[alsa-devel] snd-usb-audio Buffer Sizes and Round Trip Latency
Hi,
I want to start a discussion regarding round trip latency for class compliant USB audio interfaces on Linux. In particular, I am noticing with my USB 2.0 RME Babyface Pro audio interface that the round trip latency is considerably higher on Linux than on macOS High Sierra and Windows 10.
I tested the round trip latency using a loopback audio cable and the ReaInsert plugin included with Reaper DAW (www.reaper.fm) that can be downloaded for Windows/macOS/Linux to calculate the additional delay.
Here are the results for 48000 Hz, 24-bit on my RME Babyface Pro: === block_size/periods block_size*periods + additional_delay ~ round_trip_latency round_trip_latency = (block_size*periods + additional_delay) / 48000 * 1000
Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): 16/2 32 + 80 ~ 2.333 ms 16/3 48 + 109 ~ 3.271 ms 32/2 64 + 129 ~ 4.021 ms 32/3 96 + 166 ~ 5.458 ms 64/2 128 + 205 ~ 6.938 ms 64/3 192 + 242 ~ 9.042 ms 128/2 256 + 352 ~ 12.667 ms 128/3 384 + 496 ~ 18.334 ms 256/2 512 + 650 ~ 24.208 ms 256/3 768 + 650 ~ 29.542 ms 512/2 1024 + 634 ~ 34.542 ms 512/3 1536 + 634 ~ 45.208 ms 1024/2 2048 + 650 ~ 56.208 ms 1024/3 3072 + 650 ~ 77.542 ms 2048/2 4096 + 633 ~ 98.521 ms 2048/3 6144 + 633 ~ 141.188 ms
macOS High Sierra, Class Compliant Mode (Apple Driver): 16/2 32 + 205 ~ 4.938 ms 32/2 64 + 205 ~ 5.604 ms 64/2 128 + 205 ~ 6.938 ms 128/2 256 + 205 ~ 9.604 ms 256/2 512 + 205 ~ 14.938 ms 512/2 1024 + 205 ~ 25.604 ms 1024/2 2048 + 205 ~ 46.938 ms 2048/2 4096 + 205 ~ 89.604 ms
macOS High Sierra, PC Mode (RME Driver v3.08): 16/2 32 + 59 ~ 1.896 ms 32/2 64 + 59 ~ 2.563 ms 64/2 128 + 59 ~ 3.896 ms 128/2 256 + 59 ~ 6.563 ms 256/2 512 + 59 ~ 11.596 ms 512/2 1024 + 59 ~ 22.563 ms 1024/2 2048 + 59 ~ 43.896 ms 2048/2 4096 + 59 ~ 86.563 ms
Windows 10, PC Mode (RME Driver 1.099): 48/2 96 + 63 ~ 3.313 ms 64/2 128 + 63 ~ 3.979 ms 96/2 192 + 63 ~ 5.313 ms 128/2 256 + 63 ~ 6.646 ms 256/2 512 + 63 ~ 11.979 ms 512/2 1024 + 63 ~ 22.646 ms 1024/2 2048 + 63 ~ 43.979 ms 2048/2 4096 + 63 ~ 86.646 ms ===
Some things in particular I noticed on Linux: - additional_delay varies a bit if I close and open the audio device again - additional_delay seems to increase as the block_size increases. I can make the additional_delay stay about the same rather than increasing by setting MAX_PACKS and MAX_PACKS_HS to 1 in sound/usb/card.h. In Linux versions before 3.13 there was a nrpacks parameter for snd-usb-audio to control this but it was removed with commit https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?h... - additional_delay is not constant as block_size is increased like on macOS and Windows
I made a patch to snd-usb-audio to expose the snd-usb-audio constants as runtime adjustable module parameters (/sys/module/snd_usb_audio/parameters/) for testing (takes effect when the device is disconnected+reconnected and logs the parameter values to dmesg): https://aur.archlinux.org/cgit/aur.git/plain/parameters.patch?h=snd-usb-audi...
The patch is used in my Arch Linux AUR package for convenience (using DKMS to avoid having to recompile entire kernel): https://aur.archlinux.org/packages/snd-usb-audio-lowlatency-dkms/
Can snd-usb-audio be improved so the additional_delay is always the same when closing/opening/reconfiguring the audio device and does not increase as the block_size increases?
I noticed using USB audio on Linux at lower latencies (block_size <= 128) is more prone to audio dropouts under load compared to macOS and Windows, even with CPU power management disabled (writing 0 to /dev/cpu_dma_latency). What can be done about this?
Thanks.
Regards, Jonathan
On 10/17/18 7:58 AM, Jonathan Liu wrote:
Hi,
I want to start a discussion regarding round trip latency for class compliant USB audio interfaces on Linux. In particular, I am noticing with my USB 2.0 RME Babyface Pro audio interface that the round trip latency is considerably higher on Linux than on macOS High Sierra and Windows 10.
I tested the round trip latency using a loopback audio cable and the ReaInsert plugin included with Reaper DAW (www.reaper.fm) that can be downloaded for Windows/macOS/Linux to calculate the additional delay.
Here are the results for 48000 Hz, 24-bit on my RME Babyface Pro:
block_size/periods block_size*periods + additional_delay ~ round_trip_latency round_trip_latency = (block_size*periods + additional_delay) / 48000 * 1000
Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): 16/2 32 + 80 ~ 2.333 ms 16/3 48 + 109 ~ 3.271 ms 32/2 64 + 129 ~ 4.021 ms 32/3 96 + 166 ~ 5.458 ms 64/2 128 + 205 ~ 6.938 ms 64/3 192 + 242 ~ 9.042 ms 128/2 256 + 352 ~ 12.667 ms 128/3 384 + 496 ~ 18.334 ms 256/2 512 + 650 ~ 24.208 ms 256/3 768 + 650 ~ 29.542 ms 512/2 1024 + 634 ~ 34.542 ms 512/3 1536 + 634 ~ 45.208 ms 1024/2 2048 + 650 ~ 56.208 ms 1024/3 3072 + 650 ~ 77.542 ms 2048/2 4096 + 633 ~ 98.521 ms 2048/3 6144 + 633 ~ 141.188 ms
macOS High Sierra, Class Compliant Mode (Apple Driver): 16/2 32 + 205 ~ 4.938 ms 32/2 64 + 205 ~ 5.604 ms 64/2 128 + 205 ~ 6.938 ms 128/2 256 + 205 ~ 9.604 ms 256/2 512 + 205 ~ 14.938 ms 512/2 1024 + 205 ~ 25.604 ms 1024/2 2048 + 205 ~ 46.938 ms 2048/2 4096 + 205 ~ 89.604 ms
I couldn't figure out how to analyze your data, not sure what the extra delays mean nor how you conclude that Linux is worse than MacOS or Windows10 for small buffers?
At any rate, I looked into this some time back but had to put the work on the back burner due to other priorities. What I do remember is that there is a built-in latency due to the fact that on playback the driver submits a number of zero-filled URBs and will only add valid audio data when the first URB is retired, which means you get a constant startup latency you will never be able to catch up.
I also vaguely remember that at some point the buffer/period sizes don't matter, each period will be broken up in a series of URBs and hence you will have more wake-ups than what is configured by the period size. In short I would look into the way the data is spread on multiple URBs and check how latency is impacted by the software design.
the last thing I have in mind is that for latency analysis and comparisons, using simple devices make sense. Latency can be affected by extra processing that might be enabled in the USB device depending on user configurations or parameters. Ideally to focus on the ALSA/xHCI interaction/latency we'd want to look at really dumb devices with just an input and output terminal and no processing.
