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

Re: [alsa-devel] A bug about cache inconsistency report
by Hans Hu(SH-RD) 13 Aug '18

13 Aug '18

> > > > >Then the next step would be to fake sg-buffer from this > > > > >straight buffer. Revert the above, and modify sgbuf.c to the following: > > > > >- Allocate a large continuous buffer > > > > >- Assign each page in this large buffer > > > > > > > > >If this still works, it's not about vmap, but it just means > > > > >that the physically ordered pages do matter -- implicitly > > > > >showing that the snooping behavior isn't properly turned on / off on the controller. > > > > > > > > To fake SG-buffer, I did this test: restore all the codes to the original, then added some codes in snd_malloc_sgbuf_pages() like below, the result is badly niose. > > > > > > >Not really what I meant. Rather something like below (totally untested). > > > > > > [Hans :] I know what you mean now and complete code like below, but the result is still noise. > > > > >OK, so indeed the vmapped address does seem matter. Interesting. > > >What kind of user access does produce the noise? Does it via aplay mmap mode, too? > > > > >In anyway, if the vmap is a problem, it might be worked around a patch like below (again totally untested and not sure whether it's correct). > > > > [Hans :] As I know, In the hardware layer HDAC’s stream have two data transport path : non-snoop & snoop; In the software layer ALSA-Driver have two data transport path : mmap & not mmap(test shows, it is dependent on wav's format or mmap_flag in aplay.c). When hardware at non-snoop mode, without hardware module's help, software must mark the mem to uncacheable type: when mmap used, the mark action happened at pcm_mmap_prepare(), the not-mmap mode's mark action happened at __mark_pages_wc(). And when at not-mmap mode, the vmapped address directly used in snd_pcm_lib_write_transfer() -> copy_from_user(). > > > >Well, that's usually no problem regarding that cache coherency. > >At least it hasn't been any issue with Intel and AMD CPUs. > >Does the problem happen with Intel CPU instead of VIA? > > [Hans :] I don't have enough Intel/AMD machine here, up to now, only find one Intel's : HW: mother board is Dell 042P49, HDA controller is 8086:1c20, codec is cx20641. >And the same issue is seen on that machine? [Hans :] Yes Thanks, Hans > > Yes, of course, this path combinations(non-snoop + not-mmap) unlikely be used by most music players. > > > > [Hans :] I complete the patch in your last mail like below, not used the vunmap & re-vmap way(it has a little problem). Result is OK now(no noise anymore). > > >What problem did you get with vunmap / vmap? > > [Hans :] if patch like below, even the new vmapped address new_area be assigned to dmab->area, it will not use this time, but next. This time used dmab->area still the last one, but the old one had been vunmap, so Oops reported. >That's bad. But the problem isn't as you guessed, I guess. >The allocator re-uses the pre-allocated buffer if available, and this makes things inconsistent. I'll cook later. 保密声明：本邮件含有保密或专有信息，仅供指定收件人使用。严禁对本邮件或其内容做任何未经授权的查阅、使用、复制或转发。 CONFIDENTIAL NOTE: This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited.

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[alsa-devel] [PATCH] Suppress udev detection of sound card for some units on IEEE 1394 bus
by Takashi Sakamoto 11 Aug '18

