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

[alsa-devel] [PATCH -next] ASoC: tlv320aic23: remove unused variable 'tlv320aic23_rec_src'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/tlv320aic23.c:70:29: warning: tlv320aic23_rec_src 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/tlv320aic23.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 080a840..f8e2f4b 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -67,8 +67,6 @@ static SOC_ENUM_SINGLE_DECL(rec_src_enum, static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = SOC_DAPM_ENUM("Input Select", rec_src_enum); -static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src, - TLV320AIC23_ANLG, 2, rec_src_text); static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph, TLV320AIC23_DIGT, 1, deemph_text); -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: rt1011: remove unused variable 'dac_vol_tlv' and 'adc_vol_tlv'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/rt1011.c:981:35: warning: dac_vol_tlv defined but not used [-Wunused-const-variable=] sound/soc/codecs/rt1011.c:982:35: warning: adc_vol_tlv defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/rt1011.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 638abca..fa34565 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -978,9 +978,6 @@ static bool rt1011_readable_register(struct device *dev, unsigned int reg) } } -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9435, 37, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1739, 37, 0); - static const char * const rt1011_din_source_select[] = { "Left", "Right", -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: 88pm860x: remove unused variables 'pcm_switch_controls' and 'aif1_mux'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/88pm860x-codec.c:533:38: warning: pcm_switch_controls defined but not used [-Wunused-const-variable=] sound/soc/codecs/88pm860x-codec.c:560:38: warning: aif1_mux 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/soc/codecs/88pm860x-codec.c | 15 --------------- 1 file changed, 15 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index e982722..00b2c43 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -529,10 +529,6 @@ static const struct snd_kcontrol_new pm860x_snd_controls[] = { * DAPM Controls */ -/* PCM Switch / PCM Interface */ -static const struct snd_kcontrol_new pcm_switch_controls = - SOC_DAPM_SINGLE("Switch", PM860X_ADC_EN_2, 0, 1, 0); - /* AUX1 Switch */ static const struct snd_kcontrol_new aux1_switch_controls = SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 4, 1, 0); @@ -549,17 +545,6 @@ static const struct snd_kcontrol_new lepa_switch_controls = static const struct snd_kcontrol_new repa_switch_controls = SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 1, 1, 0); -/* PCM Mux / Mux7 */ -static const char *aif1_text[] = { - "PCM L", "PCM R", -}; - -static SOC_ENUM_SINGLE_DECL(aif1_enum, - PM860X_PCM_IFACE_3, 6, aif1_text); - -static const struct snd_kcontrol_new aif1_mux = - SOC_DAPM_ENUM("PCM Mux", aif1_enum); - /* I2S Mux / Mux9 */ static const char *i2s_din_text[] = { "DIN", "DIN1", -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: cs42l56: remove unused variable 'adc_swap_enum'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/cs42l56.c:206:30: warning: adc_swap_enum 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/cs42l56.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index b4d7627..ac569ab 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -199,14 +199,6 @@ static const struct soc_enum beep_bass_enum = SOC_ENUM_SINGLE(CS42L56_BEEP_TONE_CFG, 1, ARRAY_SIZE(beep_bass_text), beep_bass_text); -static const char * const adc_swap_text[] = { - "None", "A+B/2", "A-B/2", "Swap" -}; - -static const struct soc_enum adc_swap_enum = - SOC_ENUM_SINGLE(CS42L56_MISC_ADC_CTL, 3, - ARRAY_SIZE(adc_swap_text), adc_swap_text); - static const char * const pgaa_mux_text[] = { "AIN1A", "AIN2A", "AIN3A"}; -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] ASoC: Intel/SOF: add initial support for Tiger Lake and Elhkart Lake
by Pierre-Louis Bossart 15 Aug '19