-Pierre
macOS High Sierra, PC Mode (RME Driver v3.08): 16/2 32 + 59 ~ 1.896 ms 32/2 64 + 59 ~ 2.563 ms 64/2 128 + 59 ~ 3.896 ms 128/2 256 + 59 ~ 6.563 ms 256/2 512 + 59 ~ 11.596 ms 512/2 1024 + 59 ~ 22.563 ms 1024/2 2048 + 59 ~ 43.896 ms 2048/2 4096 + 59 ~ 86.563 ms
Windows 10, PC Mode (RME Driver 1.099): 48/2 96 + 63 ~ 3.313 ms 64/2 128 + 63 ~ 3.979 ms 96/2 192 + 63 ~ 5.313 ms 128/2 256 + 63 ~ 6.646 ms 256/2 512 + 63 ~ 11.979 ms 512/2 1024 + 63 ~ 22.646 ms 1024/2 2048 + 63 ~ 43.979 ms 2048/2 4096 + 63 ~ 86.646 ms ===
Some things in particular I noticed on Linux:
- additional_delay varies a bit if I close and open the audio device again
- additional_delay seems to increase as the block_size increases. I
can make the additional_delay stay about the same rather than increasing by setting MAX_PACKS and MAX_PACKS_HS to 1 in sound/usb/card.h. In Linux versions before 3.13 there was a nrpacks parameter for snd-usb-audio to control this but it was removed with commit https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?h...
- additional_delay is not constant as block_size is increased like on
macOS and Windows
I made a patch to snd-usb-audio to expose the snd-usb-audio constants as runtime adjustable module parameters (/sys/module/snd_usb_audio/parameters/) for testing (takes effect when the device is disconnected+reconnected and logs the parameter values to dmesg): https://aur.archlinux.org/cgit/aur.git/plain/parameters.patch?h=snd-usb-audi...
The patch is used in my Arch Linux AUR package for convenience (using DKMS to avoid having to recompile entire kernel): https://aur.archlinux.org/packages/snd-usb-audio-lowlatency-dkms/
Can snd-usb-audio be improved so the additional_delay is always the same when closing/opening/reconfiguring the audio device and does not increase as the block_size increases?
I noticed using USB audio on Linux at lower latencies (block_size <= 128) is more prone to audio dropouts under load compared to macOS and Windows, even with CPU power management disabled (writing 0 to /dev/cpu_dma_latency). What can be done about this?
Thanks.
Regards, Jonathan _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
On Mon, 22 Oct 2018, Pierre-Louis Bossart wrote:
On 10/17/18 7:58 AM, Jonathan Liu wrote:
Hi,
I want to start a discussion regarding round trip latency for class compliant USB audio interfaces on Linux. In particular, I am noticing with my USB 2.0 RME Babyface Pro audio interface that the round trip latency is considerably higher on Linux than on macOS High Sierra and Windows 10.
I tested the round trip latency using a loopback audio cable and the ReaInsert plugin included with Reaper DAW (www.reaper.fm) that can be downloaded for Windows/macOS/Linux to calculate the additional delay.
Here are the results for 48000 Hz, 24-bit on my RME Babyface Pro:
block_size/periods block_size*periods + additional_delay ~ round_trip_latency round_trip_latency = (block_size*periods + additional_delay) / 48000 * 1000
I'm with Pierre-Louis on this; I can't make heads or tails out of these formulas.
To begin with, I'm accustomed to talking about frames, periods, and buffers. What are "block"s? Are they the same as buffers?
What do these formulas mean? Is the first supposed to be a definition of round_trip_latency? If it isn't, then how do you define or measure round_trip_latency?
What is additional_delay? How is it measured or calculated?
Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): 16/2 32 + 80 ~ 2.333 ms
What are these numbers? Are these lines supposed to in the format expressed by the first formula above? If they are, how come "block_size/periods" shows up as a pair of numbers "16/2" but "block_size*periods" shows up as a single number "32"?
16/3 48 + 109 ~ 3.271 ms 32/2 64 + 129 ~ 4.021 ms 32/3 96 + 166 ~ 5.458 ms 64/2 128 + 205 ~ 6.938 ms 64/3 192 + 242 ~ 9.042 ms 128/2 256 + 352 ~ 12.667 ms 128/3 384 + 496 ~ 18.334 ms 256/2 512 + 650 ~ 24.208 ms 256/3 768 + 650 ~ 29.542 ms 512/2 1024 + 634 ~ 34.542 ms 512/3 1536 + 634 ~ 45.208 ms 1024/2 2048 + 650 ~ 56.208 ms 1024/3 3072 + 650 ~ 77.542 ms 2048/2 4096 + 633 ~ 98.521 ms 2048/3 6144 + 633 ~ 141.188 ms
macOS High Sierra, Class Compliant Mode (Apple Driver): 16/2 32 + 205 ~ 4.938 ms 32/2 64 + 205 ~ 5.604 ms 64/2 128 + 205 ~ 6.938 ms 128/2 256 + 205 ~ 9.604 ms 256/2 512 + 205 ~ 14.938 ms 512/2 1024 + 205 ~ 25.604 ms 1024/2 2048 + 205 ~ 46.938 ms 2048/2 4096 + 205 ~ 89.604 ms
What are the USB parameters for these tests? How many bytes/frame? What is the endpoint's maxpacket size? What is the speed of the USB bus?
I couldn't figure out how to analyze your data, not sure what the extra delays mean nor how you conclude that Linux is worse than MacOS or Windows10 for small buffers?
At any rate, I looked into this some time back but had to put the work on the back burner due to other priorities. What I do remember is that there is a built-in latency due to the fact that on playback the driver submits a number of zero-filled URBs and will only add valid audio data when the first URB is retired, which means you get a constant startup latency you will never be able to catch up.
In theory the number of zero-filled URBs could be reduced, maybe even eliminated.
I also vaguely remember that at some point the buffer/period sizes don't matter, each period will be broken up in a series of URBs and hence you will have more wake-ups than what is configured by the period size. In short I would look into the way the data is spread on multiple URBs and check how latency is impacted by the software design.
Agreed.
the last thing I have in mind is that for latency analysis and comparisons, using simple devices make sense. Latency can be affected by extra processing that might be enabled in the USB device depending on user configurations or parameters. Ideally to focus on the ALSA/xHCI interaction/latency we'd want to look at really dumb devices with just an input and output terminal and no processing.
-Pierre
macOS High Sierra, PC Mode (RME Driver v3.08): 16/2 32 + 59 ~ 1.896 ms 32/2 64 + 59 ~ 2.563 ms 64/2 128 + 59 ~ 3.896 ms 128/2 256 + 59 ~ 6.563 ms 256/2 512 + 59 ~ 11.596 ms 512/2 1024 + 59 ~ 22.563 ms 1024/2 2048 + 59 ~ 43.896 ms 2048/2 4096 + 59 ~ 86.563 ms
Windows 10, PC Mode (RME Driver 1.099): 48/2 96 + 63 ~ 3.313 ms 64/2 128 + 63 ~ 3.979 ms 96/2 192 + 63 ~ 5.313 ms 128/2 256 + 63 ~ 6.646 ms 256/2 512 + 63 ~ 11.979 ms 512/2 1024 + 63 ~ 22.646 ms 1024/2 2048 + 63 ~ 43.979 ms 2048/2 4096 + 63 ~ 86.646 ms ===
Some things in particular I noticed on Linux:
- additional_delay varies a bit if I close and open the audio device again
- additional_delay seems to increase as the block_size increases. I
can make the additional_delay stay about the same rather than increasing by setting MAX_PACKS and MAX_PACKS_HS to 1 in sound/usb/card.h. In Linux versions before 3.13 there was a nrpacks parameter for snd-usb-audio to control this but it was removed with commit https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?h...
