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Apple USB-C dongle cannot output 44100hz
by GitHub issues - opened 16 May '23

16 May '23

alsa-project/alsa-lib issue #322 was opened from tw-hx: I have the same issue as outlined here, with an A2049 Apple USB-C to 3.5mm dongle: https://github.com/mikebrady/shairport-sync/issues/1504 Running Arch Linux with kernel 6.3.1-arch2-1, alsa-lib 1.2.9-1. Pipewire reports that the output is locked to 48000hz when it should be switching to 41000hz, and ALSA reports that this is the only detected supported rate: ``` $ cat /proc/asound/card0/stream0 Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 288 Momentary freq = 48000 Hz (0x30.0000) Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 0x02 (2 OUT) (SYNC) Rates: 48000 - 48000 (continuous) Bits: 0 Channel map: FL FR Interface 1 Altset 2 Format: S16_LE Channels: 2 Endpoint: 0x02 (2 OUT) (SYNC) Rates: 48000 - 48000 (continuous) Bits: 0 Channel map: FL FR ``` Thanks if this can be fixed. It seems like a regression between Linux 4.x and 5.x, and halfway through the above thread the user [quantonian](https://github.com/quantonian) added the device to the ALSA quirks table and has posted a patch. Issue URL : https://github.com/alsa-project/alsa-lib/issues/322 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 0/7] ALSA: emu10k1: various improvements to the DSP-based mixer code
by Oswald Buddenhagen 16 May '23

16 May '23

Oswald Buddenhagen (7): ALSA: emu10k1: polish audigy GPR allocation ALSA: emu10k1: fix non-zero mixer control defaults in highres mode ALSA: emu10k1: validate min/max values of translated controls ALSA: emu10k1: omit non-applicable mixer controls for E-MU cards ALSA: emu10k1: skip mic capture PCM for cards without AC97 codec ALSA: emu10k1: enable bit-exact playback, part 1: DSP attenuation ALSA: emu10k1: enable bit-exact playback, part 2: voice attenuation Documentation/sound/cards/audigy-mixer.rst | 2 +- Documentation/sound/cards/sb-live-mixer.rst | 2 +- include/sound/emu10k1.h | 9 +- include/uapi/sound/emu10k1.h | 8 +- sound/pci/emu10k1/emu10k1.c | 8 +- sound/pci/emu10k1/emufx.c | 491 +++++++++++--------- sound/pci/emu10k1/emumixer.c | 15 +- sound/pci/emu10k1/emupcm.c | 4 +- 8 files changed, 291 insertions(+), 248 deletions(-) -- 2.40.0.152.g15d061e6df

2 10

Update USB-Audio.conf for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080
by GitHub pull_request - edited 16 May '23

16 May '23

alsa-project/alsa-ucm-conf pull request #315 was edited from Sunsheep: Added configuration for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080 Edit: Added Information from ALSA Information Script [https://alsa-project.org/db/?f=fe7e76a335dc8f1009ecaac9aa341fb68231aee5](ht… ``` Bus 001 Device 002: ID 0b05:1a5c ASUSTek Computer, Inc. USB Audio Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x0b05 ASUSTek Computer, Inc. idProduct 0x1a5c bcdDevice 0.02 iManufacturer 3 Generic iProduct 1 USB Audio iSerial 0 ``` Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Update USB-Audio.conf for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080
by GitHub pull_request - edited 16 May '23

16 May '23

alsa-project/alsa-ucm-conf pull request #315 was edited from Sunsheep: Added configuration for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080 Edit: Added Information from ALSA Information Script [https://alsa-project.org/db/?f=fe7e76a335dc8f1009ecaac9aa341fb68231aee5](ht… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Update USB-Audio.conf for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080
by GitHub pull_request - edited 16 May '23

