May 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/6] ASoC: qdsp6: Add support to TDM ports
by Srinivas Kandagatla 29 May '18

29 May '18

This patchset adds supports to 40 TDM ports available in q6AFE module. Among these 6 patches, there are 2 cleanup patches, first one which makes i2s use common prepare function and the second one is add macros for routings to make it easy to add new port routings. Rest of the 4 patches are tdm dt-bindings, afe port prepare support in afe, afe-dais support and adding tdm routing update. These patches have been tested by Rohit on SDM845. Thanks, Srini Srinivas Kandagatla (6): ASoC: qdsp6: dt-bindings: Add q6afe tdm dt binding ASoC: qdsp6: qdafe: add support to tdm ports ASoC: qdsp6: q6afe-dai: use q6afe_dai_prepare() for MI2S ASoC: qdsp6: q6afe-dai: add support to tdm dais ASoC: qdsp6: q6routing: Add macros for mixers ASoC: qdsp6: q6routing: Add support to all TDM Mixers .../devicetree/bindings/sound/qcom,q6afe.txt | 68 ++ include/dt-bindings/sound/qcom,q6afe.h | 80 ++ sound/soc/qcom/qdsp6/q6afe-dai.c | 621 ++++++++++- sound/soc/qcom/qdsp6/q6afe.c | 429 +++++++- sound/soc/qcom/qdsp6/q6afe.h | 20 +- sound/soc/qcom/qdsp6/q6routing.c | 1084 ++++++++++---------- 6 files changed, 1733 insertions(+), 569 deletions(-) -- 2.16.2