- additional_delay is not constant as block_size is increased like on
macOS and Windows
Perhaps this additional_delay is caused by the zero-filled URBs mentioned earlier.
We can't say anything about the effect of setting MAX_PACKS to 1 without knowing how the driver is currently fitting packets into frames and URBs. In any case, you should be able to reduce the number of packets in each URB simply by reducing the period size, since the driver strives to keep each URB not much larger than a period (as I recall -- it's been a long time since I worked on this (2013)).
I made a patch to snd-usb-audio to expose the snd-usb-audio constants as runtime adjustable module parameters (/sys/module/snd_usb_audio/parameters/) for testing (takes effect when the device is disconnected+reconnected and logs the parameter values to dmesg): https://aur.archlinux.org/cgit/aur.git/plain/parameters.patch?h=snd-usb-audi...
The patch is used in my Arch Linux AUR package for convenience (using DKMS to avoid having to recompile entire kernel): https://aur.archlinux.org/packages/snd-usb-audio-lowlatency-dkms/
Can snd-usb-audio be improved so the additional_delay is always the same when closing/opening/reconfiguring the audio device and does not increase as the block_size increases?
I noticed using USB audio on Linux at lower latencies (block_size <= 128) is more prone to audio dropouts under load compared to macOS and Windows, even with CPU power management disabled (writing 0 to /dev/cpu_dma_latency). What can be done about this?
You can reduce the CPU load. :-)
Seriously, how can you compare loads between different operating systems?
Also, note the Linux's scheduler has a number of adjustable parameters, which I am not familiar with.
Alan Stern
Hi,
On Tue, 23 Oct 2018 at 02:40, Alan Stern stern@rowland.harvard.edu wrote:
On Mon, 22 Oct 2018, Pierre-Louis Bossart wrote:
On 10/17/18 7:58 AM, Jonathan Liu wrote:
Hi,
I want to start a discussion regarding round trip latency for class compliant USB audio interfaces on Linux. In particular, I am noticing with my USB 2.0 RME Babyface Pro audio interface that the round trip latency is considerably higher on Linux than on macOS High Sierra and Windows 10.
I tested the round trip latency using a loopback audio cable and the ReaInsert plugin included with Reaper DAW (www.reaper.fm) that can be downloaded for Windows/macOS/Linux to calculate the additional delay.
Here are the results for 48000 Hz, 24-bit on my RME Babyface Pro:
block_size/periods block_size*periods + additional_delay ~ round_trip_latency round_trip_latency = (block_size*periods + additional_delay) / 48000 * 1000
I'm with Pierre-Louis on this; I can't make heads or tails out of these formulas.
To begin with, I'm accustomed to talking about frames, periods, and buffers. What are "block"s? Are they the same as buffers?
What do these formulas mean? Is the first supposed to be a definition of round_trip_latency? If it isn't, then how do you define or measure round_trip_latency?
What is additional_delay? How is it measured or calculated?
See below.
Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): 16/2 32 + 80 ~ 2.333 ms
What are these numbers? Are these lines supposed to in the format expressed by the first formula above? If they are, how come "block_size/periods" shows up as a pair of numbers "16/2" but "block_size*periods" shows up as a single number "32"?
To interpret "16/2 32 + 80 ~ 2.333 ms" Block size: 16 samples Periods: 2 (one period for playback + one period for recording when determining round trip latency) The minimum round trip latency is: 16 * 2 = 32 samples However, I measured 112 samples round trip latency which is an additional delay of 80 samples (32 + 80 = 112). 112 samples at 48000 Hz is 112 / 48000 * 1000 is approximately 2.333 ms measured round trip latency.
16/3 48 + 109 ~ 3.271 ms 32/2 64 + 129 ~ 4.021 ms 32/3 96 + 166 ~ 5.458 ms 64/2 128 + 205 ~ 6.938 ms 64/3 192 + 242 ~ 9.042 ms 128/2 256 + 352 ~ 12.667 ms 128/3 384 + 496 ~ 18.334 ms 256/2 512 + 650 ~ 24.208 ms 256/3 768 + 650 ~ 29.542 ms 512/2 1024 + 634 ~ 34.542 ms 512/3 1536 + 634 ~ 45.208 ms 1024/2 2048 + 650 ~ 56.208 ms 1024/3 3072 + 650 ~ 77.542 ms 2048/2 4096 + 633 ~ 98.521 ms 2048/3 6144 + 633 ~ 141.188 ms
macOS High Sierra, Class Compliant Mode (Apple Driver): 16/2 32 + 205 ~ 4.938 ms 32/2 64 + 205 ~ 5.604 ms 64/2 128 + 205 ~ 6.938 ms 128/2 256 + 205 ~ 9.604 ms 256/2 512 + 205 ~ 14.938 ms 512/2 1024 + 205 ~ 25.604 ms 1024/2 2048 + 205 ~ 46.938 ms 2048/2 4096 + 205 ~ 89.604 ms
What are the USB parameters for these tests? How many bytes/frame? What is the endpoint's maxpacket size? What is the speed of the USB bus?
How would I determine the USB parameters and bytes/frame?
USB port is Intel USB 3.0 port. Device is USB 2.0 high speed (480 Mbps).
Here is the lsusb output: Bus 001 Device 004: ID 2a39:3fb0 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x2a39 idProduct 0x3fb0 bcdDevice 0.01 iManufacturer 1 RME iProduct 2 Babyface Pro (71964099) iSerial 3 EF72ADBCCECA4C8 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x01a7 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 100mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 4 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 8 wTotalLength 0x0055 bmControls 0x00 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 1 bmAttributes 3 Internal programmable clock bmControls 0x03 Clock Frequency Control (read/write) bAssocTerminal 0 iClockSource 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 1 bNrChannels 12 bmChannelConfig 0x00000000 iChannelNames 0 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 1 bNrChannels 12 bmChannelConfig 0x00000000 iChannelNames 0 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 2 bCSourceID 1 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 5 bCSourceID 1 bmControls 0x0000 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000003 Front Left (FL) Front Right (FR) iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0096 1x 150 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0004 1x 4 bytes bInterval 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 12 bmChannelConfig 0x00000000 iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0384 1x 900 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0004 1x 4 bytes bInterval 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 6 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 12 bmChannelConfig 0x00000000 iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0384 1x 900 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 6 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000003 Front Left (FL) Front Right (FR) iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0096 1x 150 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 2 Babyface Pro (71964099) MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0061 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 4 Port 1 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 4 Port 1 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 7 bNrInputPins 1 baSourceID( 0) 6 BaSourcePin( 0) 1 iJack 5 Port 2 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 6 iJack 5 Port 2 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 4 Port 1 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 4 Port 1 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 5 iJack 5 Port 2 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 8 bNrInputPins 1 baSourceID( 0) 5 BaSourcePin( 0) 1 iJack 5 Port 2 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x07 EP 7 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 6 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 2 baAssocJackID( 0) 1 baAssocJackID( 1) 5 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 6 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 2 baAssocJackID( 0) 3 baAssocJackID( 1) 7 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 bNumConfigurations 0 Device Status: 0x0e00 (Bus Powered)
I couldn't figure out how to analyze your data, not sure what the extra delays mean nor how you conclude that Linux is worse than MacOS or Windows10 for small buffers?
