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

[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

1 0

[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

1 0

[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

1 0

[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

2 1

[alsa-devel] [PATCH -next] ASoC: mt6351: remove unused variable 'mt_lineout_control'
by YueHaibing 09 Aug '19

09 Aug '19

sound/soc/codecs/mt6351.c:1070:38: warning: mt_lineout_control defined but not used [-Wunused-const-variable=] 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/mt6351.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/mt6351.c b/sound/soc/codecs/mt6351.c index 4b3ce01..5c0536e 100644 --- a/sound/soc/codecs/mt6351.c +++ b/sound/soc/codecs/mt6351.c @@ -1066,11 +1066,6 @@ static int mt_mic_bias_2_event(struct snd_soc_dapm_widget *w, return 0; } -/* DAPM Kcontrols */ -static const struct snd_kcontrol_new mt_lineout_control = - SOC_DAPM_SINGLE("Switch", MT6351_AUDDEC_ANA_CON3, - RG_AUDLOLPWRUP_VAUDP32_BIT, 1, 0); - /* DAPM Widgets */ static const struct snd_soc_dapm_widget mt6351_dapm_widgets[] = { /* Digital Clock */ -- 2.7.4

2 1

[alsa-devel] [PATCH 0/6] ASoC: SOF: refactor duplicated code between Intel and i.MX8
by Pierre-Louis Bossart 09 Aug '19

09 Aug '19

During the initial i.MX patch reviews on GitHub, we realized a lot of the i.MX mailbox/window code was common with the Intel loader code except for a couple of offsets. Most of the credit goes to Daniel Baluta for working on this series which adds a better abstraction and reduces code duplication. Comments from Keyon and Ranjani helped remove unnecessary parts kept for historical reasons and improve the code overall. Daniel Baluta (6): ASoC: SOF: Remove call to snd_sof_dsp_mailbox_init ASoC: SOF: Introduce snd_sof_dsp_get_mailbox_offset ASoC: SOF: Introduce snd_sof_dsp_get_window_offset ASoC: SOF: Intel: byt: Refactor fw ready / mem windows creation ASoC: SOF: Intel: bdw: Use generic function for fw ready / mem windows creation ASoC: SOF: Intel: hda: Use generic function for fw ready / mem windows creation sound/soc/sof/intel/apl.c | 4 +- sound/soc/sof/intel/bdw.c | 161 +++----------------------------- sound/soc/sof/intel/byt.c | 169 ++++------------------------------ sound/soc/sof/intel/cnl.c | 4 +- sound/soc/sof/intel/hda-ipc.c | 150 +----------------------------- sound/soc/sof/intel/hda.h | 4 +- sound/soc/sof/loader.c | 168 +++++++++++++++++++++++++++++++++ sound/soc/sof/ops.h | 18 ++++ sound/soc/sof/sof-priv.h | 6 ++ 9 files changed, 237 insertions(+), 447 deletions(-) -- 2.20.1

3 14

[alsa-devel] [PATCH 0/3] ALSA: hda: bus cleanup
by Takashi Iwai 09 Aug '19

09 Aug '19

Hi, this is a few patches to simplify and cleanup the HD-audio bus ops. The first two patches translate the indirect calls of DMA page allocation and MMIO accesses with the direct ones, as well as eliminating the whole bus->io_ops. The last one is SOF-specific, and fixes/cleans up by calling the proper hdaudio bus init function, as formerly discussed. Takashi === Takashi Iwai (3): ALSA: hda: Remove page allocation redirection ALSA: hda: Direct MMIO accesses ASoC: SOF: Intel: Initialize hdaudio bus properly include/sound/hdaudio.h | 69 +++++++++++++-------------- include/sound/hdaudio_ext.h | 1 - sound/hda/Kconfig | 3 ++ sound/hda/ext/hdac_ext_bus.c | 60 +----------------------- sound/hda/hdac_bus.c | 36 ++++++++++++-- sound/hda/hdac_controller.c | 18 +++---- sound/hda/hdac_stream.c | 8 ++-- sound/pci/hda/Kconfig | 1 + sound/pci/hda/hda_controller.c | 6 +-- sound/pci/hda/hda_controller.h | 3 +- sound/pci/hda/hda_intel.c | 71 ++-------------------------- sound/pci/hda/hda_tegra.c | 84 +-------------------------------- sound/soc/intel/skylake/skl-messages.c | 15 +----- sound/soc/intel/skylake/skl.c | 7 ++- sound/soc/sof/intel/hda-bus.c | 85 ++++------------------------------ sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda.c | 6 +-- sound/soc/sof/intel/hda.h | 3 +- 18 files changed, 107 insertions(+), 371 deletions(-) -- 2.16.4

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[alsa-devel] [PATCH -next] ALSA: Au88x0 - remove some unused const variables
by YueHaibing 09 Aug '19