16 May '23

alsa-project/alsa-ucm-conf pull request #315 was edited from Sunsheep: Added configuration for ASUS ROG Strix B650E-I Gaming WiFi and ALC4080 Edit: Added Information from ALSA Information Script [https://alsa-project.org/db/?f=fe7e76a335dc8f1009ecaac9aa341fb68231aee5](ur… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/315.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

external rate plugin: broken 16-bit stereo stream conversion (interleaved -> non-interleaved)
by GitHub issues - edited 16 May '23

16 May '23

alsa-project/alsa-lib issue #320 was edited from ossilator: this setup: ``` Plug PCM: Rate conversion PCM (44100, sformat=S16_LE) Converter: libspeex (external) Protocol version: 10003 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 17832 period_size : 4458 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4458 period_event : 0 start_threshold : 17832 stop_threshold : 17832 silence_threshold: 0 silence_size : 0 boundary : 5019261784704417792 Slave: Hardware PCM card 0 'E-MU 0404b PCI [MAEM8852]' device 3 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_NONINTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 ``` garbles the output rather impressively, though it's still recognizable as the input, more or less. it happens both when up-sampling and down-sampling. it doesn't happen when not re-sampling. it doesn't happen when the hardware takes 32-bit samples, so i guess it's somewhere in the optimized s16 paths. it doesn't happen with mono. this is not reproducible with the upstream emu10k1 driver, as the multichannel device demands 16 channels, which causes the route conversion plugin to be inserted, which sidesteps the issue. and the regular pcm device accepts any rate. Issue URL : https://github.com/alsa-project/alsa-lib/issues/320 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

external rate plugin: broken 16-bit stream conversion (interleaved -> non-interleaved)
by GitHub issues - edited 16 May '23

16 May '23

alsa-project/alsa-lib issue #320 was edited from ossilator: this setup: ``` Plug PCM: Rate conversion PCM (44100, sformat=S16_LE) Converter: libspeex (external) Protocol version: 10003 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 17832 period_size : 4458 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4458 period_event : 0 start_threshold : 17832 stop_threshold : 17832 silence_threshold: 0 silence_size : 0 boundary : 5019261784704417792 Slave: Hardware PCM card 0 'E-MU 0404b PCI [MAEM8852]' device 3 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_NONINTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 0 ``` garbles the output rather impressively, though it's still recognizable as the input, more or less. it happens both when up-sampling and down-sampling. it doesn't happen when not re-sampling. it doesn't happen when the hardware takes 32-bit samples, so i guess it's somewhere in the optimized s16 paths. it doesn't happen with mono. this is not reproducible with the upstream emu10k1 driver, as the multichannel device demands 16 channels, which causes the route conversion plugin to be inserted, which sidesteps the issue. and the regular pcm device accepts any rate. Issue URL : https://github.com/alsa-project/alsa-lib/issues/320 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: da7219: Add Jack insertion detection polarity selection
by David Rau 16 May '23

