August 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] snd_pcm_start question
by Eric Desjardins 11 Aug '19

11 Aug '19

Hi, This is a simple question. I call: snd_pcm_prepare snd_pcm_writei and it works fine but I would like to delay playback start. There is snd_pcm_start for that. There is a method that is deprecated to specify a start mode but I was wondering what was the correct new way to do that? Thanks, Eric

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[alsa-devel] bcm2835.conf for RPi v. 4 incl. HDMI
by Pavel Hofman 10 Aug '19

10 Aug '19

Hi, may I ask for help with cards/bcm2835.conf for the new RPi? No such config exists yet: pi@raspberrypi:~ $ strace -f aplay -L 2>&1 | grep -i bcm2835 access("/usr/share/alsa/cards/bcm2835_alsa.conf", R_OK) = -1 ENOENT (No such file or directory) I understand the _alsa suffix is handled in aliases (e.g. https://www.raspberrypi.org/forums/viewtopic.php?t=221734#p1361104 ) The reason I am looking for this file are Pulseaudio profiles. PA requires front:X, surroundX, hdmi:X, etc. PCM devices, yet non of these are created due to the missing card config: pi@raspberrypi:~ $ aplay -L default Playback/recording through the PulseAudio sound server null Discard all samples (playback) or generate zero samples (capture) jack JACK Audio Connection Kit pulse PulseAudio Sound Server sysdefault:CARD=ALSA bcm2835 ALSA, bcm2835 ALSA Default Audio Device dmix:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Direct sample mixing device dmix:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Direct sample mixing device dsnoop:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Direct sample snooping device dsnoop:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Direct sample snooping device hw:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Direct hardware device without any conversions hw:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Direct hardware device without any conversions plughw:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Hardware device with all software conversions plughw:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Hardware device with all software conversions usbstream:CARD=ALSA bcm2835 ALSA USB Stream Output As a result, only the Analog Mono PA profile (based on the only existing hw:0) is available in PA: D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Mono (analog-mono) D: [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... D: [pulseaudio] alsa-util.c: Managed to open hw:0 ... D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Stereo (analog-stereo) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)conf.c: Unknown parameters 0 I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM front:0 --- D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo) D: [pulseaudio] alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)conf.c: Unknown parameters 0 I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM hdmi:0 ... D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 7.1 (HDMI) (hdmi-surround71) D: [pulseaudio] alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)conf.c: Unknown parameters 0 I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM hdmi:0 RPi v.4 has 2 HDMI outputs: pi@raspberrypi:~ $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ALSA [bcm2835 ALSA], device 0: bcm2835 ALSA [bcm2835 ALSA] Subdevices: 6/7 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 card 0: ALSA [bcm2835 ALSA], device 1: bcm2835 IEC958/HDMI [bcm2835 IEC958/HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: ALSA [bcm2835 ALSA], device 2: bcm2835 IEC958/HDMI1 [bcm2835 IEC958/HDMI1] Subdevices: 1/1 Subdevice #0: subdevice #0 Each is 2 - 8 channel: pi@raspberrypi:~ $ aplay -v --dump-hw-params -D hw:0,1 /dev/zero Playing raw data '/dev/zero' : Unsigned 8 bit, Rate 8000 Hz, Mono HW Params of device "hw:0,1": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: [32 128] CHANNELS: [2 8] RATE: [44100 192000] PERIOD_TIME: [10000 743039) PERIOD_SIZE: [441 32768] PERIOD_BYTES: [1776 524288] PERIODS: [1 75) BUFFER_TIME: (2296 743039) BUFFER_SIZE: [441 32768] BUFFER_BYTES: [1764 131072] TICK_TIME: ALL I looked at hdmi section of cards/Intel-HDA.conf but that one is very complicated. May I ask for help how the config should look like to work both for multichannel PCM and digital formats (i.e. support for the AES vars)? I very much appreciate any help. Best regards, Pavel.