At any rate, I looked into this some time back but had to put the work on the back burner due to other priorities. What I do remember is that there is a built-in latency due to the fact that on playback the driver submits a number of zero-filled URBs and will only add valid audio data when the first URB is retired, which means you get a constant startup latency you will never be able to catch up.
In theory the number of zero-filled URBs could be reduced, maybe even eliminated.
I also vaguely remember that at some point the buffer/period sizes don't matter, each period will be broken up in a series of URBs and hence you will have more wake-ups than what is configured by the period size. In short I would look into the way the data is spread on multiple URBs and check how latency is impacted by the software design.
Agreed.
the last thing I have in mind is that for latency analysis and comparisons, using simple devices make sense. Latency can be affected by extra processing that might be enabled in the USB device depending on user configurations or parameters. Ideally to focus on the ALSA/xHCI interaction/latency we'd want to look at really dumb devices with just an input and output terminal and no processing.
-Pierre
macOS High Sierra, PC Mode (RME Driver v3.08): 16/2 32 + 59 ~ 1.896 ms 32/2 64 + 59 ~ 2.563 ms 64/2 128 + 59 ~ 3.896 ms 128/2 256 + 59 ~ 6.563 ms 256/2 512 + 59 ~ 11.596 ms 512/2 1024 + 59 ~ 22.563 ms 1024/2 2048 + 59 ~ 43.896 ms 2048/2 4096 + 59 ~ 86.563 ms
Windows 10, PC Mode (RME Driver 1.099): 48/2 96 + 63 ~ 3.313 ms 64/2 128 + 63 ~ 3.979 ms 96/2 192 + 63 ~ 5.313 ms 128/2 256 + 63 ~ 6.646 ms 256/2 512 + 63 ~ 11.979 ms 512/2 1024 + 63 ~ 22.646 ms 1024/2 2048 + 63 ~ 43.979 ms 2048/2 4096 + 63 ~ 86.646 ms ===
Some things in particular I noticed on Linux:
- additional_delay varies a bit if I close and open the audio device again
- additional_delay seems to increase as the block_size increases. I
can make the additional_delay stay about the same rather than increasing by setting MAX_PACKS and MAX_PACKS_HS to 1 in sound/usb/card.h. In Linux versions before 3.13 there was a nrpacks parameter for snd-usb-audio to control this but it was removed with commit https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?h...
- additional_delay is not constant as block_size is increased like on
macOS and Windows
Perhaps this additional_delay is caused by the zero-filled URBs mentioned earlier.
We can't say anything about the effect of setting MAX_PACKS to 1 without knowing how the driver is currently fitting packets into frames and URBs. In any case, you should be able to reduce the number of packets in each URB simply by reducing the period size, since the driver strives to keep each URB not much larger than a period (as I recall -- it's been a long time since I worked on this (2013)).
I made a patch to snd-usb-audio to expose the snd-usb-audio constants as runtime adjustable module parameters (/sys/module/snd_usb_audio/parameters/) for testing (takes effect when the device is disconnected+reconnected and logs the parameter values to dmesg): https://aur.archlinux.org/cgit/aur.git/plain/parameters.patch?h=snd-usb-audi...
The patch is used in my Arch Linux AUR package for convenience (using DKMS to avoid having to recompile entire kernel): https://aur.archlinux.org/packages/snd-usb-audio-lowlatency-dkms/
Can snd-usb-audio be improved so the additional_delay is always the same when closing/opening/reconfiguring the audio device and does not increase as the block_size increases?
I noticed using USB audio on Linux at lower latencies (block_size <= 128) is more prone to audio dropouts under load compared to macOS and Windows, even with CPU power management disabled (writing 0 to /dev/cpu_dma_latency). What can be done about this?
You can reduce the CPU load. :-)
Seriously, how can you compare loads between different operating systems?
Also, note the Linux's scheduler has a number of adjustable parameters, which I am not familiar with.
Alan Stern
Regards, Jonathan
Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): 16/2 32 + 80 ~ 2.333 ms
What are these numbers? Are these lines supposed to in the format expressed by the first formula above? If they are, how come "block_size/periods" shows up as a pair of numbers "16/2" but "block_size*periods" shows up as a single number "32"?
To interpret "16/2 32 + 80 ~ 2.333 ms" Block size: 16 samples Periods: 2 (one period for playback + one period for recording when determining round trip latency) The minimum round trip latency is: 16 * 2 = 32 samples However, I measured 112 samples round trip latency which is an additional delay of 80 samples (32 + 80 = 112). 112 samples at 48000 Hz is 112 / 48000 * 1000 is approximately 2.333 ms measured round trip latency.
ok, so what problem are you trying to fix?
Are you concerned about the latency numbers (but then they seem lower on Linux and latency concerns with large buffers are a self-negating proposition)? are you concerned about the variable delay that doesn't seem to exist on MacOS or Windows? Are you trying to match the performance of the RME driver on MacOS?
I am not sure how this comparison is done btw, the delay includes both buffering on the device side before reaching the analog parts as well as buffering on the OS side. While the former should be constant, the latter depends a great deal on implementation, not sure there are direct lessons to be applied to ALSA. I also see inconsistent/non-linear results where with a larger block size the delay is smaller, e.g.
256/2 512 + 650 ~ 24.208 ms 2048/3 6144 + 633 ~ 141.188 ms
On Tue, 23 Oct 2018 16:08:22 +0200, Pierre-Louis Bossart wrote:
Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): 16/2 32 + 80 ~ 2.333 ms
What are these numbers? Are these lines supposed to in the format expressed by the first formula above? If they are, how come "block_size/periods" shows up as a pair of numbers "16/2" but "block_size*periods" shows up as a single number "32"?
To interpret "16/2 32 + 80 ~ 2.333 ms" Block size: 16 samples Periods: 2 (one period for playback + one period for recording when determining round trip latency) The minimum round trip latency is: 16 * 2 = 32 samples However, I measured 112 samples round trip latency which is an additional delay of 80 samples (32 + 80 = 112). 112 samples at 48000 Hz is 112 / 48000 * 1000 is approximately 2.333 ms measured round trip latency.
ok, so what problem are you trying to fix?
Are you concerned about the latency numbers (but then they seem lower on Linux and latency concerns with large buffers are a self-negating proposition)? are you concerned about the variable delay that doesn't seem to exist on MacOS or Windows? Are you trying to match the performance of the RME driver on MacOS?
I am not sure how this comparison is done btw, the delay includes both buffering on the device side before reaching the analog parts as well as buffering on the OS side. While the former should be constant, the latter depends a great deal on implementation, not sure there are direct lessons to be applied to ALSA. I also see inconsistent/non-linear results where with a larger block size the delay is smaller, e.g.
256/2 512 + 650 ~ 24.208 ms 2048/3 6144 + 633 ~ 141.188 ms
Independently from the measurement done in this thread, actually, there is a known latency source in the playback path in USB-audio driver code -- which I mentioned in the audio mini conf in the last year: namely, the USB-audio driver starts streaming at prepare time for playback, not at the trigger-START time. This is a sort of workaround to make the device looking similar to the standard ring-buffer behavior.