16 May '23

DA7219 can support 2 kinds of insertion detection polarity - Default polarity (Low) - Inverted polarity (High) This patch adds support for selecting insertion detection polarity to the DT binding. Signed-off-by: David Rau <David.Rau.opensource(a)dm.renesas.com> --- .../devicetree/bindings/sound/da7219.txt | 3 ++ include/sound/da7219-aad.h | 6 ++++ sound/soc/codecs/da7219-aad.c | 34 +++++++++++++++++++ 3 files changed, 43 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt index add1caf26ac2..357d1f61de1d 100644 --- a/Documentation/devicetree/bindings/sound/da7219.txt +++ b/Documentation/devicetree/bindings/sound/da7219.txt @@ -52,6 +52,8 @@ Optional properties: [<200>, <500>, <750>, <1000>] - dlg,jack-ins-deb : Debounce time for jack insertion (ms) [<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>] +- dlg,jack-ins-det-pty : Polarity for jack insertion detection + ["low", "high"] - dlg,jack-det-rate: Jack type detection latency (3/4 pole) ["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"] - dlg,jack-rem-deb : Debounce time for jack removal (ms) @@ -98,6 +100,7 @@ Example: dlg,btn-cfg = <50>; dlg,mic-det-thr = <500>; dlg,jack-ins-deb = <20>; + dlg,jack-ins-det-pty = "low"; dlg,jack-det-rate = "32ms_64ms"; dlg,jack-rem-deb = <1>; diff --git a/include/sound/da7219-aad.h b/include/sound/da7219-aad.h index 24ee7baa2589..41320522daa2 100644 --- a/include/sound/da7219-aad.h +++ b/include/sound/da7219-aad.h @@ -44,6 +44,11 @@ enum da7219_aad_jack_ins_deb { DA7219_AAD_JACK_INS_DEB_1S, }; +enum da7219_aad_jack_ins_det_pty { + DA7219_AAD_JACK_INS_DET_PTY_LOW = 0, + DA7219_AAD_JACK_INS_DET_PTY_HIGH, +}; + enum da7219_aad_jack_det_rate { DA7219_AAD_JACK_DET_RATE_32_64MS = 0, DA7219_AAD_JACK_DET_RATE_64_128MS, @@ -80,6 +85,7 @@ struct da7219_aad_pdata { enum da7219_aad_btn_cfg btn_cfg; enum da7219_aad_mic_det_thr mic_det_thr; enum da7219_aad_jack_ins_deb jack_ins_deb; + enum da7219_aad_jack_ins_det_pty jack_ins_det_pty; enum da7219_aad_jack_det_rate jack_det_rate; enum da7219_aad_jack_rem_deb jack_rem_deb; diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 993a0d00bc48..a61dc965f4fc 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -571,6 +571,19 @@ static enum da7219_aad_jack_ins_deb } } +static enum da7219_aad_jack_ins_det_pty + da7219_aad_fw_jack_ins_det_pty(struct device *dev, const char *str) +{ + if (!strcmp(str, "low")) { + return DA7219_AAD_JACK_INS_DET_PTY_LOW; + } else if (!strcmp(str, "high")) { + return DA7219_AAD_JACK_INS_DET_PTY_HIGH; + } else { + dev_warn(dev, "Invalid jack insertion detection polarity"); + return DA7219_AAD_JACK_INS_DET_PTY_LOW; + } +} + static enum da7219_aad_jack_det_rate da7219_aad_fw_jack_det_rate(struct device *dev, const char *str) { @@ -688,6 +701,12 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct device *dev) else aad_pdata->jack_ins_deb = DA7219_AAD_JACK_INS_DEB_20MS; + if (!fwnode_property_read_string(aad_np, "dlg,jack-ins-det-pty", &fw_str)) + aad_pdata->jack_ins_det_pty = + da7219_aad_fw_jack_ins_det_pty(dev, fw_str); + else + aad_pdata->jack_ins_det_pty = DA7219_AAD_JACK_INS_DET_PTY_LOW; + if (!fwnode_property_read_string(aad_np, "dlg,jack-det-rate", &fw_str)) aad_pdata->jack_det_rate = da7219_aad_fw_jack_det_rate(dev, fw_str); @@ -849,6 +868,21 @@ static void da7219_aad_handle_pdata(struct snd_soc_component *component) mask |= DA7219_ADC_1_BIT_REPEAT_MASK; } snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_7, mask, cfg); + + switch (aad_pdata->jack_ins_det_pty) { + case DA7219_AAD_JACK_INS_DET_PTY_LOW: + snd_soc_component_write(component, 0xF0, 0x8B); + snd_soc_component_write(component, 0x75, 0x80); + snd_soc_component_write(component, 0xF0, 0x00); + break; + case DA7219_AAD_JACK_INS_DET_PTY_HIGH: + snd_soc_component_write(component, 0xF0, 0x8B); + snd_soc_component_write(component, 0x75, 0x00); + snd_soc_component_write(component, 0xF0, 0x00); + break; + default: + break; + } } } -- 2.17.1

3 2

alsa-lib hangs when mounting a partition while playing a sound file using aplay
by GitHub issues - opened 16 May '23