11 Aug '18

A bug was filed to bugzilla.kernel.org for a quirk of some models which ALSA BeBoB driver supports. Bug 199365 - repeating bus resets on Firewire bus with Focusrite Saffaire 26/io https://bugzilla.kernel.org/show_bug.cgi?id=199365 Some models (two models as long as I know) have a quirk to disappear from IEEE 1394 bus at disconnections of packet streaming. Corresponding character devices are removed according to 'remove' callbacks of relevant drivers from Linux dd core. Then the models re-appear on the bus by generating bus resets and corresponding character devices are added according to 'probe' callbacks from Linux dd core. In a view of ALSA applications, this looks that plug-out/plug-in occur in a sequential order for the models when they stop playback/capture substream. For most applications, this doesn't cause large issue. However, this quirk is not good for combination of below modules in PulseAudio. PulseAudio enters endless loop to detect the models and start/stop PCM substream. - module-udev-detect - module-alsa-card - module-suspend-on-idle In detail, please read my comment no.6: https://bugzilla.kernel.org/show_bug.cgi?id=199365#c6 This commit suppressed udev detection of sound card for the issued models. For the models, 'PULSE_IGNORE' flag is added to udev rules, then module-udev-detect don't handle the models and PulseAudio never uses the models automatically. In a scenario for users to load module-alsa-card/module-alsa-sink/module-alsa-source by hand, although these modules can still stop PCM substreams with module-suspend-on-idle, PulseAudio never enters the endless loop because udev detection doesn't work for the models. In this case, as long as special files for ALSA character devices for these models are the same, corresponding sinks and sources are available even if the voluntary plug-out/plug-in occur. (Focusrite Saffire Pro 10 i/o with systemd 237) $ udevadm info -q all -p /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 P: /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 E: DEVPATH=/devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 E: ID_BUS=firewire E: ID_FOR_SEAT=sound-pci-0000_00_07_0 E: ID_ID=firewire-0x00130e01000606e0 E: ID_MODEL=Pro10IO E: ID_MODEL_FROM_DATABASE=XIO2213A/B/XIO2221 IEEE-1394b OHCI Controller [Cheetah Express] E: ID_MODEL_ID=0x000006 E: ID_PATH=pci-0000:00:07.0 E: ID_PATH_TAG=pci-0000_00_07_0 E: ID_PCI_CLASS_FROM_DATABASE=Serial bus controller E: ID_PCI_INTERFACE_FROM_DATABASE=OHCI E: ID_PCI_SUBCLASS_FROM_DATABASE=FireWire (IEEE 1394) E: ID_SERIAL=0x00130e01000606e0 E: ID_SERIAL_SHORT=0x00130e01000606e0 E: ID_VENDOR=Focusrite E: ID_VENDOR_FROM_DATABASE=Texas Instruments E: ID_VENDOR_ID=0x00130e E: SOUND_INITIALIZED=1 E: SUBSYSTEM=sound E: SYSTEMD_WANTS=sound.target E: TAGS=:systemd:seat: E: USEC_INITIALIZED=957089064 (Focusrite Saffire Pro 26 i/o with systemd 237) $ udevadm info -q all -p /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 P: /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 E: DEVPATH=/devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 E: ID_BUS=firewire E: ID_FOR_SEAT=sound-pci-0000_00_07_0 E: ID_ID=firewire-0x00130e0100030cdd E: ID_MODEL=Pro26IO E: ID_MODEL_FROM_DATABASE=XIO2213A/B/XIO2221 IEEE-1394b OHCI Controller [Cheetah Express] E: ID_MODEL_ID=0x000003 E: ID_PATH=pci-0000:00:07.0 E: ID_PATH_TAG=pci-0000_00_07_0 E: ID_PCI_CLASS_FROM_DATABASE=Serial bus controller E: ID_PCI_INTERFACE_FROM_DATABASE=OHCI E: ID_PCI_SUBCLASS_FROM_DATABASE=FireWire (IEEE 1394) E: ID_SERIAL=0x00130e0100030cdd E: ID_SERIAL_SHORT=0x00130e0100030cdd E: ID_VENDOR=Focusrite E: ID_VENDOR_FROM_DATABASE=Texas Instruments E: ID_VENDOR_ID=0x00130e E: SOUND_INITIALIZED=1 E: SUBSYSTEM=sound E: SYSTEMD_WANTS=sound.target E: TAGS=:systemd:seat: E: USEC_INITIALIZED=1071026684 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- .../alsa/mixer/profile-sets/90-pulseaudio.rules | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules b/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules index ceb61d9e5..19b826b52 100644 --- a/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules +++ b/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules @@ -19,6 +19,7 @@ SUBSYSTEM!="sound", GOTO="pulseaudio_end" ACTION!="change", GOTO="pulseaudio_end" KERNEL!="card*", GOTO="pulseaudio_end" SUBSYSTEMS=="usb", GOTO="pulseaudio_check_usb" +SUBSYSTEMS=="firewire", GOTO="pulseaudio_firewire_quirk" SUBSYSTEMS=="platform", DRIVERS=="thinkpad_acpi", ENV{PULSE_IGNORE}="1" @@ -109,4 +110,13 @@ ATTRS{idVendor}=="1038", ATTRS{idProduct}=="1250", ENV{PULSE_PROFILE_SET}="steel ATTRS{idVendor}=="1038", ATTRS{idProduct}=="1260", ENV{PULSE_PROFILE_SET}="steelseries-arctis-7-usb-audio.conf" ATTRS{idVendor}=="147a", ATTRS{idProduct}=="e055", ENV{PULSE_PROFILE_SET}="cmedia-high-speed-true-hdaudio.conf" +GOTO="pulseaudio_end" + +LABEL="pulseaudio_firewire_quirk" + +# Focusrite Saffire Pro 10/26 i/o has a quirk to disappear from IEEE 1394 bus when losing connections. +# https://bugzilla.kernel.org/show_bug.cgi?id=199365 +ENV{ID_VENDOR_ID}=="0x00130e", ENV{ID_MODEL_ID}=="0x000003", ENV{PULSE_IGNORE}="1" +ENV{ID_VENDOR_ID}=="0x00130e", ENV{ID_MODEL_ID}=="0x000006", ENV{PULSE_IGNORE}="1" + LABEL="pulseaudio_end" -- 2.17.1

