I can't check the PPA). Does your fix been merged ? because I tested today and everything works perfectly fine, but I still got the latency described in this issue..
@AlphaBril , the latenty is less than before but still few seconds we cannot handle: the Sonos device use an internal buffer.
In my first test I had a latency of 10s (9s of internal buffer), luckily its down now to 2.5/3
The options that came to mind for further reducing it:
- Not sure if this would work, but if the sound is streamed with higher bitrate/quality the buffer might fill up faster and it couldn't hold as much "time" as it does now.
- I assume multiple Sonos speakers share the sound with very low latency, so we could pretend to be a speaker and use their communication interface. Obviously that would be a lot of work to reverse engineer and to implement.
Do you know how e.g. apple w/ airplay manages to synchronize videos w/ the sound? Because if I watch a video on a mac, and stream the audio to airplay, it is in sync
I'd be also interesting how the Sonos Port product handles that, and if it suffers from the same issue. Anyone has experience on that?
In the past I have used VLC with the Audio Track Synchronisation settings. Once you have the values its pretty easy to setup again.
I wonder if there's any way to query the audio lag of the pipeline somewhere? At least for Bluetooth audio, this seems to be a solved problem and vlc syncs its video to the BT audio lag of a wireless headphone, so there must be some infrastructure handling this alreay.
(for the record, it's ca. 9s delay with vlc & noson on my system. I find it quite difficult to find the proper sync value for this by hand and would be grateful for any mechanism that automates this.)