16 May '23

alsa-project/alsa-lib issue #321 was opened from nachopitt: Hello, I'm using alsa-lib to add sound playback functionality to a Embedded Linux system. Recently I noticed that whenever I mount a UBI partition while there is a sound file being played back with the help of the aplay command, the aplay process hangs forever and the sound playback stops immediately. ``` # aplay -D output1 1.wav & Playing WAVE '1.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono # # # mount -t ubifs /dev/ubi0_5 /opt/conf/Logging UBIFS (ubi0:5): background thread "ubifs_bgt0_5" started, PID 190 UBIFS (ubi0:5): UBIFS: mounted UBI device 0, volume 5, name "Logging" UBIFS (ubi0:5): LEB size: 126976 bytes (124 KiB), min./max. I/O unit sizes: 2048 bytes/2048 bytes UBIFS (ubi0:5): FS size: 1650688 bytes (1 MiB, 13 LEBs), journal size 1142785 bytes (1 MiB, 8 LEBs) underrun!!! (at least 0.044 ms loUBIFS (ubi0:5): reserved for root: 0 bytes (0 KiB) ng) UBIFS (ubi0:5): media format: w4/r0 (latest is w5/r0), UUID 26A3D522-A34F-4FE4-BF6C-931A9FE59854, small LPT model # ``` I also prepared a small video that captures the exact moment where the issue is reproduced in case you want it to see it in action: https://www.youtube.com/watch?v=M9bW2k2CTVs The only way to recover from this scenario is to kill the stuck aplay process manually and execute it again. Last time I tried to debug this, I attached the gdbserver to the still stuck aplay process and backtrace showed the following information: ``` (gdb) bt arm-linux-gnueabihf-gdb bin/aplay (wd: ~/home_volume/git_repos/alsa/alsa-install/usr) (gdb) bt #0 0x76d96ba4 in ?? () **#1 0x76f0e222 in snd1_pcm_wait_nocheck (pcm=0x879280, timeout=-1) at pcm.c:2898** Backtrace stopped: previous frame inner to this frame (corrupt stack?) (gdb) frame 1 #1 0x76f0e222 in snd1_pcm_wait_nocheck (pcm=0x879280, timeout=-1) at pcm.c:2898 2898 err_poll = poll(pfd, npfds, timeout); (gdb) ``` It is clear to me that the `poll` function is waiting for something to change in the file descriptors and it waits infinitely due to the `timeout = -1` argument, but none of the file descriptors in `pfd` ever change or become ready after the mount command is executed during the sound playback. I'm currently using both alsa-lib and alsa-utils v1.2.6. The SoC is an NXP i.MX6 Solo which includes an Armv7-A 32bits Cortex-A9 Core CPU Platform. Linux Kernel BSP is based on 4.14.98. Most likely this is an issue with my system configuration, but since you guys are more familiar with alsa than anyone on the planet, I thought you might have some ideas to troubleshoot and debug this because I currently do not have any clue. Do you know what could cause this behavior? Is there any possibility that there is an issue in alsa-lib? Do you have any suggestions to debug and troubleshoot the error to get more information? Any help will be really appreciated. Thanks a lot. Best regards. Issue URL : https://github.com/alsa-project/alsa-lib/issues/321 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 1/4] ALSA: emu10k1: automate encoding of sub-register definitions
by Oswald Buddenhagen 15 May '23