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[alsa-devel] [PATCH] ASoC: adau17x1: Implemented safeload support
by Robert Rosengren 10 Aug '18

10 Aug '18

From: Danny Smith <dannys(a)axis.com> Safeload support has been implemented which is used when updating for instance filter parameters using alsa controls. Without safeload support audio can become distorted during update. Signed-off-by: Danny Smith <dannys(a)axis.com> Signed-off-by: Robert Rosengren <robertr(a)axis.com> --- sound/soc/codecs/adau17x1.c | 69 +++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/adau17x1.h | 1 + 2 files changed, 68 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..342ed0e04b1d 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -26,6 +26,10 @@ #include "adau17x1.h" #include "adau-utils.h" +#define ADAU17X1_SAFELOAD_TARGET_ADDRESS 0x0006 +#define ADAU17X1_SAFELOAD_TRIGGER 0x0007 +#define ADAU17X1_SAFELOAD_DATA(i) (0x0001 + (i)) + static const char * const adau17x1_capture_mixer_boost_text[] = { "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3", }; @@ -326,6 +330,18 @@ bool adau17x1_has_dsp(struct adau *adau) } EXPORT_SYMBOL_GPL(adau17x1_has_dsp); +bool adau17x1_has_safeload(struct adau *adau) +{ + switch (adau->type) { + case ADAU1761: + case ADAU1781: + return true; + default: + return false; + } +} +EXPORT_SYMBOL_GPL(adau17x1_has_safeload); + static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { @@ -957,6 +973,50 @@ int adau17x1_resume(struct snd_soc_component *component) } EXPORT_SYMBOL_GPL(adau17x1_resume); +static int adau17x1_safeload(struct sigmadsp *sigmadsp, unsigned int addr, + const uint8_t bytes[], size_t len) +{ + uint8_t buf[4]; + unsigned int i; + int ret; + + /* target address is offset by 1 */ + unsigned int addr_offset = addr - 1; + + /* write safeload addresses */ + for (i = 0; i < len / 4; i++) { + memcpy(buf, bytes + i * 4, 4); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_DATA(i), buf, 4); + if (ret < 0) + return ret; + } + + /* write target address */ + buf[0] = (addr_offset >> 24) & 0xff; + buf[1] = (addr_offset >> 16) & 0xff; + buf[2] = (addr_offset >> 8) & 0xff; + buf[3] = (addr_offset) & 0xff; + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TARGET_ADDRESS, buf, 4); + if (ret < 0) + return ret; + + /* write nbr of words to trigger address */ + buf[0] = 0x00; + buf[1] = 0x00; + buf[2] = 0x00; + buf[3] = i & 0xff; + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TRIGGER, buf, 4); + + return ret; +} + +static const struct sigmadsp_ops adau17x1_sigmadsp_ops = { + .safeload = adau17x1_safeload, +}; + int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name) @@ -1002,8 +1062,13 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, dev_set_drvdata(dev, adau); if (firmware_name) { - adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL, - firmware_name); + if (adau17x1_has_safeload(adau)) { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + &adau17x1_sigmadsp_ops, firmware_name); + } else { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + NULL, firmware_name); + } if (IS_ERR(adau->sigmadsp)) { dev_warn(dev, "Could not find firmware file: %ld\n", PTR_ERR(adau->sigmadsp)); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..8efa2df81c49 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -71,6 +71,7 @@ extern const struct snd_soc_dai_ops adau17x1_dai_ops; int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate); bool adau17x1_has_dsp(struct adau *adau); +bool adau17x1_has_safeload(struct adau *adau); #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 -- 2.11.0