15 Aug '19

Add initial/minimal support for PCI enumeration, hardware descriptions and ACPI match tables. Follow-up patches will be provided to enable multi-core support (requires new IPC calls). Machine drivers will be provided at a later point as well when there are fewer configurations. And last HDMI work is required to handle the 4 converters (which may be combined with the transition from hdac_hdmi to patch_hdmi already mentioned by Kai). Pan Xiuli (4): ASoC: Intel: common: add ACPI matching tables for Tiger Lake ASoC: SOF: Intel: initial support for Tiger Lake. ASoC: Intel: common: add ACPI matching tables for EHL ASoC: SOF: Intel: initial support for Elkhart Lake include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ehl-match.c | 18 ++++++++ .../intel/common/soc-acpi-intel-tgl-match.c | 24 ++++++++++ sound/soc/sof/intel/Kconfig | 32 ++++++++++++++ sound/soc/sof/intel/cnl.c | 32 ++++++++++++++ sound/soc/sof/intel/hda.h | 2 + sound/soc/sof/sof-pci-dev.c | 44 +++++++++++++++++++ 8 files changed, 155 insertions(+) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ehl-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-tgl-match.c -- 2.20.1

2 10

[alsa-devel] [PATCH] [v2] ASoC: Intel: boards: Add Cometlake machine driver support
by mac.chiang＠intel.com 15 Aug '19

15 Aug '19

From: Mac Chiang <mac.chiang(a)intel.com> reuse and add Cometlake support with: SSP0 for DA7219 headphone codec SSP1 for MAX98357a speaker amp codec Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v2: - included GENERIC config on APL/KBL platforms as well - reuse the same pcm_id and dai_id, modify in tplg support - replace with soc_intel_is_cml() instead --- --- sound/soc/intel/boards/Kconfig | 27 ++++++++++++++------ sound/soc/intel/boards/bxt_da7219_max98357a.c | 31 ++++++++++++++++++++++- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 12 +++++++++ sound/soc/intel/common/soc-intel-quirks.h | 5 ++++ 4 files changed, 66 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 50bf149..f3d88c87 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -256,16 +256,20 @@ config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH endif ## SND_SOC_INTEL_SKL +config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC + tristate + select SND_SOC_DA7219 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + if SND_SOC_INTEL_APL config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_DA7219 - select SND_SOC_MAX98357A - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC select SND_HDA_DSP_LOADER help This adds support for ASoC machine driver for Broxton-P platforms @@ -326,10 +330,7 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_DA7219 - select SND_SOC_MAX98357A - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC help This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for DA7219 + MAX98357A I2S audio codec. @@ -412,4 +413,14 @@ config SND_SOC_INTEL_SOF_RT5682_MACH If unsure select "N". endif ## SND_SOC_SOF_HDA_COMMON || SND_SOC_SOF_BAYTRAIL +if (SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK) + +config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH + tristate "CML_LP with DA7219 and MAX98357A in I2S Mode" + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC + +endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK + endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index