15 May '23

The idea to encode the bitfield manipulation in the register address is quite clever, but doing that by hand is ugly and error-prone. So derive it automatically from the mask instead. Macros cannot #define other macros, so we now declare enums instead. This also adds macros for decoding the register definitions. These will be used by later commits. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- v2: added comments --- include/sound/emu10k1.h | 123 ++++++++++++++++++++-------------------- 1 file changed, 60 insertions(+), 63 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 7129b9249eb3..e9b1729ade60 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -38,6 +38,32 @@ #define IP_TO_CP(ip) ((ip == 0) ? 0 : (((0x00001000uL | (ip & 0x00000FFFL)) << (((ip >> 12) & 0x000FL) + 4)) & 0xFFFF0000uL)) +// This is used to define hardware bit-fields (sub-registers) by combining +// the bit shift and count with the actual register address. The passed +// mask must represent a single run of adjacent bits. +// The non-concatenating (_NC) variant should be used directly only for +// sub-registers that do not follow the <register>_<field> naming pattern. +#define SUB_REG_NC(reg, field, mask) \ + enum { \ + field ## _MASK = mask, \ + field = reg | \ + (__builtin_ctz(mask) << 16) | \ + (__builtin_popcount(mask) << 24), \ + }; +#define SUB_REG(reg, field, mask) SUB_REG_NC(reg, reg ## _ ## field, mask) + +// Macros for manipulating values of bit-fields declared using the above macros. +// Best used with constant register addresses, as otherwise quite some code is +// generated. The actual register read/write functions handle combined addresses +// automatically, so use of these macros conveys no advantage when accessing a +// single sub-register at a time. +#define REG_SHIFT(r) (((r) >> 16) & 0x1f) +#define REG_SIZE(r) (((r) >> 24) & 0x1f) +#define REG_MASK0(r) ((1U << REG_SIZE(r)) - 1U) +#define REG_MASK(r) (REG_MASK0(r) << REG_SHIFT(r)) +#define REG_VAL_GET(r, v) ((v & REG_MASK(r)) >> REG_SHIFT(r)) +#define REG_VAL_PUT(r, v) ((v) << REG_SHIFT(r)) + // Audigy specify registers are prefixed with 'A_' /************************************************************************************************/ @@ -148,12 +174,10 @@ #define INTE_MIDIRXENABLE 0x00000001 /* Enable MIDI receive-buffer-empty interrupts */ #define WC 0x10 /* Wall Clock register */ -#define WC_SAMPLECOUNTER_MASK 0x03FFFFC0 /* Sample periods elapsed since reset */ -#define WC_SAMPLECOUNTER 0x14060010 -#define WC_CURRENTCHANNEL_MASK 0x0000003F /* Channel [0..63] currently being serviced */ +SUB_REG(WC, SAMPLECOUNTER, 0x03FFFFC0) /* Sample periods elapsed since reset */ +SUB_REG(WC, CURRENTCHANNEL, 0x0000003F) /* Channel [0..63] currently being serviced */ /* NOTE: Each channel takes 1/64th of a sample */ /* period to be serviced. */ -#define WC_CURRENTCHANNEL 0x06000010 #define HCFG 0x14 /* Hardware config register */ /* NOTE: There is no reason to use the legacy */ @@ -225,9 +249,8 @@ /* async audio source */ #define HCFG_LOCKSOUNDCACHE 0x00000008 /* 1 = Cancel bustmaster accesses to soundcache */ /* NOTE: This should generally never be used. */ -#define