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Re: [alsa-devel] : Kernal+debug installed: Help with TASCAM US-122
by Takashi Iwai 10 Aug '18

10 Aug '18

On Wed, 08 Aug 2018 18:46:13 +0200, Joseph Spencer wrote: > > I installed the kernal-debug package along with debug info (Linux > localhost.localdomain 4.17.11-200.fc28.x86_64+debug #1 SMP Mon Jul 30 > 14:53:06 UTC 2018 x86_64 x86_64 x86_64 GNU/Linux). > > I looked in /boot/config-4.17.11-200.fc28.x86_64+debug and I see that the > debug options are now enabled (attaching this file in case there are more > options I can enable). > > I repeated another failure. The output in journalctl -f appears to me to > be about the same (see attached) > > Any pointers on what I could enable or do to debug this further? If so, then try to disable i915 KMS. e.g. boot with nomodeset option, and check whether the crash still happens. Takashi

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[alsa-devel] alsa bug - Martel Electronics USB soundcard recording sampling rate not automatically detected
by Susan Cragin 09 Aug '18

09 Aug '18

I run Dragon NaturallySpeaking through wine. The program works great with my old Logitech USB soundcard. But I just bought a new state-of-the-art dragon-optimized USB soundcard and stenomask, and the correct sampling rate cannot be automatically selected. Under proc/asound the device is described as EarMicrophone /tmp/alsa-info.txt.RjFRV2vThr

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[alsa-devel] Fwd: ALSA queries
by Chakravarthi Pradeep 09 Aug '18