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[alsa-devel] Unable to get sound out of Laptop
by Nilesh Kumar 10 Aug '19

10 Aug '19

Hello all, I am unable to get sound out of my "HP Notebook - 15-bs180tx" laptop, the output of alsa-info,sh is been uploaded at http://www.alsa-project.org/db/?f=5b015c625cef0cc14b0c0551cb57bc1e35d4b40a so, I would really appreciate if someone could see the log and help me out in configuring my sound card -- Nilesh Kumar

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[alsa-devel] Writing /usr/share/alsa/cards/ files
by Paul Pawlowski 10 Aug '19

10 Aug '19

Hello, Is there any documentation on writing /usr/share/alsa/cards/ files? I painfully tried to create one in order to enable 2 separate audio outputs, but I ultimately failed to due to the lack of any sort of error reporting. Is there a way to enable any error reporting when parsing these files? It seems that the error output when parsing any hooks goes to nowhere. Also, is this the correct mailing list to get help regarding alsa driver development and configuration? Thank you, Paul Pawlowski

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[alsa-devel] [PATCH] ALSA: hda - Fix a memory leak bug
by Wenwen Wang 10 Aug '19

10 Aug '19

In snd_hda_parse_generic_codec(), 'spec' is allocated through kzalloc(). Then, the pin widgets in 'codec' are parsed. However, if the parsing process fails, 'spec' is not deallocated, leading to a memory leak. To fix the above issue, free 'spec' before returning the error. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 485edab..8f2beb1 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -6100,7 +6100,7 @@ static int snd_hda_parse_generic_codec(struct hda_codec *codec) err = snd_hda_parse_pin_defcfg(codec, &spec->autocfg, NULL, 0); if (err < 0) - return err; + goto error; err = snd_hda_gen_parse_auto_config(codec, &spec->autocfg); if (err < 0) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: SOF: Intel: Add missing include file hdac_hda.h
by YueHaibing 09 Aug '19

09 Aug '19

Building with SND_SOC_SOF_HDA_AUDIO_CODEC fails: sound/soc/sof/intel/hda-bus.c: In function sof_hda_bus_init: sound/soc/sof/intel/hda-bus.c:16:25: error: implicit declaration of function snd_soc_hdac_hda_get_ops; did you mean snd_soc_jack_add_gpiods? [-Werror=implicit-function-declaration] #define sof_hda_ext_ops snd_soc_hdac_hda_get_ops() Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: d4ff1b3917a5 ('ASoC: SOF: Intel: Initialize hdaudio bus properly") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/intel/hda-bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c index 0caec3a..e28a9c4 100644 --- a/sound/soc/sof/intel/hda-bus.c +++ b/sound/soc/sof/intel/hda-bus.c @@ -13,6 +13,7 @@ #include "hda.h" #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) +#include "../../../soc/codecs/hdac_hda.h" #define sof_hda_ext_ops snd_soc_hdac_hda_get_ops() #else #define sof_hda_ext_ops NULL -- 2.7.4

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[alsa-devel] [PATCH] ASoC: hdac_hdmi: Offload dapm update at jack detection
by Takashi Iwai 09 Aug '19