2 12

[alsa-devel] [PATCH 1/1] Add aica firmware source
by Jason Self 29 May '18

29 May '18

This provides a separately-buildable copy of the aica firmware source. The documentation in alsa-firmware indicates aica_firmware.bin was built with code from http://linuxdc.cvs.sourceforge.net/linuxdc/linux-sh-dc/sound/oss/aica/firmw… While it's still possible to obtain the firmware source code from there using rsync the linuxdc project has been defunct for a number of years and it seems a good idea to get a copy of the source code stashed away somewhere else just in case. I don't know that we should necessarily expect people to have a cross compiler installed on their system and so this patch doesn't change any of the existing Makefiles so as to leave the default behavior as-is, but anyone interested in rebuilding aica_firmware.bin would at least have the option to do so. Signed-off-by: Jason Self <j(a)jxself.org> diff --git a/aica/src/COPYING b/aica/src/COPYING new file mode 100644 index 0000000..0019417 --- /dev/null +++ b/aica/src/COPYING @@ -0,0 +1,25 @@ +(c)2000 Dan Potter + +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +1. Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. +2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. +3. Neither the name of Cryptic Allusion nor the names of its contributors + may be used to endorse or promote products derived from this software + without specific prior written permission. + +THIS SOFTWARE IS PROVIDED BY THE AUTHORS AND CONTRIBUTORS ``AS IS'' AND +ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE +FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL +DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS +OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) +HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT +LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY +OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF +SUCH DAMAGE. diff --git a/aica/src/INSTALL b/aica/src/INSTALL new file mode 100644 index 0000000..2c51ccb --- /dev/null +++ b/aica/src/INSTALL @@ -0,0 +1,7 @@ +To build this you will need the following on the host system: + + * GNU Make + * GNU C cross-compiler for ARM: + - arm-none-eabi-gcc + - arm-none-eabi-objcopy + - arm-none-eabi-as \ No newline at end of file diff --git a/aica/src/aica.c b/aica/src/aica.c new file mode 100644 index 0000000..2dc5010 --- /dev/null +++ b/aica/src/aica.c @@ -0,0 +1,220 @@ +/* This file is part of the Dreamcast function library. + * Please see libdream.c for further details. + * + * (c)2000 Dan Potter + * + * $Id: aica.c,v 1.3 2002/10/29 15:43:38 mrbrown Exp $ + */ + +#include "aica.h" + +void aica_init() { + int i, j; + + /* Initialize AICA channels */ + SNDREG32(0x2800) = 0x0000; + + for (i=0; i<64; i++) { + for (j=0; j<0x80; j+=4) + CHNREG32(i, j) = 0; + CHNREG32(i,0) = 0x8000; + CHNREG32(i,20) = 0x1f; + } + + SNDREG32(0x2800) = 0x000f; +} + +/* Translates a volume from linear form to logarithmic form (required by + the AICA chip */ +/* int logs[] = { + +0, 40, 50, 58, 63, 68, 73, 77, 80, 83, 86, 89, 92, 94, 97, 99, 101, 103, +105, 107, 109, 111, 112, 114, 116, 117, 119, 120, 122, 123, 125, 126, 127, +129, 130, 131, 133, 134, 135, 136, 137, 139, 140, 141, 142, 143, 144, 145, +146, 147, 148, 149, 150, 151, 152, 153, 154, 155, 156, 156, 157, 158, 159, +160, 161, 162, 162, 163, 164, 165, 166, 166, 167, 168, 169, 170, 170, 171, +172, 172, 173, 174, 175, 175, 176, 177, 177, 178, 179, 180, 180, 181, 182, +182, 183, 183, 184, 185, 185, 186, 187, 187, 188, 188, 189, 190, 190, 191, +191, 192, 193, 193, 194, 194, 195, 196, 196, 197, 197, 198, 198, 199, 199, +200, 201, 201, 202, 202, 203, 203, 204, 204, 205, 205, 206, 206, 207, 207, +208, 208, 209, 209, 210, 210, 211, 211, 212, 212, 213, 213, 214, 214, 215, +215, 216, 216, 217, 217, 217, 218, 218, 219, 219, 220, 220, 221, 221, 222, +222, 222, 223, 223, 224, 224, 225, 225, 225, 226, 226, 227, 227, 228, 228, +228, 229, 229, 230, 230, 230, 231, 231, 232, 232, 232, 233, 233, 234, 234, +234, 235, 235, 236, 236, 236, 237, 237, 238, 238, 238, 239, 239, 240, 240, +240, 241, 241, 241, 242, 242, 243, 243, 243, 244, 244, 244, 245, 245, 245, +246, 246, 247, 247, 247, 248, 248, 248, 249, 249, 249, 250, 250, 250, 251, +251, 251, 252, 252, 252, 253, 253, 253, 254, 254, 254, 255 + +}; */ + +int logs[] = { + 0, 15, 22, 27, 31, 35, 39, 42, 45, 47, 50, 52, 55, 57, 59, 61, + 63, 65, 67, 69, 71, 73, 74, 76, 78, 79, 81, 82, 84, 85, 87, 88, + 90, 91, 92, 94, 95, 96, 98, 99, 100, 102, 103, 104, 105, 106, + 108, 109, 110, 111, 112, 113, 114, 116, 117, 118, 119, 120, 121, + 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 134, + 135, 136, 137, 138, 138, 139, 140, 141, 142, 143, 144, 145, 146, + 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 156, + 157, 158, 159, 160, 160, 161, 162, 163, 164, 164, 165, 166, 167, + 167, 168, 169, 170, 170, 171, 172, 173, 173, 174, 175, 176, 176, + 177, 178, 178, 179, 180, 181, 181, 182, 183, 183, 184, 185, 185, + 186, 187, 187, 188, 189, 189, 190, 191, 191, 192, 193, 193, 194, + 195, 195, 196, 197, 197, 198, 199, 199, 200, 200, 201, 202, 202, + 203, 204, 204, 205, 205, 206, 207, 207, 208, 209, 209, 210, 210, + 211, 212, 212, 213, 213, 214, 215, 215, 216, 216, 217, 217, 218, + 219, 219, 220, 220, 221, 221, 222, 223, 223, 224, 224, 225, 225, + 226, 227, 227, 228, 228, 229, 229, 230, 230, 231, 232, 232, 233, + 233, 234, 234, 235, 235, 236, 236, 237, 237, 238, 239, 239, 240, + 240, 241, 241, 242, 242, 243, 243, 244, 244, 245, 245, 246, 246, + 247, 247, 248, 248, 249, 249, 250, 250, 