c0d865a..ac1dea5 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -179,10 +179,17 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) int ret; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_component *component = rtd->codec_dai->component; + int clk_freq; /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 19200000, + if (soc_intel_is_cml()) + clk_freq = 24000000; + else + clk_freq = 19200000; + + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, clk_freq, SND_SOC_CLOCK_IN); + if (ret) { dev_err(rtd->dev, "can't set codec sysclk configuration\n"); return ret; @@ -683,6 +690,25 @@ static int broxton_audio_probe(struct platform_device *pdev) broxton_dais[i].cpus->dai_name = "SSP2 Pin"; } } + } else if (soc_intel_is_cml()) { + unsigned int i; + + broxton_audio_card.name = "cmlda7219max"; + + for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { + /* MAXIM_CODEC is connected to SSP1. */ + if (!strcmp(broxton_dais[i].codecs->dai_name, + BXT_MAXIM_CODEC_DAI)) { + broxton_dais[i].name = "SSP1-Codec"; + broxton_dais[i].cpus->dai_name = "SSP1 Pin"; + } + /* DIALOG_CODEC is connected to SSP0 */ + else if (!strcmp(broxton_dais[i].codecs->dai_name, + BXT_DIALOG_CODEC_DAI)) { + broxton_dais[i].name = "SSP0-Codec"; + broxton_dais[i].cpus->dai_name = "SSP0 Pin"; + } + } } /* override plaform name, if required */ @@ -700,6 +726,7 @@ static int broxton_audio_probe(struct platform_device *pdev) static const struct platform_device_id bxt_board_ids[] = { { .name = "bxt_da7219_max98357a" }, { .name = "glk_da7219_max98357a" }, + { .name = "cml_da7219_max98357a" }, { } }; @@ -720,6 +747,8 @@ MODULE_AUTHOR("Rohit Ainapure <rohit.m.ainapure(a)intel.com>"); MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); MODULE_AUTHOR("Conrad Cooke <conrad.cooke(a)intel.com>"); MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:bxt_da7219_max98357a"); MODULE_ALIAS("platform:glk_da7219_max98357a"); +MODULE_ALIAS("platform:cml_da7219_max98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index c36c0aa..4ea32b2 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs cml_codecs = { .codecs = {"10EC5682"} }; +static struct snd_soc_acpi_codecs cml_spk_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { { .id = "INT34C2", @@ -29,6 +34,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .sof_tplg_filename = "sof-cnl-rt274.tplg", }, { + .id = "DLGS7219", + .drv_name = "cml_da7219_max98357a", + .quirk_data = &cml_spk_codecs, + .sof_fw_filename = "sof-cnl.ri", + .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + }, + { .id = "MX98357A", .drv_name = "sof_rt5682", .quirk_data = &cml_codecs, diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index 4718fd3..e6357d3 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -36,6 +36,7 @@ SOC_INTEL_IS_CPU(byt, INTEL_FAM6_ATOM_SILVERMONT); SOC_INTEL_IS_CPU(cht, INTEL_FAM6_ATOM_AIRMONT); SOC_INTEL_IS_CPU(apl, INTEL_FAM6_ATOM_GOLDMONT); SOC_INTEL_IS_CPU(glk, INTEL_FAM6_ATOM_GOLDMONT_PLUS); +SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); static inline bool soc_intel_is_byt_cr(struct platform_device *pdev) { @@ -110,6 +111,10 @@ static inline bool soc_intel_is_glk(void) return false; } +static inline bool soc_intel_is_cml(void) +{ + return false; +} #endif #endif /* _SND_SOC_INTEL_QUIRKS_H */ -- 2.7.4