Maybe moving the start at trigger (like the capture direction) would reduce this artificial latency, but it makes the driver behaving in an unexpected manner. Then it may wake up for period_elapsed soon after the stream start with a large runtime->delay value, as the data in in-flight URBs are seen as already "processed".
thanks,
Takashi
On Wed, 24 Oct 2018 at 18:13, Takashi Iwai tiwai@suse.de wrote:
On Tue, 23 Oct 2018 16:08:22 +0200, Pierre-Louis Bossart wrote:
Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): 16/2 32 + 80 ~ 2.333 ms
What are these numbers? Are these lines supposed to in the format expressed by the first formula above? If they are, how come "block_size/periods" shows up as a pair of numbers "16/2" but "block_size*periods" shows up as a single number "32"?
To interpret "16/2 32 + 80 ~ 2.333 ms" Block size: 16 samples Periods: 2 (one period for playback + one period for recording when determining round trip latency) The minimum round trip latency is: 16 * 2 = 32 samples However, I measured 112 samples round trip latency which is an additional delay of 80 samples (32 + 80 = 112). 112 samples at 48000 Hz is 112 / 48000 * 1000 is approximately 2.333 ms measured round trip latency.
ok, so what problem are you trying to fix?
Are you concerned about the latency numbers (but then they seem lower on Linux and latency concerns with large buffers are a self-negating proposition)? are you concerned about the variable delay that doesn't seem to exist on MacOS or Windows? Are you trying to match the performance of the RME driver on MacOS?
I am not sure how this comparison is done btw, the delay includes both buffering on the device side before reaching the analog parts as well as buffering on the OS side. While the former should be constant, the latter depends a great deal on implementation, not sure there are direct lessons to be applied to ALSA. I also see inconsistent/non-linear results where with a larger block size the delay is smaller, e.g.
256/2 512 + 650 ~ 24.208 ms 2048/3 6144 + 633 ~ 141.188 ms
Independently from the measurement done in this thread, actually, there is a known latency source in the playback path in USB-audio driver code -- which I mentioned in the audio mini conf in the last year: namely, the USB-audio driver starts streaming at prepare time for playback, not at the trigger-START time. This is a sort of workaround to make the device looking similar to the standard ring-buffer behavior.
Maybe moving the start at trigger (like the capture direction) would reduce this artificial latency, but it makes the driver behaving in an unexpected manner. Then it may wake up for period_elapsed soon after the stream start with a large runtime->delay value, as the data in in-flight URBs are seen as already "processed".
I observed that snd_usb_pcm_prepare calls start_endpoints which ends up submitting silent urbs (prepared by prepare_silent_urb) until ep->prepare_data_urb is set by SNDRV_PCM_TRIGGER_START in snd_usb_substream_playback_trigger.
I tried to moving the start_endpoints call from snd_usb_pcm_prepare to snd_usb_substream_playback trigger's SNDRV_PCM_TRIGGER_START case (see https://github.com/net147/linux/commit/276eae5481653a2d4034fbae56f0d5bc579ec... - it is enabled using start_playback_on_prepare=0 module option for snd-usb-audio) but I get a kernel stall in some cases with the following call trace: _raw_spin_lock+0x2c/0x30 _snd_pcm_stream_lock_irqsave+0x31/0x60 [snd_pcm] snd_pcm_period_elapsed+0x26/0xb0 [snd_pcm] prepare_playback_urb+0x368/0x640 [snd_usb_audio] ? usb_submit_urb+0x3cb/0x590 snd_usb_endpoint_start+0x148/0x300 [snd_usb_audio] start_endpoints+0x36/0x160 [snd_usb_audio] snd_usb_substream_playback_trigger+0x152/0x1a0 [snd_usb_audio] snd_pcm_action+0x117/0x150 [snd_pcm] snd_pcm_common_ioctl+0x588/0xdb0 [snd_pcm] ? mprotect_fixup+0x1ec/0x2f0 snd_pcm_ioctl+0x23/0x30 [snd_pcm] do_vfs_ioctl+0xa6/0x760 ? syscall_trace_enter+0x1be/0x2b0 __x64_sys_ioctl+0x62/0x90 do_syscall_64+0x5b/0x170 entry_SYSCALL_64_after_hwframe+0x44/0xa9
Any ideas?
Thanks.
Regards, Jonathan
On Mon, 22 Apr 2019 12:50:15 +0200, Jonathan Liu wrote:
On Wed, 24 Oct 2018 at 18:13, Takashi Iwai tiwai@suse.de wrote:
On Tue, 23 Oct 2018 16:08:22 +0200, Pierre-Louis Bossart wrote:
> Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): > 16/2 32 + 80 ~ 2.333 ms
What are these numbers? Are these lines supposed to in the format expressed by the first formula above? If they are, how come "block_size/periods" shows up as a pair of numbers "16/2" but "block_size*periods" shows up as a single number "32"?
To interpret "16/2 32 + 80 ~ 2.333 ms" Block size: 16 samples Periods: 2 (one period for playback + one period for recording when determining round trip latency) The minimum round trip latency is: 16 * 2 = 32 samples However, I measured 112 samples round trip latency which is an additional delay of 80 samples (32 + 80 = 112). 112 samples at 48000 Hz is 112 / 48000 * 1000 is approximately 2.333 ms measured round trip latency.
ok, so what problem are you trying to fix?
Are you concerned about the latency numbers (but then they seem lower on Linux and latency concerns with large buffers are a self-negating proposition)? are you concerned about the variable delay that doesn't seem to exist on MacOS or Windows? Are you trying to match the performance of the RME driver on MacOS?
I am not sure how this comparison is done btw, the delay includes both buffering on the device side before reaching the analog parts as well as buffering on the OS side. While the former should be constant, the latter depends a great deal on implementation, not sure there are direct lessons to be applied to ALSA. I also see inconsistent/non-linear results where with a larger block size the delay is smaller, e.g.
256/2 512 + 650 ~ 24.208 ms 2048/3 6144 + 633 ~ 141.188 ms
Independently from the measurement done in this thread, actually, there is a known latency source in the playback path in USB-audio driver code -- which I mentioned in the audio mini conf in the last year: namely, the USB-audio driver starts streaming at prepare time for playback, not at the trigger-START time. This is a sort of workaround to make the device looking similar to the standard ring-buffer behavior.
Maybe moving the start at trigger (like the capture direction) would reduce this artificial latency, but it makes the driver behaving in an unexpected manner. Then it may wake up for period_elapsed soon after the stream start with a large runtime->delay value, as the data in in-flight URBs are seen as already "processed".
I observed that snd_usb_pcm_prepare calls start_endpoints which ends up submitting silent urbs (prepared by prepare_silent_urb) until ep->prepare_data_urb is set by SNDRV_PCM_TRIGGER_START in snd_usb_substream_playback_trigger.
I tried to moving the start_endpoints call from snd_usb_pcm_prepare to snd_usb_substream_playback trigger's SNDRV_PCM_TRIGGER_START case (see https://github.com/net147/linux/commit/276eae5481653a2d4034fbae56f0d5bc579ec...