09 Aug '18

---------- Forwarded message --------- From: Chakravarthi Pradeep <doubleq7(a)gmail.com> Date: Sun 5 Aug, 2018, 14:19 Subject: Re: [alsa-devel] ALSA queries To: <tiwai(a)suse.de> Hello Takashi, Thanks for your reply. periods_min = the minimal number of periods period_bytes_min = the minimal size of bytes for one period period_bytes_max = the maximal size of bytes for one period periods_min = the minimal number of periods : what is meaning of periods , is it minimal number of interrupts ? period_bytes_min = the minimal size of bytes for one period : it means, minimal size of bytes per interrupt in case of device, Is it correct ? what about period_max ? It's most likely the ALSA PCM core's safety stop; your driver seem to have missed snd_pcm_period_elapsed() calls, so the hwptr isn't updated, resulting in XRUN. ALSA PCM core checks such XRUN condition with the own timer. I'm attaching my driver thread along with this mail. Can you please let me know if I have missed something in audio thread. ?. How to make sure in driver that, trigger stop should be called only when stop command is sent from application. I'm getting cut cut cut ... noise along with audio in VLC application. Initially for 2 or 4 seconds, cut cut cut noise is not heard in VLC application, However after almost after 5 sec ,I can hear cut cut cut noise in VLC application. With my analysis, hwptr is getting updated properly however I have doubt in app_ptr. I'm attaching the excl sheet with hw_ptr,app_ptr and buf_pos values. How to remove the cut cut cut ... noise in audio ? Regards, Chakravarthi On Fri, Aug 3, 2018 at 1:05 PM Takashi Iwai <tiwai(a)suse.de> wrote: > > On Thu, 02 Aug 2018 16:31:13 +0200, > Chakravarthi Pradeep wrote: > > > > I'm working on ALSA driver for PCIe card. My ALSA driver and it's > > initializing struct snd_pcm_hardware with below parameter. > > > > /************************ code start > > ************************************************/ > > static struct snd_pcm_hardware audio_pcm_hardware = { > > .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | > > SNDRV_PCM_INFO_MMAP_VALID | > > SNDRV_PCM_INFO_BLOCK_TRANSFER | > > SNDRV_PCM_INFO_RESUME ), > > .formats = (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S24_LE), > > .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | > > SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), > > .rate_min = 44100, > > .rate_max = 192000, > > .channels_min = 2, > > .channels_max = 8, > > .buffer_bytes_max = 76800, /*75 kbytes */ > > .period_bytes_min = 512,//4410, /* (channel (2) * sample_rate (44100) > > * bit width (24)) / (60 * 8) */ > > .period_bytes_max = 16*1024, > > .periods_min = 10, > > .periods_max = 255, > > > > }; > > /************************ code end > > ************************************************/ > > > > 1) I did not understand what is significance of periods_min , > > period_bytes_min , period_bytes_max and periods_max. Can you please > > tell me what is importance of these parameter and what value should be > > mentioned according into ALSA. > > These three defines the values your hardware may accept: > periods_min = the minimal number of periods > period_bytes_min = the minimal size of bytes for one period > period_bytes_max = the maximal size of bytes for one period > > > 2) snd_pcm_ops trigger callback is getting called in the driver when > > application sends "start" command. But ALSA driver is stopping by > > itself after one frame is copied to ALSA framework, without waiting > > for "stop" command. > > > > For instance: > > In trigger callback , I'm getting these logs after one frame is copied. > > Trigger:Start (When Play button is selected/clicked in application, > > Start command is sent to ALSA driver) > > Dma transfer is completed. > > Trigger:Stop. (When Stop button is selected/clicked in application, > > Stop command is sent to ALSA driver. But stop button is not clicked in > > this case) > > It's most likely the ALSA PCM core's safety stop; your driver seem to > have missed snd_pcm_period_elapsed() calls, so the hwptr isn't > updated, resulting in XRUN. ALSA PCM core checks such XRUN condition > with the own timer. > > > Takashi -- Thanks and Regards Chakravarthi Pradeep.K Ph: 91 9980434900

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[alsa-devel] [PATCH] ASoC: adav80x: mark expected switch fall-through
by Gustavo A. R. Silva 09 Aug '18

09 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1056531 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/adav80x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index db21ecb..8b9ca7e 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -648,6 +648,7 @@ static int adav80x_set_pll(struct snd_soc_component *component, int pll_id, pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; break; } + /* fall through */ default: return -EINVAL; } -- 2.7.4

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[alsa-devel] [PATCH] ASoC: adau1761: Mark expected switch fall-though
by Gustavo A. R. Silva 09 Aug '18

09 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 114879 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/adau1761.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index be136e9..bef3e9e 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component) ARRAY_SIZE(adau1761_jack_detect_controls)); if (ret) return ret; - case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */ + /* fall through */ + case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes, ARRAY_SIZE(adau1761_no_dmic_routes)); if (ret) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: da7219: Add delays to capture path to remove DC offset noise
by Adam Thomson 09 Aug '18