09 Aug '19

hdac_hdmi_present_sense() calls the audio component to get ELD update, then it reports the jack status change and updates DAPM graph accordingly. This works when it's called from the normal code paths. However, it may lead to a dead lock when it's called from the audio component notifier. Namely, the DAPM update involves with the runtime PM, and it eventually calls again the audio component get_power() ops. Since i915 driver already takes a mutex around the audio component ops calls, we'll eventually get the mutex doubly. As a workaround, in this patch, only the jack state is updated in the code path from hdac_hdmi_eld_notify_cb(), and the DAPM update is deferred to a work so that it's processed in another context. Reported-by: Imre Deak <imre.deak(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a bug fix I did in the last year when we received some bug report, but forgotten in my archive. I believe the fix is still valid although the bug may trigger rarely. sound/soc/codecs/hdac_hdmi.c | 57 +++++++++++++++++++++++++++++++++++++------- 1 file changed, 48 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 29918954e740..18c173e6a13b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -88,8 +88,10 @@ struct hdac_hdmi_port { hda_nid_t mux_nids[HDA_MAX_CONNECTIONS]; struct hdac_hdmi_eld eld; const char *jack_pin; + bool is_connect; struct snd_soc_dapm_context *dapm; const char *output_pin; + struct work_struct dapm_work; }; struct hdac_hdmi_pcm { @@ -163,11 +165,7 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, { struct hdac_device *hdev = port->pin->hdev; - if (is_connect) - snd_soc_dapm_enable_pin(port->dapm, port->jack_pin); - else - snd_soc_dapm_disable_pin(port->dapm, port->jack_pin); - + port->is_connect = is_connect; if (is_connect) { /* * Report Jack connect event when a device is connected @@ -193,10 +191,32 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, if (pcm->jack_event > 0) pcm->jack_event--; } +} +static void hdac_hdmi_port_dapm_update(struct hdac_hdmi_port *port) +{ + if (port->is_connect) + snd_soc_dapm_enable_pin(port->dapm, port->jack_pin); + else + snd_soc_dapm_disable_pin(port->dapm, port->jack_pin); snd_soc_dapm_sync(port->dapm); } +static void hdac_hdmi_jack_dapm_work(struct work_struct *work) +{ + struct hdac_hdmi_port *port; + + port = container_of(work, struct hdac_hdmi_port, dapm_work); + hdac_hdmi_port_dapm_update(port); +} + +static void hdac_hdmi_jack_report_sync(struct hdac_hdmi_pcm *pcm, + struct hdac_hdmi_port *port, bool is_connect) +{ + hdac_hdmi_jack_report(pcm, port, is_connect); + hdac_hdmi_port_dapm_update(port); +} + /* MST supported verbs */ /* * Get the no devices that can be connected to a port on the Pin widget. @@ -904,7 +924,7 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, list_for_each_entry_safe(p, p_next, &pcm->port_list, head) { if (p == port && p->id == port->id && p->pin == port->pin) { - hdac_hdmi_jack_report(pcm, port, false); + hdac_hdmi_jack_report_sync(pcm, port, false); list_del(&p->head); } } @@ -918,7 +938,7 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, if (!strcmp(cvt_name, pcm->cvt->name)) { list_add_tail(&port->head, &pcm->port_list); if (port->eld.monitor_present && port->eld.eld_valid) { - hdac_hdmi_jack_report(pcm, port, true); + hdac_hdmi_jack_report_sync(pcm, port, true); mutex_unlock(&hdmi->pin_mutex); return ret; } @@ -1281,16 +1301,20 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, * report jack here. It will be done in usermode mux * control select. */ - if (pcm) + if (pcm) { hdac_hdmi_jack_report(pcm, port, false); + schedule_work(&port->dapm_work); + } mutex_unlock(&hdmi->pin_mutex); return; } if (port->eld.monitor_present && port->eld.eld_valid) { - if (pcm) + if (pcm) { hdac_hdmi_jack_report(pcm, port, true); + schedule_work(&port->dapm_work); + } print_hex_dump_debug("ELD: ", DUMP_PREFIX_OFFSET, 16, 1, port->eld.eld_buffer, port->eld.eld_size, false); @@ -1319,6 +1343,7 @@ static int hdac_hdmi_add_ports(struct hdac_device *hdev, for (i = 0; i < max_ports; i++) { ports[i].id = i; ports[i].pin = pin; + INIT_WORK(&ports[i].dapm_work, hdac_hdmi_jack_dapm_work); } pin->ports = ports; pin->num_ports = max_ports; @@ -2083,8 +2108,20 @@ static int hdac_hdmi_dev_probe(struct hdac_device *hdev) return ret; } +static void clear_dapm_works(struct hdac_device *hdev) +{ + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + struct hdac_hdmi_pin *pin; + int i; + + list_for_each_entry(pin, &hdmi->pin_list, head) + for (i = 0; i < pin->num_ports; i++) + cancel_work_sync(&pin->ports[i].dapm_work); +} + static int hdac_hdmi_dev_remove(struct hdac_device *hdev) { + clear_dapm_works(hdev); snd_hdac_display_power(hdev->bus, hdev->addr, false); return 0; @@ -2103,6 +2140,8 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) if (!bus) return 0; + clear_dapm_works(hdev); + /* * Power down afg. * codec_read is preferred over codec_write to set the power state. -- 2.16.4