- it is enabled using start_playback_on_prepare=0 module option for
snd-usb-audio) but I get a kernel stall in some cases with the following call trace: _raw_spin_lock+0x2c/0x30 _snd_pcm_stream_lock_irqsave+0x31/0x60 [snd_pcm] snd_pcm_period_elapsed+0x26/0xb0 [snd_pcm] prepare_playback_urb+0x368/0x640 [snd_usb_audio] ? usb_submit_urb+0x3cb/0x590 snd_usb_endpoint_start+0x148/0x300 [snd_usb_audio] start_endpoints+0x36/0x160 [snd_usb_audio] snd_usb_substream_playback_trigger+0x152/0x1a0 [snd_usb_audio] snd_pcm_action+0x117/0x150 [snd_pcm] snd_pcm_common_ioctl+0x588/0xdb0 [snd_pcm] ? mprotect_fixup+0x1ec/0x2f0 snd_pcm_ioctl+0x23/0x30 [snd_pcm] do_vfs_ioctl+0xa6/0x760 ? syscall_trace_enter+0x1be/0x2b0 __x64_sys_ioctl+0x62/0x90 do_syscall_64+0x5b/0x170 entry_SYSCALL_64_after_hwframe+0x44/0xa9
Any ideas?
This is because snd_pcm_period_elapsed() is called from prepare_data_urb callback that is called also at start_endpoints().
I guess we'd need to move the hwptr accounting and snd_pcm_period_elapsed() call into retire_data_urb callback in the case of start-at-trigger for playback.
thanks,
Takashi
On Wed, 24 Apr 2019 16:05:53 +0200, Takashi Iwai wrote:
On Mon, 22 Apr 2019 12:50:15 +0200, Jonathan Liu wrote:
On Wed, 24 Oct 2018 at 18:13, Takashi Iwai tiwai@suse.de wrote:
On Tue, 23 Oct 2018 16:08:22 +0200, Pierre-Louis Bossart wrote:
>> Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): >> 16/2 32 + 80 ~ 2.333 ms What are these numbers? Are these lines supposed to in the format expressed by the first formula above? If they are, how come "block_size/periods" shows up as a pair of numbers "16/2" but "block_size*periods" shows up as a single number "32"?
To interpret "16/2 32 + 80 ~ 2.333 ms" Block size: 16 samples Periods: 2 (one period for playback + one period for recording when determining round trip latency) The minimum round trip latency is: 16 * 2 = 32 samples However, I measured 112 samples round trip latency which is an additional delay of 80 samples (32 + 80 = 112). 112 samples at 48000 Hz is 112 / 48000 * 1000 is approximately 2.333 ms measured round trip latency.
ok, so what problem are you trying to fix?
Are you concerned about the latency numbers (but then they seem lower on Linux and latency concerns with large buffers are a self-negating proposition)? are you concerned about the variable delay that doesn't seem to exist on MacOS or Windows? Are you trying to match the performance of the RME driver on MacOS?
I am not sure how this comparison is done btw, the delay includes both buffering on the device side before reaching the analog parts as well as buffering on the OS side. While the former should be constant, the latter depends a great deal on implementation, not sure there are direct lessons to be applied to ALSA. I also see inconsistent/non-linear results where with a larger block size the delay is smaller, e.g.
256/2 512 + 650 ~ 24.208 ms 2048/3 6144 + 633 ~ 141.188 ms
Independently from the measurement done in this thread, actually, there is a known latency source in the playback path in USB-audio driver code -- which I mentioned in the audio mini conf in the last year: namely, the USB-audio driver starts streaming at prepare time for playback, not at the trigger-START time. This is a sort of workaround to make the device looking similar to the standard ring-buffer behavior.
Maybe moving the start at trigger (like the capture direction) would reduce this artificial latency, but it makes the driver behaving in an unexpected manner. Then it may wake up for period_elapsed soon after the stream start with a large runtime->delay value, as the data in in-flight URBs are seen as already "processed".
I observed that snd_usb_pcm_prepare calls start_endpoints which ends up submitting silent urbs (prepared by prepare_silent_urb) until ep->prepare_data_urb is set by SNDRV_PCM_TRIGGER_START in snd_usb_substream_playback_trigger.
I tried to moving the start_endpoints call from snd_usb_pcm_prepare to snd_usb_substream_playback trigger's SNDRV_PCM_TRIGGER_START case (see https://github.com/net147/linux/commit/276eae5481653a2d4034fbae56f0d5bc579ec...
- it is enabled using start_playback_on_prepare=0 module option for
snd-usb-audio) but I get a kernel stall in some cases with the following call trace: _raw_spin_lock+0x2c/0x30 _snd_pcm_stream_lock_irqsave+0x31/0x60 [snd_pcm] snd_pcm_period_elapsed+0x26/0xb0 [snd_pcm] prepare_playback_urb+0x368/0x640 [snd_usb_audio] ? usb_submit_urb+0x3cb/0x590 snd_usb_endpoint_start+0x148/0x300 [snd_usb_audio] start_endpoints+0x36/0x160 [snd_usb_audio] snd_usb_substream_playback_trigger+0x152/0x1a0 [snd_usb_audio] snd_pcm_action+0x117/0x150 [snd_pcm] snd_pcm_common_ioctl+0x588/0xdb0 [snd_pcm] ? mprotect_fixup+0x1ec/0x2f0 snd_pcm_ioctl+0x23/0x30 [snd_pcm] do_vfs_ioctl+0xa6/0x760 ? syscall_trace_enter+0x1be/0x2b0 __x64_sys_ioctl+0x62/0x90 do_syscall_64+0x5b/0x170 entry_SYSCALL_64_after_hwframe+0x44/0xa9
Any ideas?
This is because snd_pcm_period_elapsed() is called from prepare_data_urb callback that is called also at start_endpoints().
I guess we'd need to move the hwptr accounting and snd_pcm_period_elapsed() call into retire_data_urb callback in the case of start-at-trigger for playback.
I meant something like below. Only lightly tested.