251, 251, 252, 252, 253, 254, 255 +}; +int vol_to_log(int vol) { +/* vol = 0xff - (vol & 0xff); */ +/* vol = 0xff - logs[vol & 0xff]; */ +/* vol = 128 - ((vol & 0xff) / 2); */ + vol = 0xff - logs[128 + ((vol & 0xff) / 2)]; + return vol; +} + +/* Sets up a sound channel completely. This is generally good if you want + a quick and dirty way to play notes. If you want a more comprehensive + set of routines (more like PC wavetable cards) see below. + + ch is the channel to play on (0 - 63) + smpptr is the pointer to the sound data; if you're running off the + SH4, then this ought to be (ptr - 0xa0800000); otherwise it's just + ptr. Basically, it's an offset into sound ram. + mode is one of the mode constants (16 bit, 8 bit, ADPCM) + nsamp is the number of samples to play (not number of bytes!) + freq is the sampling rate of the sound + vol is the volume, 0 to 0xff (0xff is louder) + pan is a panning constant -- 0 is left, 128 is center, 255 is right. + + This routine (and the similar ones) owe a lot to Marcus' sound example -- + I hadn't gotten quite this far into dissecting the individual regs yet. */ +void aica_play(int ch, unsigned long smpptr, int mode, int loopst, int loopend, + int freq, int vol, int pan, int loopflag) { + unsigned long freq_lo, freq_base = 5644800; + int freq_hi = 7; + int i, j; + + /* Stop the channel (if it's already playing) */ + /*for (i=CHNREG8(ch, 41); i<=0xff; i++) { + CHNREG8(ch, 41) = i; + } */ + aica_stop(ch); + for (i=0; i<256; i++) { + asm("nop"); + asm("nop"); + asm("nop"); + asm("nop"); + } + + /* Convert the incoming volume and pan into hardware values */ + vol = vol_to_log(vol); + + /* For the moment this is going to have to suffice, until we really + figure out what these mean. */ + if (pan == 0x80) + pan = 0; + else if (pan < 0x80) + pan = 0x1f; + else + pan = 0xf; + + /* Envelope setup. The first of these is the loop point, + e.g., where the sample starts over when it loops. The second + is the loop end. This is the full length of the sample when + you are not looping, or the loop end point when you are (though + storing more than that is a waste of memory if you're not doing + volume enveloping). */ + CHNREG32(ch, 8) = loopst & 0xffff; + CHNREG32(ch, 12) = loopend & 0xffff; + + /* Need to convert frequency to floating point format + (freq_hi is exponent, freq_lo is mantissa) + Formula is ferq = 44100*2^freq_hi*(1+freq_lo/1024) */ + while (freq < freq_base && freq_hi > -8) { + freq_base >>= 1; + --freq_hi; + } + freq_lo = (freq<<10) / freq_base; + + /* Write resulting values */ + CHNREG32(ch, 24) = (freq_hi << 11) | (freq_lo & 1023); + + /* Set volume, pan, and some other things that we don't know what + they do =) */ + CHNREG8(ch, 36) = pan; + CHNREG8(ch, 37) = 0xf; + CHNREG8(ch, 40) = 0x24; + CHNREG8(ch, 41) = vol; /* Vol starts at zero so we can ramp */ + + /* If we supported volume envelopes (which we don't yet) then + this value would set that up. The top 4 bits determine the + envelope speed. f is the fastest, 1 is the slowest, and 0 + seems to be an invalid value and does weird things). The + default (below) sets it into normal mode (play and terminate/loop). + CHNREG32(ch, 16) = 0xf010; + */ + CHNREG32(ch, 16) = 0x1f; /* No volume envelope */ + + + /* Set sample format, buffer address, and looping control. If + 0x0200 mask is set on reg 0, the sample loops infinitely. If + it's not set, the sample plays once and terminates. We'll + also set the bits to start playback here. */ + CHNREG32(ch, 4) = smpptr & 0xffff; + if (loopflag) + CHNREG32(ch, 0) = 0xc000 | 0x0200 | (mode<<7) | (smpptr >> 16); /* Loops */ + else + CHNREG32(ch, 0) = 0xc000 | 0x0000 | (mode<<7) | (smpptr >> 16); /* No loop */ + + + /* Enable playback */ + /* CHNREG32(ch, 0) |= 0xc000; */ + + /*for (i=0xff; i>=vol; i--) + CHNREG8(ch, 41) = i; */ +} + +/* Stop the sound on a given channel */ +void aica_stop(int ch) { + CHNREG32(ch, 0) = (CHNREG32(ch, 0) & ~0x4000) | 0x8000; +} + + +/* The rest of these routines can change the channel in mid-stride so you + can do things like vibrato and panning effects. */ + +/* Set channel volume */ +void aica_vol(int ch, int vol) { + vol = vol_to_log(vol); + CHNREG8(ch, 41) = vol; +} + +/* Set channel pan */ +void aica_pan(int ch, int pan) { + if (pan == 0x80) + pan = 0; + else if (pan < 0x80) + pan = 0x1f; + else + pan = 0xf; + CHNREG8(ch, 36) = pan; +} + +/* Set channel frequency */ +void aica_freq(int ch, int freq) { + unsigned long freq_lo, freq_base = 5644800; + int freq_hi = 7; + + while (freq < freq_base && freq_hi > -8) { + freq_base >>= 1; + freq_hi--; + } + freq_lo = (freq<<10) / freq_base; + CHNREG32(ch, 24) = (freq_hi << 11) | (freq_lo & 1023); +} + diff --git a/aica/src/aica.h b/aica/src/aica.h new file mode 100644 index 0000000..4c41e33 --- /dev/null +++ b/aica/src/aica.h @@ -0,0 +1,25 @@ +/* $Id: aica.h,v 1.3 2002/10/29 15:43:51 mrbrown Exp $ */ + +#ifndef __AICA_H +#define __AICA_H + +/* volatile unsigned char *dc_snd_base = (unsigned char *)0x00800000; */ +#define dc_snd_base ((volatile unsigned char *)0x00800000) + +/* Some convienence macros */ +#define SNDREG32A(x) ((volatile unsigned long *)(dc_snd_base + (x))) +#define SNDREG32(x) (*SNDREG32A(x)) +#define SNDREG8A(x) (dc_snd_base + (x)) +#define SNDREG8(x) (*SNDREG8A(x)) +#define CHNREG32A(chn, x) SNDREG32A(0x80*(chn) + (x)) +#define CHNREG32(chn, x) (*CHNREG32A(chn, x)) +#define CHNREG8A(chn, x) SNDREG8A(0x80*(chn) + (x)) +#define CHNREG8(chn, x) (*CHNREG8A(chn, x)) + +/* Sound modes */ +#define SM_8BIT 1 +#define SM_16BIT 0 +#define SM_ADPCM 2 + +#endif /* __AICA_H */ + diff --git a/aica/src/aica_cmd_iface.h b/aica/src/aica_cmd_iface.h new file mode 100644 index 0000000..df1cac9 --- /dev/null +++ b/aica/src/aica_cmd_iface.h @@ -0,0 +1,29 @@ +/* $Id: aica_cmd_iface.h,v 1.2 2002/10/29 15:45:10 mrbrown Exp $ */ + +#ifndef __ARM_AICA_CMD_IFACE_H +#define __ARM_AICA_CMD_IFACE_H + +#ifndef __ARCH_TYPES_H +typedef unsigned long uint32; +#endif + +/* Make this 8 dwords long for one aica bus queue */ +typedef struct { + uint32 cmd; /* Command ID */ + uint32 pos; /* Sample position */ + uint32 length; /* Sample length */ + uint32 freq; /* Frequency */ + uint32 vol; /* Volume 0-255 */ + uint32 pan; /* Pan 0-255 */ + uint32 sfmt; /* Sound format */ + uint32 flags; /* flags */ +} aica_channel; + +/* Command values */ +#define AICA_CMD_KICK 0x80000000 +#define AICA_CMD_NONE 0 +#define AICA_CMD_START 1 +#define AICA_CMD_STOP 2 +#define AICA_CMD_VOL 3 + +#endif /* __ARM_AICA_CMD_IFACE_H */ diff --git a/aica/src/crt0.s b/aica/src/crt0.s new file mode 100644 index 0000000..438f9c1 --- /dev/null +++ b/aica/src/crt0.s @@ -0,0 +1,104 @@ +# Adapted from Marcus' AICA example among a few other sources =) +# +# $Id: crt0.s,v 1.3 2002/10/29 15:45:49 mrbrown Exp $ + +.text +.globl arm_main +.globl timer +.globl jps + +# Exception vectors + b start + b undef + b softint + b pref_abort + b data_abort + b rsrvd + b irq + + +# FIQ code adapted from the Marcus AICA example +fiq: + # Save regs + #stmdb sp!, {r0-r14} + + # Grab interrupt type (store as parameter) + ldr r8,intreq + ldr r9,[r8] + and r9,r9,#7 + cmp r9,#2 + bne fiq_done + + # Type 2 is timer interrupt. Increment timer variable. + adr r8,timer + ldr r9,[r8] + add r9,r9,#1 + str r9,[r8] + + # Request a new timer interrupt. We'll calculate the number + # put in here based on the "jps" (jiffies per second). + ldr r8, timer_control + mov r9,#256-(44100/4410) +# ldr r9,jps + str r9,[r8,#0x10] + mov r9,#0x40 + str r9,[r8,#0x24] + b fiq_done + + # Return from interrupt +fiq_done: + + # Clear interrupt + ldr r8,intclr + mov r9,#1 + str r9,[r8] + str r9,[r8] + str r9,[r8] + str r9,[r8] + + # Restore regs and return + #ldmdb sp!, {r0-r14} + subs pc,r14,#4 + +intreq: + .long 0x00802d00 +intclr: + .long 0x00802d04 +timer_control: + .long 0x00802880 +timer: + .long 0 +jps: + .long 256-(44100/1000) + + +start: + # Setup a basic stack, disable IRQ, enable FIQ + mov sp,#0xb000 + mrs r10,CPSR + orr r10,r10,#0x80 + bic r10,r10,#0x40 + msr CPSR_all,r10 + + # Call the main for the SPU + bl arm_main + + # Loop infinitely if we get here +here: b here + + +# Handlers we don't bother to catch +undef: +softint: + mov pc,r14 +pref_abort: +data_abort: +irq: +rsrvd: + sub pc,r14,#4 + + + + + + diff --git a/aica/src/main.c b/aica/src/main.c new file mode 100644 index 0000000..2a50482 --- /dev/null +++ b/aica/src/main.c @@ -0,0 +1,123 @@ +/* Streaming sound driver + * + * (c)2000 Dan Potter + * + * This slightly more complicated version allows for sound effect channels, + * and full sampling rate, panning, and volume control for each. The two + * streaming channels are still always first and always occur at 0x11000 and + * 0x21000. All other sample data can begin at 0x31000. "pos" only works for + * input on the two streaming channels (which will always have the same + * "pos" value). + * + * $Id: main.c,v 1.3 2002/10/29 15:48:47 mrbrown Exp $ + */ + +#include "aica.h" +#include "aica_cmd_iface.h" + +static int buffer_index; +static int active; + +/****************** Timer *******************************************/ + +extern volatile int timer; + +void timer_wait(int jiffies) { + int fin = timer + jiffies; + while (timer <= fin) + ; +} + +/****************** Main Program ************************************/ + +/* Set channel id at 0x80280d (byte), read position at 0x802814 (long) */ + +volatile uint32 *cmd = (volatile uint32 *)0x10000; +volatile aica_channel *chans = (volatile aica_channel *)0x10004; + +void start_channel(int chn) { + int samplesize = 1; + int stereo = (chans[chn].flags & 1); + if (chans[chn].sfmt == SM_16BIT) samplesize = 2; + if (stereo) + { + aica_play(0, 0x11000 , chans[chn].sfmt, 0, 0x8000/samplesize, chans[chn].freq, chans[chn].vol, 0, 1); + aica_play(1, 0x21000, chans[chn].sfmt, 0, 0x8000/samplesize, chans[chn].freq, chans[chn].vol, 0xff, 1); + } +else + { + aica_play(chn, 0x11000, chans[chn].sfmt, 0, 0x8000/samplesize, chans[chn].freq, chans[chn].vol, chans[chn].pan, 1); + } +} + +void stop_channel(int chn) { + if (chn == 0) { + aica_stop(0); + aica_stop(1); + } else { + aica_stop(chn); + } +} + +void vol_channel(int chn) { + if (chn == 0) { + aica_vol(0, chans[chn].vol); + aica_vol(1, chans[chn].vol); + } else { + aica_vol(chn, chans[chn].vol); + } +} + +void process_cmd(uint32 cmd) { + /* cmd is channel to look at +1 */ + cmd--; + switch(chans[cmd].cmd) { + case AICA_CMD_NONE: + break; + case AICA_CMD_START: + start_channel(cmd); + active = 0; + break; + case AICA_CMD_STOP: + stop_channel(cmd); + active = 0; + break; + case AICA_CMD_VOL: + vol_channel(cmd); + break; + } +} + +int arm_main() { + int cmdl; + int flags, shift_buffer, pass; + /* Initialize the AICA part of the SPU */ + aica_init(); + + + /* Observe channel 0 */ + SNDREG8(0x280d) = 0; + + /* Wait for a command */ + while(1) { + /* Check for a command */ + cmdl = *cmd; + if (cmdl & AICA_CMD_KICK) { + *cmd = 0; + process_cmd(cmdl & ~AICA_CMD_KICK); + } + + /* Update position counters */ + chans[0].pos = SNDREG32(0x2814); + + /* Little delay to prevent memory lock */ + timer_wait(10); + + } +} + + + + + + -- 2.7.4