09 Aug '19

sound/pci/au88x0/au88x0_xtalk.c:121:28: warning: asXtalkWideCoefsRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:152:28: warning: asXtalkNarrowCoefsRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:175:28: warning: asXtalkCoefsNegPipe defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:183:28: warning: asXtalkCoefsNumTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:191:28: warning: asXtalkCoefsDenTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:199:28: warning: asXtalkOutStateTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:20:20: warning: sXtalkWideKRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:231:28: warning: asDiamondCoefsRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:24:20: warning: sXtalkWideShiftRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:30:20: warning: sXtalkNarrowKRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:34:20: warning: sXtalkNarrowShiftRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:38:28: warning: asXtalkGainsDefault defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:43:28: warning: asXtalkGainsTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:48:28: warning: asXtalkGains1Chan defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:67:28: warning: alXtalkDlineTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:77:30: warning: asXtalkInStateTest defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:92:20: warning: sDiamondKRightXt defined but not used [-Wunused-const-variable=] sound/pci/au88x0/au88x0_xtalk.c:96:20: warning: sDiamondShiftRightXt defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/pci/au88x0/au88x0_xtalk.c | 85 ----------------------------------------- 1 file changed, 85 deletions(-) diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c index 084fcbf..143646c 100644 --- a/sound/pci/au88x0/au88x0_xtalk.c +++ b/sound/pci/au88x0/au88x0_xtalk.c @@ -17,39 +17,20 @@ static short const sXtalkWideKLeftEq = 0x269C; static short const sXtalkWideKRightEq = 0x269C; static short const sXtalkWideKLeftXt = 0xF25E; -static short const sXtalkWideKRightXt = 0xF25E; static short const sXtalkWideShiftLeftEq = 1; static short const sXtalkWideShiftRightEq = 1; static short const sXtalkWideShiftLeftXt = 0; -static short const sXtalkWideShiftRightXt = 0; static unsigned short const wXtalkWideLeftDelay = 0xd; static unsigned short const wXtalkWideRightDelay = 0xd; static short const sXtalkNarrowKLeftEq = 0x468D; static short const sXtalkNarrowKRightEq = 0x468D; static short const sXtalkNarrowKLeftXt = 0xF82E; -static short const sXtalkNarrowKRightXt = 0xF82E; static short const sXtalkNarrowShiftLeftEq = 0x3; static short const sXtalkNarrowShiftRightEq = 0x3; static short const sXtalkNarrowShiftLeftXt = 0; -static short const sXtalkNarrowShiftRightXt = 0; static unsigned short const wXtalkNarrowLeftDelay = 0x7; static unsigned short const wXtalkNarrowRightDelay = 0x7; -static xtalk_gains_t const asXtalkGainsDefault = { - 0x4000, 0x4000, 0x4000, 0x4000, 0x4000, - 0x4000, 0x4000, 0x4000, 0x4000, 0x4000 -}; - -static xtalk_gains_t const asXtalkGainsTest = { - 0x7fff, 0x8000, 0x0000, 0x0000, 0x0001, - 0xffff, 0x4000, 0xc000, 0x0002, 0xfffe -}; - -static xtalk_gains_t const asXtalkGains1Chan = { - 0x7FFF, 0, 0, 0, 0, - 0x7FFF, 0, 0, 0, 0, -}; - // Input gain for 4 A3D slices. One possible input pair is left zero. static xtalk_gains_t const asXtalkGainsAllChan = { 0x7FFF, 0x7FFF, 0x7FFF, 0x7FFF, 0, @@ -64,20 +45,11 @@ static xtalk_dline_t const alXtalkDlineZeros = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }; -static xtalk_dline_t const alXtalkDlineTest = { - 0x0000fc18, 0xfff03e8, 0x000186a0, 0xfffe7960, 1, 0xffffffff, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0 -}; static xtalk_instate_t const asXtalkInStateZeros = { 0, 0, 0, 0 }; -static xtalk_instate_t const asXtalkInStateTest = { - 0x0080, 0xff80, 0x0001, 0xffff -}; - static xtalk_state_t const asXtalkOutStateZeros = { {0, 0, 0, 0}, {0, 0, 0, 0}, @@ -89,11 +61,9 @@ static xtalk_state_t const asXtalkOutStateZeros = { static short const sDiamondKLeftEq = 0x401d; static short const sDiamondKRightEq = 0x401d; static short const sDiamondKLeftXt = 0xF90E; -static short const sDiamondKRightXt = 0xF90E; static short const sDiamondShiftLeftEq = 1; static short const sDiamondShiftRightEq = 1; static short const sDiamondShiftLeftXt = 0; -static short const sDiamondShiftRightXt = 0; static unsigned short const wDiamondLeftDelay = 0xb; static unsigned short const wDiamondRightDelay = 0xb; @@ -118,13 +88,6 @@ static xtalk_coefs_t const asXtalkWideCoefsLeftXt = { {0x77dc, 0xc79e, 0xffb8, 0x000a, 0}, {0, 0, 0, 0, 0} }; -static xtalk_coefs_t const asXtalkWideCoefsRightXt = { - {0x55c6, 0xc97b, 0x005b, 0x0047, 0}, - {0x6a60, 0xca20, 0xffc6, 0x0040, 0}, - {0x6411, 0xd711, 0xfca1, 0x0190, 0}, - {0x77dc, 0xc79e, 0xffb8, 0x000a, 0}, - {0, 0, 0, 0, 0} -}; static xtalk_coefs_t const asXtalkNarrowCoefsLeftEq = { {0x50B5, 0xD07C, 0x026D, 0xFD21, 0}, {0x460F, 0xE44F, 0xF75E, 0xEFA6, 0}, @@ -149,14 +112,6 @@ static xtalk_coefs_t const asXtalkNarrowCoefsLeftXt = { {0, 0, 0, 0, 0} }; -static xtalk_coefs_t const asXtalkNarrowCoefsRightXt = { - {0x3CB2, 0xDF49, 0xF6EA, 0x095B, 0}, - {0x6777, 0xC915, 0xFEAF, 0x00B1, 0}, - {0x7762, 0xC7D9, 0x025B, 0xFDA6, 0}, - {0x6B7A, 0xD2AA, 0xF2FB, 0x0B64, 0}, - {0, 0, 0, 0, 0} -}; - static xtalk_coefs_t const asXtalkCoefsZeros = { {0, 0, 0, 0, 0}, {0, 0, 0, 0, 0}, @@ -172,37 +127,6 @@ static xtalk_coefs_t const asXtalkCoefsPipe = { {0, 0, 0x0FA0, 0, 0}, {0, 0, 0x1180, 0, 0}, }; -static xtalk_coefs_t const asXtalkCoefsNegPipe = { - {0, 0, 0xF380, 0, 0}, - {0, 0, 0xF380, 0, 0}, - {0, 0, 0xF380, 0, 0}, - {0, 0, 0xF380, 0, 0}, - {0, 0, 0xF200, 0, 0} -}; - -static xtalk_coefs_t const asXtalkCoefsNumTest = { - {0, 0, 0xF380, 0x8000, 0x6D60}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0} -}; - -static xtalk_coefs_t const asXtalkCoefsDenTest = { - {0xC000, 0x2000, 0x4000, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0} -}; - -static xtalk_state_t const asXtalkOutStateTest = { - {0x7FFF, 0x0004, 0xFFFC, 0}, - {0xFE00, 0x0008, 0xFFF8, 0x4000}, - {0x0200, 0x0010, 0xFFF0, 0xC000}, - {0x8000, 0x0020, 0xFFE0, 0}, - {0, 0, 0, 0} -}; static xtalk_coefs_t const asDiamondCoefsLeftEq = { {0x0F1E, 0x2D05, 0xF8E3, 0x07C8, 0}, @@ -228,15 +152,6 @@ static xtalk_coefs_t const asDiamondCoefsLeftXt = { {0, 0, 0, 0, 0} }; -static xtalk_coefs_t const asDiamondCoefsRightXt = { - {0x3B50, 0xFE08, 0xF959, 0x0060, 0}, - {0x9FCB, 0xD8F1, 0x00A2, 0x003A, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0} -}; - - /**/ /* XTalk EQ and XT */ static void vortex_XtalkHw_SetLeftEQ(vortex_t * vortex, short arg_0, short arg_4, -- 2.7.4