Takashi
--- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -84,6 +84,7 @@ static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ static bool ignore_ctl_error; static bool autoclock = true; +static bool lowlatency; static char *quirk_alias[SNDRV_CARDS];
bool snd_usb_use_vmalloc = true; @@ -105,6 +106,8 @@ MODULE_PARM_DESC(ignore_ctl_error, "Ignore errors from USB controller for mixer interfaces."); module_param(autoclock, bool, 0444); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); +module_param(lowlatency, bool, 0444); +MODULE_PARM_DESC(lowlatency, "Low latency playback"); module_param_array(quirk_alias, charp, NULL, 0444); MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444); @@ -487,6 +490,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, chip->card = card; chip->setup = device_setup[idx]; chip->autoclock = autoclock; + chip->lowlatency = lowlatency; atomic_set(&chip->active, 1); /* avoid autopm during probing */ atomic_set(&chip->usage_count, 0); atomic_set(&chip->shutdown, 0); diff --git a/sound/usb/card.h b/sound/usb/card.h index 79fa2a19fb7b..244c80ff8e33 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -48,6 +48,7 @@ struct snd_urb_ctx { int index; /* index for urb array */ int packets; /* number of packets per urb */ int packet_size[MAX_PACKS_HS]; /* size of packets for next submission */ + bool period_elapsed; struct list_head ready_list; };
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 056af0a57b22..165bf2de6a37 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -927,7 +927,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
/* for playback, submit the URBs now; otherwise, the first hwptr_done * updates for all URBs would happen at the same time when starting */ - if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK && + !subs->stream->chip->lowlatency) ret = start_endpoints(subs);
unlock: @@ -1542,7 +1543,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, struct snd_usb_endpoint *ep = subs->data_endpoint; struct snd_urb_ctx *ctx = urb->context; unsigned int counts, frames, bytes; - int i, stride, period_elapsed = 0; + int i, stride; unsigned long flags;
stride = runtime->frame_bits >> 3; @@ -1551,6 +1552,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); subs->frame_limit += ep->max_urb_frames; + ctx->period_elapsed = 0; for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; @@ -1566,7 +1568,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; subs->frame_limit = 0; - period_elapsed = 1; + ctx->period_elapsed = 1; if (subs->fmt_type == UAC_FORMAT_TYPE_II) { if (subs->transfer_done > 0) { /* FIXME: fill-max mode is not @@ -1589,7 +1591,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, } } /* finish at the period boundary or after enough frames */ - if ((period_elapsed || + if ((ctx->period_elapsed || subs->transfer_done >= subs->frame_limit) && !snd_usb_endpoint_implicit_feedback_sink(ep)) break; @@ -1640,7 +1642,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = bytes; - if (period_elapsed) + if (!subs->stream->chip->lowlatency && ctx->period_elapsed) snd_pcm_period_elapsed(subs->pcm_substream); }
@@ -1654,6 +1656,7 @@ static void retire_playback_urb(struct snd_usb_substream *subs, unsigned long flags; struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; struct snd_usb_endpoint *ep = subs->data_endpoint; + struct snd_urb_ctx *ctx = urb->context; int processed = urb->transfer_buffer_length / ep->stride; int est_delay;
@@ -1695,12 +1698,16 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
out: spin_unlock_irqrestore(&subs->lock, flags); + + if (subs->stream->chip->lowlatency && ctx->period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); }
static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_usb_substream *subs = substream->runtime->private_data; + int err;
switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -1709,6 +1716,14 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: subs->data_endpoint->prepare_data_urb = prepare_playback_urb; subs->data_endpoint->retire_data_urb = retire_playback_urb; + if (subs->stream->chip->lowlatency) { + err = start_endpoints(subs); + if (err < 0) { + subs->data_endpoint->prepare_data_urb = NULL; + subs->data_endpoint->retire_data_urb = NULL; + return err; + } + } subs->running = 1; return 0; case SNDRV_PCM_TRIGGER_STOP: diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b9faeca645fd..71bc58b11ca0 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -64,6 +64,7 @@ struct snd_usb_audio { bool keep_iface; /* keep interface/altset after closing * or parameter change */ + bool lowlatency;
struct usb_host_interface *ctrl_intf; /* the audio control interface */ };
On Tue, 23 Oct 2018, Jonathan Liu wrote:
Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): 16/2 32 + 80 ~ 2.333 ms
What are these numbers? Are these lines supposed to in the format expressed by the first formula above? If they are, how come "block_size/periods" shows up as a pair of numbers "16/2" but "block_size*periods" shows up as a single number "32"?
To interpret "16/2 32 + 80 ~ 2.333 ms" Block size: 16 samples
Is this what ALSA would call the number of frames per period? I presume your sample is the same as an ALSA frame. (As I recall, in ALSA each frame in a stereo stream contains two samples. You _are_ using stereo, right? And each sample would be 3 bytes for 24-bit audio. Also, in ALSA the period size and block size are the sizes in bytes, not in frames.)
Periods: 2 (one period for playback + one period for recording when determining round trip latency)
In other words, one period per block in each direction?
The minimum round trip latency is: 16 * 2 = 32 samples However, I measured 112 samples round trip latency which is an additional delay of 80 samples (32 + 80 = 112). 112 samples at 48000 Hz is 112 / 48000 * 1000 is approximately 2.333 ms measured round trip latency.
16/3 48 + 109 ~ 3.271 ms
Presumably this indicates three periods, then. Is that two in the outward direction and one in the inward direction, or vice versa?
32/2 64 + 129 ~ 4.021 ms 32/3 96 + 166 ~ 5.458 ms 64/2 128 + 205 ~ 6.938 ms 64/3 192 + 242 ~ 9.042 ms 128/2 256 + 352 ~ 12.667 ms 128/3 384 + 496 ~ 18.334 ms 256/2 512 + 650 ~ 24.208 ms 256/3 768 + 650 ~ 29.542 ms 512/2 1024 + 634 ~ 34.542 ms 512/3 1536 + 634 ~ 45.208 ms 1024/2 2048 + 650 ~ 56.208 ms 1024/3 3072 + 650 ~ 77.542 ms 2048/2 4096 + 633 ~ 98.521 ms 2048/3 6144 + 633 ~ 141.188 ms
As compared to the other systems, it appears that in Linux the additional delay increases with the period size. This could be a result of the initial zero-filled URBs, since the size or number of those URBs may depend on the other settings.
macOS High Sierra, Class Compliant Mode (Apple Driver): 16/2 32 + 205 ~ 4.938 ms 32/2 64 + 205 ~ 5.604 ms 64/2 128 + 205 ~ 6.938 ms 128/2 256 + 205 ~ 9.604 ms 256/2 512 + 205 ~ 14.938 ms 512/2 1024 + 205 ~ 25.604 ms 1024/2 2048 + 205 ~ 46.938 ms 2048/2 4096 + 205 ~ 89.604 ms
What are the USB parameters for these tests? How many bytes/frame? What is the endpoint's maxpacket size? What is the speed of the USB bus?
How would I determine the USB parameters and bytes/frame?
USB port is Intel USB 3.0 port. Device is USB 2.0 high speed (480 Mbps).
Here is the lsusb output:
Both too much information and too little. Instead, let's see the device's entry in /sys/kernel/debug/usb/devices, copied at a time while the test is running. That will omit a lot of irrelevant information and will indicate which of all the possible device settings is the one actually in use.
If you want to get a better idea for exactly what is happening at the USB level, you can collect a usbmon trace while running a test. Also, it wouldn't hurt to see the values of max_packs_per_urb, urb_packs, max_packs_per_period, urbs_per_period, ep->max_urb_frames, and ep->nurbs from data_ep_set_params() in the audio driver.
Alan Stern
Hi,
On Wed, 24 Oct 2018 at 02:10, Alan Stern stern@rowland.harvard.edu wrote:
On Tue, 23 Oct 2018, Jonathan Liu wrote:
Linux 4.17.14, Class Compliant Mode (snd-usb-audio, ALSA backend): 16/2 32 + 80 ~ 2.333 ms
What are these numbers? Are these lines supposed to in the format expressed by the first formula above? If they are, how come "block_size/periods" shows up as a pair of numbers "16/2" but "block_size*periods" shows up as a single number "32"?
To interpret "16/2 32 + 80 ~ 2.333 ms" Block size: 16 samples
Is this what ALSA would call the number of frames per period? I presume your sample is the same as an ALSA frame. (As I recall, in ALSA each frame in a stereo stream contains two samples. You _are_ using stereo, right? And each sample would be 3 bytes for 24-bit audio. Also, in ALSA the period size and block size are the sizes in bytes, not in frames.)
Yes, I am using 2 channels input and 2 channels output for testing on Linux.
Periods: 2 (one period for playback + one period for recording when determining round trip latency)
In other words, one period per block in each direction?
Yes.
The minimum round trip latency is: 16 * 2 = 32 samples However, I measured 112 samples round trip latency which is an additional delay of 80 samples (32 + 80 = 112). 112 samples at 48000 Hz is 112 / 48000 * 1000 is approximately 2.333 ms measured round trip latency.
16/3 48 + 109 ~ 3.271 ms
Presumably this indicates three periods, then. Is that two in the outward direction and one in the inward direction, or vice versa?
Yes, one period is always for capture and the remaining periods are for playback.
