Re: [Alsa-user] Get SPDIF-in sample rate
2017-07-19 8:46 GMT+02:00 Clemens Ladisch <clem...@ladisch.de>: > The HID interface is managed by another driver. The kernel driver > already knows how to write to these registers with a control request > instead (see below). Could you check if a control read request works, > or if the response still goes through the HID pipe? > I've tried that, but unfortunately it didn't work. The input request reads zero bytes. Well, I think that means I'll have to use my workarount with libhidapi. Below is my code. I don't feel very confident on all those bmRequestType, bRequest, wValue and other parameters, I just used the ones from your example. May be later I will be able to sniff the communication of the card with its proprietary Windows driver, and if I get a different results I'll revert back to you. int cm106_read(libusb_device_handle *handle, char reg, uint16_t *data) { int res; unsigned char buf[4] = {0x30, // 0x20 to write, 0x30 to read 0x00, // DATAL 0x00, // DATAH reg}; // Register address if ((res = libusb_control_transfer(handle, LIBUSB_ENDPOINT_OUT | LIBUSB_REQUEST_TYPE_CLASS | LIBUSB_RECIPIENT_ENDPOINT, LIBUSB_REQUEST_SET_CONFIGURATION, 0, 0, buf, 4, 0)) != 4) return res; if ((res = libusb_control_transfer(handle, LIBUSB_ENDPOINT_IN | LIBUSB_REQUEST_TYPE_CLASS | LIBUSB_RECIPIENT_ENDPOINT, LIBUSB_REQUEST_SET_CONFIGURATION, 0, 0, buf, 4, 0)) < 0) // <-- THIS FUNCTION RETURNS 0 return res; if (res != 3) { fprintf(stderr, "DEBUG Invalid response length" return -1; } if (buf[0] & 0xe0 != 0x20) // No register data in the input { fprintf(stderr, "DEBUG data: %02X %02X %02X\n", buf[0], buf[1], buf[2]); return -1; } *data = (((uint16_t)buf[2]) << 8) + buf[1]; return 0; } // Just for completeness, the below function successfully writes the data into cm106 registers int cm106_write(libusb_device_handle *handle, char reg, uint16_t data) { unsigned char buf[4] = {0x20, // 0x20 to write, 0x30 to read data & 0xff,// DATAL (data >> 8) & 0xff, // DATAH reg}; // Register address return libusb_control_transfer(handle, LIBUSB_ENDPOINT_OUT | LIBUSB_REQUEST_TYPE_CLASS | LIBUSB_RECIPIENT_ENDPOINT, LIBUSB_REQUEST_SET_CONFIGURATION, 0, 0, buf, 4, 0); } Thank you for your support! Best regards Denis Shulyaka -- Check out the vibrant tech community on one of the world's most engaging tech sites, Slashdot.org! http://sdm.link/slashdot___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Get SPDIF-in sample rate
2017-07-18 8:47 GMT+02:00 Clemens Ladisch <clem...@ladisch.de>: > In general, SPDIF inputs do not resample, and if you try to record with > the wrong sample rate, the result has the original sample rate and is > just labelled wrong. > That is a good news, but I would still prefer to know the sample rate because if I want to output it, I would need to open the output device with the correct sample rate, or I will still have a resampling. > Does ALSA provide any general interface for the userspace applications > > that indicates the SPDIF source sample rate? > > Yes, but the USB audio driver does not implement it because the USB > audio specification does not define such an interface between the driver > and the device. > What could I use if I had a non-USB sound card with SPDIF-in? > At the moment, you have to write the code yourself. But if you've > tested it, we would be interested in integrating it into the kernel > driver. > The below program outputs the current SPDIF-In sample rate of my card. The only problem with it is that my sense of perfection hurts when I'm working around the driver like that. All device-specific parts should be in kernel. [root@speaker cm106spdif]# gcc cm106spdif.c -l hidapi-libusb [root@speaker cm106spdif]# ./a.out 44.1K [root@speaker cm106spdif]# cat cm106spdif.c #include #include #include #include #include int cm106_read(hid_device *handle, char reg, uint16_t *data) { unsigned char buf[5] = {0x00,// report id for hidapi 0x30,// 0x20 to write, 0x30 to read 0x00,// DATAL 0x00,// DATAH reg};// Register address if (hid_write(handle, buf, 5) != 5) return -1; if (hid_read(handle, buf, 5) != 3) return -2; if (buf[0] & 0xe0 != 0x20)// No register data in the input report return -3; *data = (((uint16_t)buf[2]) << 8) + buf[1]; return 0; } int cm106_write(hid_device *handle, char reg, uint16_t data) { unsigned char buf[5] = {0x00,// report id for hidapi 0x20,// 0x20 to write, 0x30 to read data & 0xff,// DATAL (data >> 8),// DATAH reg};// Register address if (hid_write(handle, buf, 5) != 5) return -1; return 0; } int main(void) { hid_device *handle; uint16_t data = 0; unsigned char SPDFI_FREQ; if ( !(handle = hid_open(0xd8c, 0x102, NULL)) ) err(1, "hid_open: %ls", hid_error(handle)); if (cm106_read(handle, 3, ) < 0) err(2, "read: %ls", hid_error(handle)); //printf("data=%04X\n", data); SPDFI_FREQ = (data & 0x0180) >> 7; switch(SPDFI_FREQ) { case 0: printf("44.1K\n"); break; case 2: printf("48K\n"); break; case 3: printf("32K\n"); break; default: printf("reserved\n"); } hid_close(handle); hid_exit(); return 0; } Best regards, Denis Shulyaka -- Check out the vibrant tech community on one of the world's most engaging tech sites, Slashdot.org! http://sdm.link/slashdot___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Get SPDIF-in sample rate