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[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Fixup mute led on HP Spectre x360
by Takashi Iwai 29 May '18

29 May '18

From: Tom Briden <tom(a)decompile.me.uk> This patch adds the mute LED control for HP Spectre x360 Kabylake model. The mute LED is controlled via VREF bits on NID 0x1b, so we need a new fixup function. Note that this doesn't fix the other issues like the missing speaker output on the machine. They will be addressed by later patches. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=189331 Signed-off-by: Tom Briden <tom(a)decompile.me.uk> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3af93841abe1..6369964c117a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3679,6 +3679,19 @@ static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, } } +static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_polarity = 0; + spec->mute_led_nid = 0x1b; + spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; + spec->gen.vmaster_mute_enum = 1; + codec->power_filter = led_power_filter; + } +} + /* update LED status via GPIO */ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, bool enabled) @@ -5413,6 +5426,7 @@ enum { ALC269_FIXUP_HP_MUTE_LED, ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC269_FIXUP_HP_MUTE_LED_MIC2, + ALC269_FIXUP_HP_MUTE_LED_MIC3, ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_HP_GPIO_MIC1_LED, ALC269_FIXUP_HP_LINE1_MIC1_LED, @@ -5672,6 +5686,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led_mic2, }, + [ALC269_FIXUP_HP_MUTE_LED_MIC3] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_hp_mute_led_mic3, + }, [ALC269_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_gpio_led, @@ -6494,6 +6512,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), + SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), -- 2.17.0