3 2

[alsa-devel] [PATCH v2 1/2] ALSA: hda - Add a new match function for only undef configurations
by Hui Wang 15 Aug '19

15 Aug '19

With the existing pintbl, we already have many entries in it. it is better to figure out a new match to reduce the size of the pintbl. For example, there are over 10 entries in the pintbl for: 0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE If we define a new tbl like below, and with the new adding match function, we can remove those over 10 entries: SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1a, 0x40000000},), Here we put 0x19 and 0x1a in the tbl just because these two pins are undefined on Dell laptops with the codec alc255, and these two pins will be overwritten by ALC255_FIXUP_DELL1_MIC_NO_PRESENCE. In summary: the new match will check vendor id and codec id first, then check the pin_cfg defined in the tbl, only all pin_cfgs in the tbl are undef and the corresponding pin_cfgs on the laptop are undef too, this match returns true. This new match function has lower priority than existing match functions, so the existing tbls still work as before after applying this patch. And for the same vendor and same codec, we could define at most one undef tbl for it, for example for the Dell laptops with alc255 codec, some of them need to apply DELL1_MIC, the rest need to apply DELL2_MIC, here we could only define undef tbl for DELL1_MIC or DELL2_MIC, we could not define undef tbls for both of them. My plan is to change the existing tbl to undef tbl for MIC_NO_PRESENCE fixups gradually. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 62 ++++++++++++++++++++++++++++----- 1 file changed, 54 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 92390d457567..9bc2bfa0128c 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -915,6 +915,53 @@ static bool pin_config_match(struct hda_codec *codec, return true; } +/* Match the undef tbl with the initial verb table on the machine. + * Two nitices here: + * - define 1 undef tbl at most for the same vendor and same codec + * - the undef tbl has lower priority than normal tbls for matching + */ +static bool pin_config_match_undef(struct hda_codec *codec, + const struct hda_pintbl *pins) +{ + bool match = false; + + for (; pins->nid; pins++) { + const struct hda_pincfg *pin; + int i; + + if ((pins->val & 0xf0000000) != 0x40000000) + return false; + + match = false; + snd_array_for_each(&codec->init_pins, i, pin) { + if (pin->nid != pins->nid) + continue; + if ((pin->cfg & 0xf0000000) != 0x40000000) + return false; + match = true; + break; + } + + if (match == false) + return false; + } + + return match; +} + +static void snd_hda_set_found_fixup(struct hda_codec *codec, + const struct snd_hda_pin_quirk *pq, + const struct hda_fixup *fixlist) +{ + codec->fixup_id = pq->value; +#ifdef CONFIG_SND_DEBUG_VERBOSE + codec->fixup_name = pq->name; + codec_dbg(codec, "%s: picked fixup %s (pin match)\n", + codec->core.chip_name, codec->fixup_name); +#endif + codec->fixup_list = fixlist; +} + /** * snd_hda_pick_pin_fixup - Pick up a fixup matching with the pin quirk list * @codec: the HDA codec @@ -925,7 +972,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, const struct hda_fixup *fixlist) { - const struct snd_hda_pin_quirk *pq; + const struct snd_hda_pin_quirk *pq, *undef_bak_pq = NULL; if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET) return; @@ -936,16 +983,15 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, if (codec->core.vendor_id != pq->codec) continue; if (pin_config_match(codec, pq->pins)) { - codec->fixup_id = pq->value; -#ifdef CONFIG_SND_DEBUG_VERBOSE - codec->fixup_name = pq->name; - codec_dbg(codec, "%s: picked fixup %s (pin match)\n", - codec->core.chip_name, codec->fixup_name); -#endif - codec->fixup_list = fixlist; + snd_hda_set_found_fixup(codec, pq, fixlist); return; } + if (pin_config_match_undef(codec, pq->pins)) + undef_bak_pq = pq; } + + if (undef_bak_pq != NULL) + snd_hda_set_found_fixup(codec, undef_bak_pq, fixlist); } EXPORT_SYMBOL_GPL(snd_hda_pick_pin_fixup); -- 2.17.1

2 2

[alsa-devel] [PATCH 1/2] ALSA: hda - Add a new match function for only undef configurations
by Hui Wang 15 Aug '19