HCFG_LOCKTANKCACHE_MASK 0x00000004 /* 1 = Cancel bustmaster accesses to tankcache */ +SUB_REG(HCFG, LOCKTANKCACHE, 0x00000004) /* 1 = Cancel bustmaster accesses to tankcache */ /* NOTE: This should generally never be used. */ -#define HCFG_LOCKTANKCACHE 0x01020014 #define HCFG_MUTEBUTTONENABLE 0x00000002 /* 1 = Master mute button sets AUDIOENABLE = 0. */ /* NOTE: This is a 'cheap' way to implement a */ /* master mute function on the mute button, and */ @@ -382,55 +405,38 @@ // which the current registers "swerve" gradually. #define CPF 0x00 /* Current pitch and fraction register */ -#define CPF_CURRENTPITCH_MASK 0xffff0000 /* Current pitch (linear, 0x4000 == unity pitch shift) */ -#define CPF_CURRENTPITCH 0x10100000 +SUB_REG(CPF, CURRENTPITCH, 0xffff0000) /* Current pitch (linear, 0x4000 == unity pitch shift) */ #define CPF_STEREO_MASK 0x00008000 /* 1 = Even channel interleave, odd channel locked */ #define CPF_STOP_MASK 0x00004000 /* 1 = Current pitch forced to 0 */ #define CPF_FRACADDRESS_MASK 0x00003fff /* Linear fractional address of the current channel */ #define PTRX 0x01 /* Pitch target and send A/B amounts register */ -#define PTRX_PITCHTARGET_MASK 0xffff0000 /* Pitch target of specified channel */ -#define PTRX_PITCHTARGET 0x10100001 -#define PTRX_FXSENDAMOUNT_A_MASK 0x0000ff00 /* Linear level of channel output sent to FX send bus A */ -#define PTRX_FXSENDAMOUNT_A 0x08080001 -#define PTRX_FXSENDAMOUNT_B_MASK 0x000000ff /* Linear level of channel output sent to FX send bus B */ -#define PTRX_FXSENDAMOUNT_B 0x08000001 +SUB_REG(PTRX, PITCHTARGET, 0xffff0000) /* Pitch target of specified channel */ +SUB_REG(PTRX, FXSENDAMOUNT_A, 0x0000ff00) /* Linear level of channel output sent to FX send bus A */ +SUB_REG(PTRX, FXSENDAMOUNT_B, 0x000000ff) /* Linear level of channel output sent to FX send bus B */ #define CVCF 0x02 /* Current volume and filter cutoff register */ -#define CVCF_CURRENTVOL_MASK 0xffff0000 /* Current linear volume of specified channel */ -#define CVCF_CURRENTVOL 0x10100002 -#define CVCF_CURRENTFILTER_MASK 0x0000ffff /* Current filter cutoff frequency of specified channel */ -#define CVCF_CURRENTFILTER 0x10000002 +SUB_REG(CVCF, CURRENTVOL, 0xffff0000) /* Current linear volume of specified channel */ +SUB_REG(CVCF, CURRENTFILTER, 0x0000ffff) /* Current filter cutoff frequency of specified channel */ #define VTFT 0x03 /* Volume target and filter cutoff target register */ -#define VTFT_VOLUMETARGET_MASK 0xffff0000 /* Volume target of specified channel */ -#define VTFT_VOLUMETARGET 0x10100003 -#define VTFT_FILTERTARGET_MASK 0x0000ffff /* Filter cutoff target of specified channel */ -#define VTFT_FILTERTARGET 0x10000003 +SUB_REG(VTFT, VOLUMETARGET, 0xffff0000) /* Volume target of specified channel */ +SUB_REG(VTFT, FILTERTARGET, 0x0000ffff) /* Filter cutoff target of specified channel */ #define Z1 0x05 /* Filter delay memory 1 register */ #define Z2 0x04 /* Filter delay memory 2 register */ #define PSST 0x06 /* Send C amount and loop start address register */ -#define PSST_FXSENDAMOUNT_C_MASK 0xff000000 /* Linear level of channel output sent to FX send bus C */ - -#define PSST_FXSENDAMOUNT_C 0x08180006 - -#define