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[alsa-devel] [PATCH] ALSA: hda - Add a quirk model for fixing Huawei Matebook X right speaker
by Takashi Iwai 09 Aug '19

09 Aug '19

From: Tomas Espeleta <tomas.espeleta(a)gmail.com> [ This is rather a revival of the patch Tomas sent in months ago, but applying only with the quirk model option -- tiwai ] Hard coded coefficients to make Huawuei Matebook X right speaker work. The Matebook X has a ALC298, please refer to bug 197801 on how these numbers were reverse engineered from the Windows driver The reversed engineered sequence represents a repeating pattern of verbs, and the only values that are changing periodically are written on indexes 0x23 and 0x25: 0x500, 0x23 0x400, VALUE1 0x500, 0x25 0x400, VALUE2 * skipped reading sequences (0x500 - 0xc00 sequences are ignored) * static values from reverse engineering are used NOTE: since a significant risk is still considered, this is provided as an experimental fix that isn't applied as default for now. For enabling the fix, you'll have to choose huawei-mbx-stereo via model option of snd-hda-intel module. If we get feedback from users that this works stably, we may apply it per default. [ Some coding style fixes and replacement with AC_VERB_* by tiwai ] BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=197801 Signed-off-by: Tomas Espeleta <tomas.espeleta(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/hd-audio/models.rst | 3 ++ sound/pci/hda/patch_realtek.c | 74 +++++++++++++++++++++++++++++++++ 2 files changed, 77 insertions(+) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 7d7c191102a7..11298f0ce44d 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -260,6 +260,9 @@ alc295-hp-x360 HP Spectre X360 fixups alc-sense-combo Headset button support for Chrome platform +huawei-mbx-stereo + Enable initialization verbs for Huawei MBX stereo speakers; + might be risky, try this at your own risk ALC66x/67x/892 ============== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index de224cbea7a0..cb7baa65b298 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3762,6 +3762,72 @@ static void alc269_x101_hp_automute_hook(struct hda_codec *codec, vref); } +/* + * Magic sequence to make Huawei Matebook X right speaker working (bko#197801) + */ +struct hda_alc298_mbxinit { + unsigned char value_0x23; + unsigned char value_0x25; +}; + +static void alc298_huawei_mbx_stereo_seq(struct hda_codec *codec, + const struct hda_alc298_mbxinit *initval, + bool first) +{ + snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x0); + alc_write_coef_idx(codec, 0x26, 0xb000); + + if (first) + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_GET_PIN_SENSE, 0x0); + + snd_hda_codec_write(codec, 0x6, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x80); + alc_write_coef_idx(codec, 0x26, 0xf000); + alc_write_coef_idx(codec, 0x23, initval->value_0x23); + + if (initval->value_0x23 != 0x1e) + alc_write_coef_idx(codec, 0x25, initval->value_0x25); + + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0xb010); +} + +static void alc298_fixup_huawei_mbx_stereo(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* Initialization magic */ + static const struct hda_alc298_mbxinit dac_init[] = { + {0x0c, 0x00}, {0x0d, 0x00}, {0x0e, 0x00}, {0x0f, 0x00}, + {0x10, 0x00}, {0x1a, 0x40}, {0x1b, 0x82}, {0x1c, 0x00}, + {0x1d, 0x00}, {0x1e, 0x00}, {0x1f, 0x00}, + {0x20, 0xc2}, {0x21, 0xc8}, {0x22, 0x26}, {0x23, 0x24}, + {0x27, 0xff}, {0x28, 0xff}, {0x29, 0xff}, {0x2a, 0x8f}, + {0x2b, 0x02}, {0x2c, 0x48}, {0x2d, 0x34}, {0x2e, 0x00}, + {0x2f, 0x00}, + {0x30, 0x00}, {0x31, 0x00}, {0x32, 0x00}, {0x33, 0x00}, + {0x34, 0x00}, {0x35, 0x01}, {0x36, 0x93}, {0x37, 0x0c}, + {0x38, 0x00}, {0x39, 0x00}, {0x3a, 0xf8}, {0x38, 0x80}, + {} + }; + const struct hda_alc298_mbxinit *seq; + + if (action != HDA_FIXUP_ACT_INIT) + return; + + /* Start */ + snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x00); + snd_hda_codec_write(codec, 0x06, 0, AC_VERB_SET_DIGI_CONVERT_3, 0x80); + alc_write_coef_idx(codec, 0x26, 0xf000); + alc_write_coef_idx(codec, 0x22, 0x31); + alc_write_coef_idx(codec, 0x23, 0x0b); + alc_write_coef_idx(codec, 0x25, 0x00); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0xb010); + + for (seq = dac_init; seq->value_0x23; seq++) + alc298_huawei_mbx_stereo_seq(codec, seq, seq == dac_init); +} + static void alc269_fixup_x101_headset_mic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5787,6 +5853,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MACH_WX9_PINS, + ALC298_FIXUP_HUAWEI_MBX_STEREO, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, @@ -6095,6 +6162,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_MIC_MUTE_LED }, + [ALC298_FIXUP_HUAWEI_MBX_STEREO] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_huawei_mbx_stereo, + .chained = true, + .chain_id = ALC255_FIXUP_MIC_MUTE_LED + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -7272,6 +7345,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-headset-jack"}, {.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"}, {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, + {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.16.4