15 Aug '19

With the existing pintbl, we already have many entries in it. it is better to figure out a new match to reduce the size of the pintbl. For example, there are over 10 entries in the pintbl for: 0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE If we define a new tbl like below, and with the new adding match function, we can remove those over 10 entries: SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1a, 0x40000000},), Here we put 0x19 and 0x1a in the tbl just because these two pins are undefined on Dell laptops with the codec alc255, and these two pins will be overwritten by ALC255_FIXUP_DELL1_MIC_NO_PRESENCE. In summary: the new match will check vendor id and codec id first, then check the pin_cfg defined in the tbl, only all pin_cfgs in the tbl are undef and the corresponding pin_cfgs on the laptop are undef too, this match returns true. This new match function has lower priority than existing match functions, so the existing tbls still work as before after applying this patch. My plan is to change the existing tbl to undef tbl for MIC_NO_PRESENCE fixups gradually. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 32 +++++++++++++++++++++++++++++++- 1 file changed, 31 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 92390d457567..cfada7401b86 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -915,6 +915,36 @@ static bool pin_config_match(struct hda_codec *codec, return true; } +/* match the pintbl which only contains specific pins with undef configuration */ +static bool pin_config_match_undef(struct hda_codec *codec, + const struct hda_pintbl *pins) +{ + bool match = false; + + for (; pins->nid; pins++) { + const struct hda_pincfg *pin; + int i; + + if ((pins->val & 0xf0000000) != 0x40000000) + return false; + + match = false; + snd_array_for_each(&codec->init_pins, i, pin) { + if (pin->nid != pins->nid) + continue; + if ((pin->cfg & 0xf0000000) != 0x40000000) + return false; + match = true; + break; + } + + if (match == false) + return false; + } + + return match; +} + /** * snd_hda_pick_pin_fixup - Pick up a fixup matching with the pin quirk list * @codec: the HDA codec @@ -935,7 +965,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, continue; if (codec->core.vendor_id != pq->codec) continue; - if (pin_config_match(codec, pq->pins)) { + if (pin_config_match(codec, pq->pins) || pin_config_match_undef(codec, pq->pins)) { codec->fixup_id = pq->value; #ifdef CONFIG_SND_DEBUG_VERBOSE codec->fixup_name = pq->name; -- 2.17.1

2 4

[alsa-devel] [PATCH 0/4] A few more HD-audio core fixes / cleanups
by Takashi Iwai 14 Aug '19

14 Aug '19

Hi, this is a series of yet another patches for HD-audio core stuff. The first two are workarounds for possible races in the driver registration and other two are merely cleanups. Takashi === Takashi Iwai (4): ALSA: hda - Drop unsol event handler for Intel HDMI codecs ALSA: hda: Add codec on bus address table lately ALSA: hda: Drop export of snd_hdac_bus_add/remove_device() ALSA: hda: Unexport a few more stuff include/sound/hdaudio.h | 9 --------- sound/hda/hdac_bus.c | 8 ++++---- sound/hda/hdac_device.c | 27 ++++++++++++++------------- sound/hda/hdac_regmap.c | 1 + sound/hda/local.h | 7 +++++++ sound/pci/hda/patch_hdmi.c | 9 ++++++++- 6 files changed, 34 insertions(+), 27 deletions(-) -- 2.16.4

1 4

[alsa-devel] [PATCH] Fix an OOB bug in parse_audio_mixer_unit
by Hui Peng 14 Aug '19

14 Aug '19

The `uac_mixer_unit_descriptor` shown as below is read from the device side. In `parse_audio_mixer_unit`, `baSourceID` field is accessed from index 0 to `bNrInPins` - 1, the current implementation assumes that descriptor is always valid (the length of descriptor is no shorter than 5 + `bNrInPins`). If a descriptor read from the device side is invalid, it may trigger out-of-bound memory access. ``` struct uac_mixer_unit_descriptor { __u8 bLength; __u8 bDescriptorType; __u8 bDescriptorSubtype; __u8 bUnitID; __u8 bNrInPins; __u8 baSourceID[]; } ``` This patch fixes the bug by add a sanity check on the length of the descriptor. Signed-off-by: Hui Peng <benquike(a)gmail.com> Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> --- sound/usb/mixer.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 7498b5191b68..38202ce67237 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2091,6 +2091,15 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, struct usb_audio_term iterm; int input_pins, num_ins, num_outs; int pin, ich, err; + int desc_len = (int) ((unsigned long) state->buffer + + state->buflen - (unsigned long) raw_desc); + + if (desc_len < sizeof(*desc) + desc->bNrInPins) { + usb_audio_err(state->chip, + "descriptor %d too short\n", + unitid); + return -EINVAL; + } err = uac_mixer_unit_get_channels(state, desc); if (err < 0) { -- 2.22.1

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