PSST_LOOPSTARTADDR_MASK 0x00ffffff /* Loop start address of the specified channel */ -#define PSST_LOOPSTARTADDR 0x18000006 +SUB_REG(PSST, FXSENDAMOUNT_C, 0xff000000) /* Linear level of channel output sent to FX send bus C */ +SUB_REG(PSST, LOOPSTARTADDR, 0x00ffffff) /* Loop start address of the specified channel */ #define DSL 0x07 /* Send D amount and loop end address register */ -#define DSL_FXSENDAMOUNT_D_MASK 0xff000000 /* Linear level of channel output sent to FX send bus D */ - -#define DSL_FXSENDAMOUNT_D 0x08180007 - -#define DSL_LOOPENDADDR_MASK 0x00ffffff /* Loop end address of the specified channel */ -#define DSL_LOOPENDADDR 0x18000007 +SUB_REG(DSL, FXSENDAMOUNT_D, 0xff000000) /* Linear level of channel output sent to FX send bus D */ +SUB_REG(DSL, LOOPENDADDR, 0x00ffffff) /* Loop end address of the specified channel */ #define CCCA 0x08 /* Filter Q, interp. ROM, byte size, cur. addr register */ -#define CCCA_RESONANCE_MASK 0xf0000000 /* Lowpass filter resonance (Q) height */ -#define CCCA_RESONANCE 0x041c0008 +SUB_REG(CCCA, RESONANCE, 0xf0000000) /* Lowpass filter resonance (Q) height */ #define CCCA_INTERPROM_MASK 0x0e000000 /* Selects passband of interpolation ROM */ /* 1 == full band, 7 == lowpass */ /* ROM 0 is used when pitch shifting downward or less */ @@ -447,27 +453,24 @@ #define CCCA_INTERPROM_7 0x0e000000 /* Select interpolation ROM 7 */ #define CCCA_8BITSELECT 0x01000000 /* 1 = Sound memory for this channel uses 8-bit samples */ /* 8-bit samples are unsigned, 16-bit ones signed */ -#define CCCA_CURRADDR_MASK 0x00ffffff /* Current address of the selected channel */ -#define CCCA_CURRADDR 0x18000008 +SUB_REG(CCCA, CURRADDR, 0x00ffffff) /* Current address of the selected channel */ #define CCR 0x09 /* Cache control register */ -#define CCR_CACHEINVALIDSIZE 0x07190009 -#define CCR_CACHEINVALIDSIZE_MASK 0xfe000000 /* Number of invalid samples before the read address */ +SUB_REG(CCR, CACHEINVALIDSIZE, 0xfe000000) /* Number of invalid samples before the read address */ #define CCR_CACHELOOPFLAG 0x01000000 /* 1 = Cache has a loop service pending */ #define CCR_INTERLEAVEDSAMPLES 0x00800000 /* 1 = A cache service will fetch interleaved samples */ /* Auto-set from CPF_STEREO_MASK */ #define CCR_WORDSIZEDSAMPLES 0x00400000 /* 1 = A cache service will fetch word sized samples */ /* Auto-set from CCCA_8BITSELECT */ -#define CCR_READADDRESS 0x06100009 -#define CCR_READADDRESS_MASK 0x003f0000 /* Next cached sample to play */ -#define CCR_LOOPINVALSIZE 0x0000fe00 /* Number of invalid samples in cache prior to loop */ +SUB_REG(CCR, READADDRESS, 0x003f0000) /* Next cached sample to play */ +SUB_REG(CCR, LOOPINVALSIZE, 0x0000fe00) /* Number of invalid samples in cache prior to loop */ /* NOTE: This is valid only if CACHELOOPFLAG is set */ #define CCR_LOOPFLAG 0x00000100 /* Set for a single sample period when a loop occurs */ -#define CCR_CACHELOOPADDRHI 0x000000ff /* CLP_LOOPSTARTADDR's hi byte if CACHELOOPFLAG is set */ +SUB_REG(CCR, CACHELOOPADDRHI, 0x000000ff) /* CLP_LOOPSTARTADDR's hi byte if CACHELOOPFLAG is set */ #define CLP 0x0a /* Cache loop register (valid if CCR_CACHELOOPFLAG = 1) */ /* NOTE: This register is