Google helped me find this thread- looks like the same issue I have been trying to figure out.
Disabling upsampling seems to be a workaround but I just wanted to check that this was still happening to those who were able to reproduce it?
Sorry Damien3, its been a weird summer.
I have no problem with Disabled, x2 only or to DSD.
It’s worst with use maximum frequency. I’m just pretending that setting doesn’t exist now 
Has any solution been found for this? I have been hoping that with all the updates it would have been fixed. It happens to me on all upsampling to DSD and all higher rates of PCM. Most noticeable on long tracks. I am on Windows 10 with 8GB memory. With upsampling to DSD the channels get reversed on every other track as well. I am using Qobuz.
Hello @GB46,
this issue is more related on your DAC manufacturer as it depense en on how they want to deal with upsampling. Some of them want it do be only done by the DAC and it also appears that if you put the setting “maximum frequency” you can have dropouts issue or the song can be cut before the end. Therefore you need to either reduce the setting to “Power of 2” or “x2” to check if you still have the issue.
Many thanks for your swift reply. Using ‘Power of 2’ or ‘device max’ produces the problem. Restricting to ‘x2’ or turning off upsampling works with no problems. Do you think this is also the issue with channel switching (left to right and vice versa) when converting to DSD? For your information the DAC is OPPO 205 via usb input.
We didn’t had such report about it, do you have a setting in your Oppo for channel switching?
No, that is not available, only possible to switch the polarity of the XLR terminals which would I guess be a phase change. It is a bit academic to me though as going to DSD as ‘upsampled’ also causes the cutting short of the tracks (nearly always but depends on the track length.) Incidentally it may not be obvious unless playing classical music where orchestra layout is usually consistent. You find yourself sitting behind the orchestra instead of in front!
Increasing the buffer size to 24GB fixed my issue; thanks for the solution.I now have “device max freq” upscaling.
I also got this issue, cut off some track before end, when upsamping use Sox, but iZotope is ok, and not active upsampling is also ok.
I think Sox library not use right.
I continue to experience a half second or less cutoff at the end of all tracks, upsampled or not, with or without signal processing (Dirac Live VST 3 plugin). Studio (now version 5.5) has not fixed this issue.
I listen to classical, opera and jazz. This is no issue on most classical and opera albums and all jazz, as there is enough runout track length after a final note, so the music itself is not affected. HOWEVER, a number of classical pieces and operas contains points tracks must play without cutoff as there is no break between movements or “arias” (e.g. Beethoven 5 (movements 3 and 4) and 6 (movements 3-5), most Wagner, Verdi Falstaff and Otello). truncating the lead into the next movement in these pieces is intolerable, AND DOES NOT OCCUR WITH OTHER SOFTWARE. Here is my debug info:
Current version: 1.5.5 (10505)
Connected account: daniel.gsovski@gmail.com (Daniel Gsovski)
SIGNAL PROCESSING:
Polarity Inversion:
Globally: OFF
Per track: OFF
Effects plugins ACTIVE in offline mode
VST3 plugin #0: C:\Program Files\Common Files\VST3\DiracLiveProcessor.vst3
ClassID: 56535456376C6564697261636C697665
VST3 plugin #1: None
VST3 plugin #2: None
VST3 plugin #3: None
UPSAMPLING:
r8brain custom frequencies
r8brain filter parameters
Bandwidth = 99.5%
Stop band attenuation 218dB
Phase linear
AUDIO VOLUME:
Max allowed volume: 100
Replay Gain: by album
SW volume control: ON
LIBRARY SETTINGS:
Sync list: 1 folders
AUTO: D:\MyMusic
Library database path: C:\Users\dgsov\AppData\Local\Audirvana\Audirvana\AudirvanaDatabase.sqlite
Remote Control server:
Listening on 192.168.86.25 on port 54977
ACTIVE STREAMING SERVICES
TIDAL: Connected as HIFI
=================== AUDIO DEVICE ========================
Max. memory for audio buffers: 632MB
Local Audio Engine: Kernel Streaming
Preferred device:
Speakers (iFi (by AMR) HD+ USB Audio)
Model UID:MMDEVAPI\AudioEndpoints
UID:\?\SWD#MMDEVAPI#{0.0.0.00000000}.{84ab8581-e19f-4240-97cb-bbcaf0bdf0fb}#{e6327cad-dcec-4949-ae8a-991e976a79d2}
Currently playing in Integer Mode:
Device: 2ch 32bits Integer, 8 bytes per frame 176.4kHz
Active Sample Rate: 176.4kHz