1 1

[alsa-devel] [PATCH 1/1] Fix up aica firmware licensing
by Jason Self 29 May '18

29 May '18

The 3-clause BSD that this firmware uses requires more than just keeping the copyright notice, like license.txt currently says. Rather, it has has that "redistributions in binary form (which is what the ALSA Project is doing, since only the firmware binary is provided) must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution." This means that more than just the copyright notice needs to be included. This change is intended to address this by adding the parts that were omitted originally. Signed-off-by: Jason Self <j(a)jxself.org> diff --git a/aica/license.txt b/aica/license.txt index 4b4cdbc..9de48bd 100644 --- a/aica/license.txt +++ b/aica/license.txt @@ -1,8 +1,34 @@ This firmware is licenced under the KOS licence - a modified form of the BSD licence (for details see http://gamedev.allusion.net/softprj/kos/faq.php). -It is fully and freely distributable provided the following notice is preserved somewhere -in the code: Copyright (c) Dan Potter 2000. +It is fully and freely distributable provided the following notices +are preserved: + +Copyright (c) Dan Potter 2000. + +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: +1. Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. +2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. +3. Neither the name of Cryptic Allusion nor the names of its contributors + may be used to endorse or promote products derived from this software + without specific prior written permission. + +THIS SOFTWARE IS PROVIDED BY THE AUTHORS AND CONTRIBUTORS ``AS IS'' AND +ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE +IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE +ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE +FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL +DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS +OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) +HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT +LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY +OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF +SUCH DAMAGE. The code this firmware was built with is part of the KOS distribution and can also be reviewed here: http://linuxdc.cvs.sourceforge.net/linuxdc/linux-sh-dc/sound/oss/aica/firmw… -- 2.7.4

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[alsa-devel] out of sync capture samples with MMAP on MIPS
by Daniel Danzberger 29 May '18

29 May '18

I am using alsa's mmap() interface to capture audio samples on some AR71xx MIPS devices via USB and sometimes the sampled audio sounds very disordered. After debugging deeper into this issue I found that samples are fine in the kernel's vmalloc'ed buffer (runtime->dma_area). But once they get accessed from userspace via mmaped memory, the samples differ. I wrote a module + userspace program to demonstrate the issue: git@github.com:dddaniel/mmaptest.git The userspace tool reads the data a couple of times with a small delay. As you see below, after a couple of reads the data gets closer to look like in the kernel buffer: --- root@OpenWrt:/# mmaptest-user mmap addr: 0x77a04000 [ 760.464968] mmap page 7573000 at va 87573000 check memory ...FAIL (at byte 0) check memory ...FAIL (at byte 96) check memory ...FAIL (at byte 96) check memory ...FAIL (at byte 96) check memory ...FAIL (at byte 128) --- Does someone here experienced similar issues ?

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[alsa-devel] [PATCH 1/3] ALSA: hda: Add Acer Aspire 5930G to the power_save blacklist
by Hans de Goede 29 May '18

29 May '18

Power-saving is causing plopping sounds on the Acer Aspire 5930G, add it to the blacklist. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b0c8c79848a9..7ca29b890425 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2206,6 +2206,8 @@ static int azx_probe(struct pci_dev *pci, * to causes problems on these devices. */ static struct snd_pci_quirk power_save_blacklist[] = { + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ -- 2.17.0

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[alsa-devel] [PATCH] ALSA: xen: ensure nul-terminated device name
by Arnd Bergmann 29 May '18

29 May '18

gcc-8 warns that pcm_instance->name is not necessarily terminated correctly if the input is more than 80 characters long or lacks a termination byte itself: In function 'strncpy', inlined from 'cfg_device' at sound/xen/xen_snd_front_cfg.c:399:3, inlined from 'xen_snd_front_cfg_card' at sound/xen/xen_snd_front_cfg.c:509:9: include/linux/string.h:254:9: error: '__builtin_strncpy' specified bound 80 equals destination size [-Werror=stringop-truncation] return __builtin_strncpy(p, q, size); Using strlcpy() instead of strncpy() makes this a bit safer. Fixes: fd3b36045c2c ("ALSA: xen-front: Read sound driver configuration from Xen store") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/xen/xen_snd_front_cfg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/xen/xen_snd_front_cfg.c b/sound/xen/xen_snd_front_cfg.c index 38c7e1eefbb9..684b5f1d51ac 100644 --- a/sound/xen/xen_snd_front_cfg.c +++ b/sound/xen/xen_snd_front_cfg.c @@ -396,7 +396,7 @@ static int cfg_device(struct xen_snd_front_info *front_info, str = xenbus_read(XBT_NIL, device_path, XENSND_FIELD_DEVICE_NAME, NULL); if (!IS_ERR(str)) { - strncpy(pcm_instance->name, str, sizeof(pcm_instance->name)); + strlcpy(pcm_instance->name, str, sizeof(pcm_instance->name)); kfree(str); } -- 2.9.0

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[alsa-devel] [PATCH] ALSA: usb-audio: Allow non-vmalloc buffer for PCM buffers
by Takashi Iwai 29 May '18