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[alsa-devel] [PATCH -next] ALSA: ac97: remove unused variable 'snd_ac97_controls_master_mono'
by YueHaibing 09 Aug '19

09 Aug '19

sound/pci/ac97/ac97_codec.c:599:38: warning: snd_ac97_controls_master_mono defined but not used [-Wunused-const-variable=] 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/pci/ac97/ac97_codec.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 96b4601..66f6c3b 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -596,11 +596,6 @@ static int snd_ac97_put_volsw(struct snd_kcontrol *kcontrol, return err; } -static const struct snd_kcontrol_new snd_ac97_controls_master_mono[2] = { -AC97_SINGLE("Master Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), -AC97_SINGLE("Master Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1) -}; - static const struct snd_kcontrol_new snd_ac97_controls_tone[2] = { AC97_SINGLE("Tone Control - Bass", AC97_MASTER_TONE, 8, 15, 1), AC97_SINGLE("Tone Control - Treble", AC97_MASTER_TONE, 0, 15, 1) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda - Apply workaround for another AMD chip 1022:1487
by Takashi Iwai 09 Aug '19

09 Aug '19

MSI MPG X570 board is with another AMD HD-audio controller (PCI ID 1022:1487) and it requires the same workaround applied for X370, etc (PCI ID 1022:1457). BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a6d8c0d77b84..99fc0917339b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2508,6 +2508,9 @@ static const struct pci_device_id azx_ids[] = { /* AMD, X370 & co */ { PCI_DEVICE(0x1022, 0x1457), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, + /* AMD, X570 & co */ + { PCI_DEVICE(0x1022, 0x1487), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* AMD Stoney */ { PCI_DEVICE(0x1022, 0x157a), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | -- 2.16.4

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