Bridge settings:
Sample rate limitation: 192kHz
Sample rate switching latency: none
Limit bitdepth to 24bit: OFF
Mute during sample rate change: ON
Selected device:
Local audio device
Speakers (iFi (by AMR) HD+ USB Audio)
Manufacturer: AMR
Model Name: Unknown manufacturer
Model UID: TUSBAUDIO_ENUM\VID_20B1&PID_3008&KS
UID: \?\SWD#MMDEVAPI#{0.0.0.00000000}.{84ab8581-e19f-4240-97cb-bbcaf0bdf0fb}#{e6327cad-dcec-4949-ae8a-991e976a79d2}
Kernel Streaming capable
6 available sample rates up to 192000Hz
44100
48000
88200
96000
176400
192000
Volume Control
Physical: No
Virtual: No
MQA capability
Auto-detect MQA devices: Yes
Not a MQA device, user set to MQA Renderer
DSD capability: Unhandled
Device audio channels
Preferred stereo channels L:0 R:1
Channel bitmap: Ox3, layout:
Channel 0 mapped to 0
Channel 1 mapped to 1
Audio channels in use
Number of channels: 2
Use as stereo device only: No
Simple stereo device: No
1 output streams:
Number of active channels: 2, in 1 stream(s)
Channel #0 :Stream 0 channel 0
Channel #1 :Stream 0 channel 1
2 ch Integer PCM 32bit little endian 44.1kHz
2 ch Integer PCM 32bit little endian 48kHz
2 ch Integer PCM 32bit little endian 88.2kHz
2 ch Integer PCM 32bit little endian 96kHz
2 ch Integer PCM 32bit little endian 176.4kHz
2 ch Integer PCM 32bit little endian 192kHz
2 ch Integer PCM 24bit little endian 44.1kHz
2 ch Integer PCM 24bit little endian 48kHz
2 ch Integer PCM 24bit little endian 88.2kHz
2 ch Integer PCM 24bit little endian 96kHz
2 ch Integer PCM 24bit little endian 176.4kHz
2 ch Integer PCM 24bit little endian 192kHz
2 ch Integer PCM 16bit little endian 44.1kHz
2 ch Integer PCM 16bit little endian 48kHz
2 ch Integer PCM 16bit little endian 88.2kHz
2 ch Integer PCM 16bit little endian 96kHz
2 ch Integer PCM 16bit little endian 176.4kHz
2 ch Integer PCM 16bit little endian 192kHz
Local devices found : 3
Device #0: Speakers (iFi (by AMR) HD+ USB Audio) Manufacturer: AMR Model UID: TUSBAUDIO_ENUM\VID_20B1&PID_3008&KS UID: \?\SWD#MMDEVAPI#{0.0.0.00000000}.{84ab8581-e19f-4240-97cb-bbcaf0bdf0fb}#{e6327cad-dcec-4949-ae8a-991e976a79d2} Model Name: Unknown manufacturer
Device #1: JRiver Media Center 25 (JRiver Media Center) Manufacturer: JRiver Model UID: JRVADHID UID: \?\SWD#MMDEVAPI#{0.0.0.00000000}.{660b9a66-71c6-4489-91a7-8571f1708737}#{e6327cad-dcec-4949-ae8a-991e976a79d2} Model Name: Unknown manufacturer
Device #2: Speakers (Realtek(R) Audio) Manufacturer: Realtek Model UID: HDAUDIO\FUNC_01&VEN_10EC&DEV_0257&SUBSYS_17AA3821&REV_1000 UID: \?\SWD#MMDEVAPI#{0.0.0.00000000}.{b573a1dc-31c0-4c03-9233-fa0eca2aded5}#{e6327cad-dcec-4949-ae8a-991e976a79d2} Model Name: Realtek(R) Audio
UPnP devices found : 2
Device #0OPPO BDP-103 UID: uuid:140479c0-58f3-1cef-84bf-70f11c50d603 Location: http://192.168.86.24:2870/dmr.xml
Device #1[TV] Samsung UID: uuid:0a21fe81-00aa-1000-adfd-0c891011c71e Location: http://192.168.86.20:7676/smp_19_
This can’t be an issue which is beyond you. It is particularly infuriating given the unrivaled sound quality Audirvana delivers, which has been significantly improved with Studio. Please give us our sanity back.
DG
I am having the exact same issue since I updated to 3.5.46.
Nothing else changed in my system.
Damien, are you planning on releasing a patch for this please?
I have experienced this issue and I note that you also have a Samsung TV (different model to me) which Audirvana ‘sees’ . It may be a complete ‘red herring’ but just wondered if there could be a connection though I don’t understand why that should be.
Try with very low buffer (1 minute). Tested yesterday with A 3.5. I don’t understand exactly why huge buffer and huge amount of RAM is necesary. Yesterday I set the buffer to 355 MB (1 minute at 384 kHz), unchecked “large WASAPI buffer”, “additional latency when changed spl rate” set to “none”. As a result (even on Qobuz) the track is buffered 1 minute when i press play, then after +/- 30 seconds buffered again and so on. No problems with SQ, skips or performance of playback. And I have a slow computer with an old SATA hdd…
UPDATE: I think I understand something. For some reasons with some tracks the software buffers just part of the track, even with enough RAM and buffer. This is the bug. For example if a track has 3:37 Audirvana loads only 3:32 and at 3:32 skip. But seems like with very low buffer, small than the entire track, the entire song is played without problems.
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