29 May '18

Currently, USB-audio driver allocates the PCM buffer via vmalloc(), as this serves merely as an intermediate buffer that is copied to each URB transfer buffer. This works well in general on x86, but on some archs this may result in cache coherency issues when mmap is used. OTOH, it works also on such arch unless mmap is used. This patch is a step for mitigating the inconvenience; a new module option "use_vmalloc" is provided so that user can choose to allocate the DMA coherent buffer instead of the existing vmalloc buffer. The drawback is that it'd be the standard dma_alloc_coherent() calls and the system would require contiguous pages on non-x86 archs. Note that it's a global option and not dynamically switchable since the buffer is pre-allocated at the probe time. In theory, it's possible to be switchable, but it'd be trickier and racier. As default use_vmalloc option is set to true, so that the old behavior is kept. For allowing the coherent mmap on ARM or MIPS, pass use_vmalloc=0 option explicitly. Reported-by: Daniel Danzberger <daniel(a)dd-wrt.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa-configuration.rst | 7 ++++ sound/usb/card.c | 4 ++ sound/usb/pcm.c | 59 +++++++++++++++++++++++++++--- sound/usb/pcm.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 2 + 6 files changed, 70 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index aed6b4fb8e46..b1052e18292d 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2224,6 +2224,13 @@ quirk_alias Quirk alias list, pass strings like ``0123abcd:5678beef``, which applies the existing quirk for the device 5678:beef to a new device 0123:abcd. +use_vmalloc + Use vmalloc() for allocations of the PCM buffers (default: yes). + For architectures with non-coherent memory like ARM or MIPS, the + mmap access may give inconsistent results with vmalloc'ed + buffers. If mmap is used on such architectures, turn off this + option, so that the DMA-coherent buffers are allocated and used + instead. This module supports multiple devices, autoprobe and hotplugging. diff --git a/sound/usb/card.c b/sound/usb/card.c index c80224807e8f..a1ed798a1c6b 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -86,6 +86,8 @@ static bool ignore_ctl_error; static bool autoclock = true; static char *quirk_alias[SNDRV_CARDS]; +bool snd_usb_use_vmalloc = true; + module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); module_param_array(id, charp, NULL, 0444); @@ -105,6 +107,8 @@ module_param(autoclock, bool, 0444); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); 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); +MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes)."); /* * we keep the snd_usb_audio_t instances by ourselves for merging diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 897a2cbef6de..78d1cad08a0a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -728,7 +728,11 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + if (snd_usb_use_vmalloc) + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + else + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (ret < 0) return ret; @@ -781,7 +785,11 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) snd_usb_endpoint_deactivate(subs->data_endpoint); snd_usb_unlock_shutdown(subs->stream->chip); } - return snd_pcm_lib_free_vmalloc_buffer(substream); + + if (snd_usb_use_vmalloc) + return snd_pcm_lib_free_vmalloc_buffer(substream); + else + return snd_pcm_lib_free_pages(substream); } /* @@ -1702,9 +1710,50 @@ static const struct snd_pcm_ops snd_usb_capture_ops = { .mmap = snd_pcm_lib_mmap_vmalloc, }; +static const struct snd_pcm_ops snd_usb_playback_dev_ops = { + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_usb_hw_params, + .hw_free = snd_usb_hw_free, + .prepare = snd_usb_pcm_prepare, + .trigger = snd_usb_substream_playback_trigger, + .pointer = snd_usb_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, +}; + +static const struct snd_pcm_ops snd_usb_capture_dev_ops = { + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_usb_hw_params, + .hw_free = snd_usb_hw_free, + .prepare = snd_usb_pcm_prepare, + .trigger = snd_usb_substream_capture_trigger, + .pointer = snd_usb_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, +}; + void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream) { - snd_pcm_set_ops(pcm, stream, - stream == SNDRV_PCM_STREAM_PLAYBACK ? - &snd_usb_playback_ops : &snd_usb_capture_ops); + const struct snd_pcm_ops *ops; + + if (snd_usb_use_vmalloc) + ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? + &snd_usb_playback_ops : &snd_usb_capture_ops; + else + ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? + &snd_usb_playback_dev_ops : &snd_usb_capture_dev_ops; + snd_pcm_set_ops(pcm, stream, ops); +} + +void snd_usb_preallocate_buffer(struct snd_usb_substream *subs) +{ + struct snd_pcm *pcm = subs->stream->pcm; + struct snd_pcm_substream *s = pcm->streams[subs->direction].substream; + struct device *dev = subs->dev->bus->controller; + + if (!snd_usb_use_vmalloc) + snd_pcm_lib_preallocate_pages(s, SNDRV_DMA_TYPE_DEV_SG, + dev, 64*1024, 512*1024); } diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index 35740d5ef268..f77ec58bf1a1 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -10,6 +10,7 @@ void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt); +void snd_usb_preallocate_buffer(struct snd_usb_substream *subs); #endif /* __USBAUDIO_PCM_H */ diff --git a/sound/usb/stream.c b/sound/usb/stream.c index d16e1c23f4e9..729afd808cc4 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -106,6 +106,8 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->ep_num = fp->endpoint; if (fp->channels > subs->channels_max) subs->channels_max = fp->channels; + + snd_usb_preallocate_buffer(subs); } /* kctl callbacks for usb-audio channel maps */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 7b28cbde22c0..b9faeca645fd 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -127,4 +127,6 @@ struct snd_usb_audio_quirk { int snd_usb_lock_shutdown(struct snd_usb_audio *chip); void snd_usb_unlock_shutdown(struct snd_usb_audio *chip); +extern bool snd_usb_use_vmalloc; + #endif /* __USBAUDIO_H */ -- 2.16.3