Hi, I have a CM106-based usb sound card (0d8c:0102) with SPDIF-In. My aim is to decode the compressed audio formats that may come from this source. And the first task for this is to get the original sample rate of the source to avoid any resampling. According to the CM106 datasheet, one can read the SPDIF-In sample rate from SPDFI_FREQ bits (8~7 of register 03). What is the correct way to do so? Can ALSA read those register bits? Does ALSA provide any general interface for the userspace applications that indicates the SPDIF source sample rate? Should I attempt to communicate with the hardware with libusb, or can I do it with ALSA only? Please advise. Best regards, Denis Shulyaka -- Check out the vibrant tech community on one of the world's most engaging tech sites, Slashdot.org! http://sdm.link/slashdot___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Hardware loopback
Hi, 2014-01-16 12:04 GMT+04:00 Clemens Ladisch cladi...@googlemail.com: Denis Shulyaka wrote: 2014/1/14 Clemens Ladisch cladi...@googlemail.com: This device's descriptor do not define a mixer control for this control. Apparently, you have to set the SPDIFMIX bit. Well, do you have any idea on how can I do it? I'm OK with compiling a custom kernel, but I don't have any experience with ALSA code. Just a small hint would be appreciated) Can I somehow define it in the descriptor? This is not a mixer control that could be described with a descriptor; one needs to write a custom control that writes to that vendor-specific register. I'll write a patch ... Regards, Clemens Well, I've finally managed to set the SPDIFMIX bit in by modifying the snd_usb_cm6206_boot_quirk() function. However it led me to another problem: PCM streams are played fine and are routed to the analog speakers without CPU interaction, but DTS streams sound like a loud noise. Do I understand it right that the card hardware is not able to decode DTS itself and it merely relies on a proprietary software driver for DTS support? Can Alsa and PulseAudio deal with DTS? I just need to get the idea what DTS is... Best regards, Denis Shulyaka -- Open source business process management suite built on Java and Eclipse Turn processes into business applications with Bonita BPM Community Edition Quickly connect people, data, and systems into organized workflows Winner of BOSSIE, CODIE, OW2 and Gartner awards http://p.sf.net/sfu/Bonitasoft___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Hardware loopback
1 PCM AudioStreaming Interface Descriptor: bLength14 bDescriptorType36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType2 Discrete tSamFreq[ 0]44100 tSamFreq[ 1]48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes5 Transfer TypeIsochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType33 bcdHID 1.00 bCountryCode0 Not supported bNumDescriptors 1 bDescriptorType34 Report wDescriptorLength 50 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes3 Transfer TypeInterrupt Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 Device Status: 0x (Bus Powered) pi@speaker ~ $ Best regards, Denis Shulyaka lsusb_v Description: Binary data -- CenturyLink Cloud: The Leader in Enterprise Cloud Services. Learn Why More Businesses Are Choosing CenturyLink Cloud For Critical Workloads, Development Environments Everything In Between. Get a Quote or Start a Free Trial Today. http://pubads.g.doubleclick.net/gampad/clk?id=119420431iu=/4140/ostg.clktrk___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Hardware loopback
Hi, 2014/1/14 Clemens Ladisch cladi...@googlemail.com: This device's descriptor do not define a mixer control for this control. Apparently, you have to set the SPDIFMIX bit. Well, do you have any idea on how can I do it? I'm OK with compiling a custom kernel, but I don't have any experience with ALSA code. Just a small hint would be appreciated) Can I somehow define it in the descriptor? Best regards, Denis Shulyaka -- CenturyLink Cloud: The Leader in Enterprise Cloud Services. Learn Why More Businesses Are Choosing CenturyLink Cloud For Critical Workloads, Development Environments Everything In Between. Get a Quote or Start a Free Trial Today. http://pubads.g.doubleclick.net/gampad/clk?id=119420431iu=/4140/ostg.clktrk ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Hardware loopback