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[alsa-devel] [PATCH] ALSA: hda - Inform too slow responses
by Takashi Iwai 09 Aug '19

09 Aug '19

For the sake of debugging, show each response that took too long time and its corresponding last verb. Since such an error happens in series once when started, use the ratelimited variant for suppressing the flood. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 462559442f7a..6387c7e90918 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -794,6 +794,7 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, unsigned long timeout; unsigned long loopcounter; int do_poll = 0; + bool warned = false; again: timeout = jiffies + msecs_to_jiffies(1000); @@ -813,9 +814,17 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, spin_unlock_irq(&bus->reg_lock); if (time_after(jiffies, timeout)) break; - if (hbus->needs_damn_long_delay || loopcounter > 3000) +#define LOOP_COUNT_MAX 3000 + if (hbus->needs_damn_long_delay || + loopcounter > LOOP_COUNT_MAX) { + if (loopcounter > LOOP_COUNT_MAX && !warned) { + dev_dbg_ratelimited(chip->card->dev, + "too slow response, last cmd=%#08x\n", + bus->last_cmd[addr]); + warned = true; + } msleep(2); /* temporary workaround */ - else { + } else { udelay(10); cond_resched(); } -- 2.16.4

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[alsa-devel] [PATCH -next] ASoC: ml26124: remove unused variable 'ngth'
by YueHaibing 09 Aug '19

09 Aug '19

In file included from ./include/sound/tlv.h:10:0, from sound/soc/codecs/ml26124.c:19: sound/soc/codecs/ml26124.c:59:35: warning: ngth defined but not used [-Wunused-const-variable=] static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); ^ ./include/uapi/sound/tlv.h:64:15: note: in definition of macro SNDRV_CTL_TLVD_DECLARE_DB_SCALE unsigned int name[] = { \ ^~~~ sound/soc/codecs/ml26124.c:59:14: note: in expansion of macro DECLARE_TLV_DB_SCALE static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); ^~~~~~~~~~~~~~~~~~~~ It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/ml26124.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 3abd278..55823bc 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -56,7 +56,6 @@ static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); -static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); static const char * const ml26124_companding[] = {"16bit PCM", "u-law", "A-law"}; -- 2.7.4

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