normally not used */ -#define CLP_CACHELOOPADDR 0x0000ffff /* Cache loop address low word */ +SUB_REG(CLP, CACHELOOPADDR, 0x0000ffff) /* Cache loop address low word */ #define FXRT 0x0b /* Effects send routing register */ /* NOTE: It is illegal to assign the same routing to */ @@ -537,20 +540,17 @@ #define IP_UNITY 0x0000e000 /* Unity pitch shift */ #define IFATN 0x19 /* Initial filter cutoff and attenuation register */ -#define IFATN_FILTERCUTOFF_MASK 0x0000ff00 /* Initial filter cutoff frequency in exponential units */ +SUB_REG(IFATN, FILTERCUTOFF, 0x0000ff00) /* Initial filter cutoff frequency in exponential units */ /* 6 most significant bits are semitones */ /* 2 least significant bits are fractions */ -#define IFATN_FILTERCUTOFF 0x08080019 -#define IFATN_ATTENUATION_MASK 0x000000ff /* Initial attenuation in 0.375dB steps */ -#define IFATN_ATTENUATION 0x08000019 +SUB_REG(IFATN, ATTENUATION, 0x000000ff) /* Initial attenuation in 0.375dB steps */ #define PEFE 0x1a /* Pitch envelope and filter envelope amount register */ -#define PEFE_PITCHAMOUNT_MASK 0x0000ff00 /* Pitch envlope amount */ +SUB_REG(PEFE, PITCHAMOUNT, 0x0000ff00) /* Pitch envlope amount */ /* Signed 2's complement, +/- one octave peak extremes */ -#define PEFE_PITCHAMOUNT 0x0808001a -#define PEFE_FILTERAMOUNT_MASK 0x000000ff /* Filter envlope amount */ +SUB_REG(PEFE, FILTERAMOUNT, 0x000000ff) /* Filter envlope amount */ /* Signed 2's complement, +/- six octaves peak extremes */ -#define PEFE_FILTERAMOUNT 0x0800001a + #define FMMOD 0x1b /* Vibrato/filter modulation from LFO register */ #define FMMOD_MODVIBRATO 0x0000ff00 /* Vibrato LFO modulation depth */ @@ -793,22 +793,19 @@ #define SRCS_SPDIFRATE_96 0x00080000 #define MICIDX 0x63 /* Microphone recording buffer index register */ -#define MICIDX_MASK 0x0000ffff /* 16-bit value */ -#define MICIDX_IDX 0x10000063 +SUB_REG(MICIDX, IDX, 0x0000ffff) #define ADCIDX 0x64 /* ADC recording buffer index register */ -#define ADCIDX_MASK 0x0000ffff /* 16 bit index field */ -#define ADCIDX_IDX 0x10000064 +SUB_REG(ADCIDX, IDX, 0x0000ffff) #define A_ADCIDX 0x63 -#define A_ADCIDX_IDX 0x10000063 +SUB_REG(A_ADCIDX, IDX, 0x0000ffff) #define A_MICIDX 0x64 -#define A_MICIDX_IDX 0x10000064 +SUB_REG(A_MICIDX, IDX, 0x0000ffff) #define FXIDX 0x65 /* FX recording buffer index register */ -#define FXIDX_MASK 0x0000ffff /* 16-bit value */ -#define FXIDX_IDX 0x10000065 +SUB_REG(FXIDX, IDX, 0x0000ffff) /* The 32-bit HLIEx and HLIPx registers all have one bit per channel control/status */ #define HLIEL 0x66 /* Channel half loop interrupt enable low register */ @@ -852,8 +849,8 @@ #define A_SPDIF_44100 0x00000080 #define A_SPDIF_MUTED 0x000000c0 -#define A_I2S_CAPTURE_RATE_MASK 0x00000e00 /* This sets the capture PCM rate, but it is */ -#define A_I2S_CAPTURE_RATE 0x03090076 /* unclear if this sets the ADC rate as well. */ +SUB_REG_NC(A_EHC, A_I2S_CAPTURE_RATE, 0x00000e00) /* This sets the capture PCM rate, but it is */ + /* unclear if this sets the ADC rate as well. */ #define A_I2S_CAPTURE_48000 0x0 #define A_I2S_CAPTURE_192000 0x1 #define A_I2S_CAPTURE_96000 0x2 -- 2.40.0.152.g15d061e6df

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