32/2 64 + 129 ~ 4.021 ms 32/3 96 + 166 ~ 5.458 ms 64/2 128 + 205 ~ 6.938 ms 64/3 192 + 242 ~ 9.042 ms 128/2 256 + 352 ~ 12.667 ms 128/3 384 + 496 ~ 18.334 ms 256/2 512 + 650 ~ 24.208 ms 256/3 768 + 650 ~ 29.542 ms 512/2 1024 + 634 ~ 34.542 ms 512/3 1536 + 634 ~ 45.208 ms 1024/2 2048 + 650 ~ 56.208 ms 1024/3 3072 + 650 ~ 77.542 ms 2048/2 4096 + 633 ~ 98.521 ms 2048/3 6144 + 633 ~ 141.188 ms
As compared to the other systems, it appears that in Linux the additional delay increases with the period size. This could be a result of the initial zero-filled URBs, since the size or number of those URBs may depend on the other settings.
macOS High Sierra, Class Compliant Mode (Apple Driver): 16/2 32 + 205 ~ 4.938 ms 32/2 64 + 205 ~ 5.604 ms 64/2 128 + 205 ~ 6.938 ms 128/2 256 + 205 ~ 9.604 ms 256/2 512 + 205 ~ 14.938 ms 512/2 1024 + 205 ~ 25.604 ms 1024/2 2048 + 205 ~ 46.938 ms 2048/2 4096 + 205 ~ 89.604 ms
What are the USB parameters for these tests? How many bytes/frame? What is the endpoint's maxpacket size? What is the speed of the USB bus?
How would I determine the USB parameters and bytes/frame?
USB port is Intel USB 3.0 port. Device is USB 2.0 high speed (480 Mbps).
Here is the lsusb output:
Both too much information and too little. Instead, let's see the device's entry in /sys/kernel/debug/usb/devices, copied at a time while the test is running. That will omit a lot of irrelevant information and will indicate which of all the possible device settings is the one actually in use.
T: Bus=01 Lev=01 Prnt=01 Port=04 Cnt=01 Dev#= 2 Spd=480 MxCh= 0 D: Ver= 2.00 Cls=ef(misc ) Sub=02 Prot=01 MxPS=64 #Cfgs= 1 P: Vendor=2a39 ProdID=3fb0 Rev= 0.01 S: Manufacturer=RME S: Product=Babyface Pro (71964099) S: SerialNumber=EF72ADBCCECA4C8 C:* #Ifs= 4 Cfg#= 1 Atr=80 MxPwr=100mA A: FirstIf#= 0 IfCount= 4 Cls=01(audio) Sub=00 Prot=20 I:* If#= 0 Alt= 0 #EPs= 0 Cls=01(audio) Sub=01 Prot=20 Driver=snd-usb-audio I: If#= 1 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio I:* If#= 1 Alt= 1 #EPs= 2 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=03(O) Atr=05(Isoc) MxPS= 150 Ivl=125us E: Ad=83(I) Atr=11(Isoc) MxPS= 4 Ivl=1ms I: If#= 1 Alt= 2 #EPs= 2 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=03(O) Atr=05(Isoc) MxPS= 900 Ivl=125us E: Ad=83(I) Atr=11(Isoc) MxPS= 4 Ivl=1ms I: If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio I: If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=84(I) Atr=05(Isoc) MxPS= 900 Ivl=125us I:* If#= 2 Alt= 2 #EPs= 1 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=84(I) Atr=05(Isoc) MxPS= 150 Ivl=125us I:* If#= 3 Alt= 0 #EPs= 2 Cls=01(audio) Sub=03 Prot=00 Driver=snd-usb-audio E: Ad=07(O) Atr=02(Bulk) MxPS= 512 Ivl=0ms E: Ad=86(I) Atr=02(Bulk) MxPS= 512 Ivl=0ms
If you want to get a better idea for exactly what is happening at the USB level, you can collect a usbmon trace while running a test. Also, it wouldn't hurt to see the values of max_packs_per_urb, urb_packs, max_packs_per_period, urbs_per_period, ep->max_urb_frames, and ep->nurbs from data_ep_set_params() in the audio driver.
Maybe in a few weeks. I suspect the additional latency is mainly in the playback direction.
Regards, Jonathan
On Wed, 24 Oct 2018, Jonathan Liu wrote:
Both too much information and too little. Instead, let's see the device's entry in /sys/kernel/debug/usb/devices, copied at a time while the test is running. That will omit a lot of irrelevant information and will indicate which of all the possible device settings is the one actually in use.
T: Bus=01 Lev=01 Prnt=01 Port=04 Cnt=01 Dev#= 2 Spd=480 MxCh= 0 D: Ver= 2.00 Cls=ef(misc ) Sub=02 Prot=01 MxPS=64 #Cfgs= 1 P: Vendor=2a39 ProdID=3fb0 Rev= 0.01 S: Manufacturer=RME S: Product=Babyface Pro (71964099) S: SerialNumber=EF72ADBCCECA4C8 C:* #Ifs= 4 Cfg#= 1 Atr=80 MxPwr=100mA A: FirstIf#= 0 IfCount= 4 Cls=01(audio) Sub=00 Prot=20 I:* If#= 0 Alt= 0 #EPs= 0 Cls=01(audio) Sub=01 Prot=20 Driver=snd-usb-audio
This is the control interface; it is not directly involved.
I: If#= 1 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio I:* If#= 1 Alt= 1 #EPs= 2 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=03(O) Atr=05(Isoc) MxPS= 150 Ivl=125us E: Ad=83(I) Atr=11(Isoc) MxPS= 4 Ivl=1ms
This is one of the interfaces in use; it handles playback data (i.e., data sent to the device). The maxpacket size is 150 bytes, which is 25 frames at 3 bytes/sample and 2 channels. The interval is 125 us, giving a maximum throughput of 200 frames/ms, comfortably larger than the bandwidth being used (48 frames/ms).
I: If#= 1 Alt= 2 #EPs= 2 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=03(O) Atr=05(Isoc) MxPS= 900 Ivl=125us E: Ad=83(I) Atr=11(Isoc) MxPS= 4 Ivl=1ms I: If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio I: If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=84(I) Atr=05(Isoc) MxPS= 900 Ivl=125us I:* If#= 2 Alt= 2 #EPs= 1 Cls=01(audio) Sub=02 Prot=20 Driver=snd-usb-audio E: Ad=84(I) Atr=05(Isoc) MxPS= 150 Ivl=125us
This is the other interface being used for audio data; it handles the record direction. Parameters are the same as for playback.
I:* If#= 3 Alt= 0 #EPs= 2 Cls=01(audio) Sub=03 Prot=00 Driver=snd-usb-audio E: Ad=07(O) Atr=02(Bulk) MxPS= 512 Ivl=0ms E: Ad=86(I) Atr=02(Bulk) MxPS= 512 Ivl=0ms
I don't know what this interface is for. It's probably not directly relevant to the issue.
If you want to get a better idea for exactly what is happening at the USB level, you can collect a usbmon trace while running a test. Also, it wouldn't hurt to see the values of max_packs_per_urb, urb_packs, max_packs_per_period, urbs_per_period, ep->max_urb_frames, and ep->nurbs from data_ep_set_params() in the audio driver.
Maybe in a few weeks. I suspect the additional latency is mainly in the playback direction.
That seems likely, especially in the light of Takashi's comments.
Alan Stern
participants (4)
-
Alan Stern -
Jonathan Liu -
Pierre-Louis Bossart -
Takashi Iwai