09 Aug '18

On some platforms it has been noted that a pop noise can be witnessed when capturing audio, mainly for first time after a headset jack has been inserted. This is due to a DC offset in the Mic PGA and so to avoid this delays are required when powering up the capture path. This commit rectifies the problem by adding delays post Mic PGA and post Mixin PGA. The post Mic PGA delay is determined based on Mic Bias voltage, and is only applied the first time after a headset jack is inserted. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7219-aad.c | 5 +++++ sound/soc/codecs/da7219.c | 44 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/da7219.h | 8 ++++++-- 3 files changed, 49 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index a49ab75..2c7d508 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -59,6 +59,7 @@ static void da7219_aad_btn_det_work(struct work_struct *work) container_of(work, struct da7219_aad_priv, btn_det_work); struct snd_soc_component *component = da7219_aad->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 statusa, micbias_ctrl; bool micbias_up = false; int retries = 0; @@ -86,6 +87,8 @@ static void da7219_aad_btn_det_work(struct work_struct *work) if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES) dev_warn(component->dev, "Mic bias status check timed out"); + da7219->micbias_on_event = true; + /* * Mic bias pulse required to enable mic, must be done before enabling * button detection to prevent erroneous button readings. @@ -439,6 +442,8 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_1, DA7219_BUTTON_CONFIG_MASK, 0); + da7219->micbias_on_event = false; + /* Disable mic bias */ snd_soc_dapm_disable_pin(dapm, "Mic Bias"); snd_soc_dapm_sync(dapm); diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index c0144f2..e46e9f4 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -768,6 +768,30 @@ static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, * DAPM Events */ +static int da7219_mic_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (da7219->micbias_on_event) { + /* + * Delay only for first capture after bias enabled to + * avoid possible DC offset related noise. + */ + da7219->micbias_on_event = false; + msleep(da7219->mic_pga_delay); + } + break; + default: + break; + } + + return 0; +} + static int da7219_dai_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -937,12 +961,12 @@ static int da7219_gain_ramp_event(struct snd_soc_dapm_widget *w, SND_SOC_DAPM_INPUT("MIC"), /* Input PGAs */ - SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL, - DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), - SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL, - DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), + SND_SOC_DAPM_PGA_E("Mic PGA", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_mic_pga_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("Mixin PGA", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_settling_event, SND_SOC_DAPM_POST_PMU), /* Input Filters */ SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT, @@ -1847,6 +1871,14 @@ static void da7219_handle_pdata(struct snd_soc_component *component) snd_soc_component_write(component, DA7219_MICBIAS_CTRL, micbias_lvl); + /* + * Calculate delay required to compensate for DC offset in + * Mic PGA, based on Mic Bias voltage. + */ + da7219->mic_pga_delay = DA7219_MIC_PGA_BASE_DELAY + + (pdata->micbias_lvl * + DA7219_MIC_PGA_OFFSET_DELAY); + /* Mic */ switch (pdata->mic_amp_in_sel) { case DA7219_MIC_AMP_IN_SEL_DIFF: diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 1b00023..3a00686 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -781,8 +781,10 @@ #define DA7219_SYS_STAT_CHECK_DELAY 50 /* Power up/down Delays */ -#define DA7219_SETTLING_DELAY 40 -#define DA7219_MIN_GAIN_DELAY 30 +#define DA7219_SETTLING_DELAY 40 +#define DA7219_MIN_GAIN_DELAY 30 +#define DA7219_MIC_PGA_BASE_DELAY 100 +#define DA7219_MIC_PGA_OFFSET_DELAY 40 enum da7219_clk_src { DA7219_CLKSRC_MCLK = 0, @@ -828,6 +830,8 @@ struct da7219_priv { bool master; bool alc_en; + bool micbias_on_event; + unsigned int mic_pga_delay; u8 gain_ramp_ctrl; }; -- 1.9.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Mark expected switch fall-through
by Gustavo A. R. Silva 09 Aug '18

09 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1357413 ("Missing break in switch") Addresses-Coverity-ID: 114917 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/usb/mixer.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 73e811f..c63c84b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -856,12 +856,14 @@ static int check_input_term(struct mixer_build *state, int id, term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ else /* UAC_VERSION_2 */ term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ + /* fall through */ case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 */ if (protocol == UAC_VERSION_1 && !term->type) term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */ else if (protocol == UAC_VERSION_2 && !term->type) term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ + /* fall through */ case UAC2_EXTENSION_UNIT_V2: { struct uac_processing_unit_descriptor *d = p1; -- 2.7.4

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