Hi list, Please forgive if it's the wrong place to post this question. I'm trying to build a sound server connected to my speaker set that will act as: 1) SPDIF receiver from TV 2) Network sound server My hardware: 1) Raspberry Pi 2) USB sound card based on CM106 chip. My Software: 1) Raspbian 2) Linux 3.10.25 3) Pulseaudio 2.0 (will try to compile 4.0 tonight) For this to work as a SPDIF receiver I need to somehow connect SPDIF IN and analog out. So my question is, What is the best way to configure it? If I get it right, I have three options here: 1) Configure a loopback on hardware level for the best latency and CPU usage 2) Configure a loopback on alsa level 3) Configure a loopback on pulseaudio level since I will use it for mixing anyway I am currently trying to understand if my hardware supports Option 1. I have found a CM106 datasheet which suggests that setting bit 2 (SPDIFLOOP) of register REG1 will enable SPDIF loop-back, however it is not clear whether the sound from SPDIF IN will be routed to SPDIF OUT or analog output and also whether this feature is what I'm looking for. If the first option is not available, I would also ask for your advice on which of the two last options is preferable. Thanks in advance! Best regards, Denis Shulyaka -- CenturyLink Cloud: The Leader in Enterprise Cloud Services. Learn Why More Businesses Are Choosing CenturyLink Cloud For Critical Workloads, Development Environments Everything In Between. Get a Quote or Start a Free Trial Today. http://pubads.g.doubleclick.net/gampad/clk?id=119420431iu=/4140/ostg.clktrk___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Hardware loopback
Hi Clemens, 2014/1/13 Clemens Ladisch cladi...@googlemail.com As you can see in the block diagram, the S/PDIF input can be routed to both the S/PDIF output and to the analong front outputs. Loopback usually means the same kind of connection, so there should be a normal mixer control for the front output. Thank you for the response. Unfortunately I am unable to find such control myself. Below is my amixer output: pi@speaker ~ $ amixer -c 0 info Card hw:0 'Device'/'USB Sound Device at usb-bcm2708_usb-1.2.1, full speed' Mixer name: 'USB Mixer' Components: 'USB0d8c:0102' Controls : 15 Simple ctrls : 6 pi@speaker ~ $ amixer -c 0 scontents Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right - Rear Left - Rear Right - Front Center - Woofer - Side Left - Side Right Limits: Playback 0 - 197 Mono: Front Left: Playback 90 [46%] [-20.06dB] [on] Front Right: Playback 90 [46%] [-20.06dB] [on] Rear Left: Playback 90 [46%] [-20.06dB] [on] Rear Right: Playback 90 [46%] [-20.06dB] [on] Front Center: Playback 90 [46%] [-20.06dB] [on] Woofer: Playback 90 [46%] [-20.06dB] [on] Side Left: Playback 90 [46%] [-20.06dB] [on] Side Right: Playback 90 [46%] [-20.06dB] [on] Simple mixer control 'PCM',0 Capabilities: cvolume cswitch cswitch-joined penum Capture channels: Front Left - Front Right Limits: Capture 0 - 6928 Front Left: Capture 4096 [59%] [0.01dB] [on] Front Right: Capture 4096 [59%] [0.01dB] [on] Simple mixer control 'PCM Capture Source',0 Capabilities: enum Items: 'Mic' 'Line' 'IEC958 In' 'Mixer' Item0: 'IEC958 In' Simple mixer control 'Line',0 Capabilities: pvolume cvolume pswitch pswitch-joined cswitch cswitch-joined penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 8065 Capture 0 - 6928 Front Left: Playback 6144 [76%] [0.01dB] [off] Capture 0 [0%] [-16.00dB] [off] Front Right: Playback 6144 [76%] [0.01dB] [off] Capture 0 [0%] [-16.00dB] [off] Simple mixer control 'Mic',0 Capabilities: pvolume cvolume pswitch pswitch-joined cswitch cswitch-joined penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 8065 Capture 0 - 6928 Front Left: Playback 6153 [76%] [0.03dB] [off] Capture 4096 [59%] [0.01dB] [on] Front Right: Playback 6153 [76%] [0.03dB] [off] Capture 4096 [59%] [0.01dB] [on] Simple mixer control 'IEC958 In',0 Capabilities: cswitch cswitch-joined penum Capture channels: Mono Mono: Capture [on] The two controls related to SPDIF In are 'PCM Capture Source' and 'IEC958 In'. I'm not sure what they mean. The latter seems to enable/disable the mix of the SPDIF In into capture stream (i.e., Input). For the first one I have no idea. I will try to experiment with different values of the controls, but unfortunately for the following two weeks I will not have any SPDIF signal source to play with. Best regards, Denis Shulyaka -- CenturyLink Cloud: The Leader in Enterprise Cloud Services. Learn Why More Businesses Are Choosing CenturyLink Cloud For Critical Workloads, Development Environments Everything In Between. Get a Quote or Start a Free Trial Today. http://pubads.g.doubleclick.net/gampad/clk?id=119420431iu=/4140/ostg.clktrk ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user