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Re: [alsa-devel] Echo Audiofire 12 issues
by Takashi Sakamoto 29 May '18

29 May '18

Hi, (C.C.ed alsa-devel) On May 28 2018 07:05, Adriano Michael Petrosillo wrote: > Hi, > > I'm contacting you regarding a few issues with my Echo Audiofire 12. It > has the latest firmware (5.8). With a recent kernel (4.15.0-22-generic) > it is recognised by the OS but it works unreliably, leading to system > instability. Please explain about your experiences which made you be unreliable and feel instability to your system so that the others such as me can get them. > Here is the dmesg output: > > [ 6019.402918] firewire_core 0000:04:07.0: PHY ID mismatch in self ID: 0 > != 1 > [ 6019.402931] firewire_core 0000:04:07.0: topology build failed > [ 6019.934272] firewire_core 0000:04:07.0: rediscovered device fw0 > [ 6065.194414] firewire_core 0000:04:07.0: phy config: new root=ffc1, > gap_count=5 > [ 6065.700801] firewire_core 0000:04:07.0: created device fw1: GUID > 0014865e75918c0e, S400 > [ 6067.619626] snd-fireworks fw1.0: Invalid CIP header for AMDTP: > 00000000:00000000 > [ 6067.619663] snd-fireworks fw1.0: Invalid CIP header for AMDTP: > 00000000:00000000 > [ 6067.619670] snd-fireworks fw 1.0: Invalid CIP header for AMDTP: > 00000000:00000000 > [ 6067.619676] snd-fireworks fw1.0: Invalid CIP header for AMDTP: > 00000000:00000000 > [ 6067.619683] snd-fireworks fw1.0: Invalid CIP header for AMDTP: > 00000000:00000000 > [ 6067.619689] snd-fireworks fw1.0: Invalid CIP header for AMDTP: > 00000000:00000000 > [ 6067.619695] snd-fireworks fw1.0: Invalid CIP header for AMDTP: > 00000000:00000000 > [ 6067.619701] snd-fireworks fw1.0: Invalid CIP header for AMDTP: > 00000000:00000000 > [ 6067.619706] snd-fireworks fw1.0: Invalid CIP header for AMDTP: > 00000000:00000000 > [ 6067.619712] snd-fireworks fw1.0: Invalid CIP header for AMDTP: > 00000000:00000000 In the beginning of packet streaming, your unit transfers packets with invalid CIP header, as shown. But enough later it starts to transfer the packet correctly. The above message is irrelevant to your issue. > It is also impossible to use as a system card, and alsamixer is unable > to set the volumes for the 12 channels, but here I'm probably doing > something wrong. If you want to improve support for the card, I can > cooperate, I have no coding skills unfor tunately. ALSA fireworks driver adds no control element set in Linux system. This is a design not to bring complexity into kernel stuffs. For control feature to DSP on your unit, please use 'hinawa-fireworks-cui' in 'hinawa-utils'[1] with 'libhinawa'[2], or ffado-dbus-server/ffado-mixer[3]. [1] https://github.com/takaswie/hinawa-utils [2] https://github.com/takaswie/libhinawa [3] http://subversion.ffado.org/ Regards Takashi Sakamoto

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[alsa-devel] [PATCH] Fixed typo on ac97_fc_rec_volume_put.
by Guy Chronister 28 May '18

28 May '18

Signed-off-by: Guy Chronister <guylovesbritt(a)gmail.com> --- sound/pci/oxygen/oxygen_mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 4ca12665ff73..64a82688d7fb 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -730,7 +730,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl, oldreg = oxygen_read_ac97(chip, 1, AC97_REC_GAIN); newreg = oldreg & ~0x0707; newreg = newreg | (value->value.integer.value[0] & 7); - newreg = newreg | ((value->value.integer.value[0] & 7) << 8); + newreg = newreg | ((value->value.integer.value[1] & 7) << 8); change = newreg != oldreg; if (change) oxygen_write_ac97(chip, 1, AC97_REC_GAIN, newreg); -- 2.17.0

2 1