February 2016 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH 18/19] ASoC: Intel: Atom: add support for RT5642
by Luka Karinja 11 Feb '16

11 Feb '16

> The machine driver is not loaded when the BIOS uses the 10EC5642 > _HID. Add it to the white list of known _HIDs, codec_name is > already taken care of by previous commit > > Tested on Asus T100TAF. > I'm trying to get sound working on my T100TAF i did apply all the 19 patches on top of broonie/sound and added UCM files to my system. Does the TAF need different UCM file than the T100TA? aplay -l **** List of PLAYBACK Hardware Devices **** card 0: bytcrrt5640 [bytcr-rt5640], device 0: Baytrail Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: bytcrrt5640 [bytcr-rt5640], device 1: Deep-Buffer Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 dmesg: [ 5.209239] Adding 1049596k swap on /dev/mmcblk0p4. Priority:-1 extents:1 across:1049596k SSFS [ 5.219099] input: PC Speaker as /devices/platform/pcspkr/input/input6 [ 5.273802] i2c i2c-10: Added multiplexed i2c bus 13 [ 5.274629] intel_sst_acpi 80860F28:00: LPE base: 0x50a00000 size:0x200000 [ 5.274638] intel_sst_acpi 80860F28:00: IRAM base: 0x50ac0000 [ 5.274671] intel_sst_acpi 80860F28:00: DRAM base: 0x50b00000 [ 5.274685] intel_sst_acpi 80860F28:00: SHIM base: 0x50b40000 [ 5.274700] intel_sst_acpi 80860F28:00: Mailbox base: 0x50b44000 [ 5.274712] intel_sst_acpi 80860F28:00: DDR base: 0x20000000 [ 5.274842] intel_sst_acpi 80860F28:00: Got drv data max stream 25 [ 5.290156] Error: Driver 'pcspkr' is already registered, aborting... [ 5.311924] random: nonblocking pool is initialized [ 5.397912] EXT4-fs (mmcblk0p2): mounted filesystem with ordered data mode. Opts: data=ordered [ 5.506882] genirq: Flags mismatch irq 8. 00000083 (volume_down) vs. 00000000 (rtc0) [ 5.507025] gpio-keys gpio-keys.0.auto: Unable to claim irq 8; error -16 [ 5.507283] gpio-keys: probe of gpio-keys.0.auto failed with error -16 [ 5.507518] input: gpio-keys as /devices/platform/gpio-keys.1.auto/input/input8 [ 5.519513] iTCO_vendor_support: vendor-support=0 [ 5.591021] iTCO_wdt: Intel TCO WatchDog Timer Driver v1.11 [ 5.591153] iTCO_wdt: Found a Bay Trail SoC TCO device (Version=3, TCOBASE=0x0460) [ 5.604447] iTCO_wdt: initialized. heartbeat=30 sec (nowayout=0) [ 5.613586] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> media-cpu-dai mapping ok [ 5.613794] bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> deepbuffer-cpu-dai mapping ok [ 5.613922] compress asoc: snd-soc-dummy-dai <-> compress-cpu-dai mapping ok [ 5.615555] bytcr_rt5640 bytcr_rt5640: rt5640-aif1 <-> ssp2-port mapping ok [ 5.642183] asus_wmi: ASUS WMI generic driver loaded [ 5.663786] usbcore: registered new interface driver usbmouse [ 5.665362] usbcore: registered new interface driver usbkbd [ 5.681990] asus_wmi: Initialization: 0x1 [ 5.682121] asus_wmi: BIOS WMI version: 7.9 [ 5.682219] asus_wmi: SFUN value: 0x37 [ 5.685337] input: Asus WMI hotkeys as /devices/platform/asus-nb-wmi/input/input9 [ 5.708007] asus_wmi: Number of fans: 1 [ 5.801413] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 5.801529] input: SIS0817:00 0457:1071 as /devices/platform/80860F41:03/i2c-11/i2c-SIS0817:00/0018:0457:1071.0004/input/input10 [ 5.802167] hid-multitouch 0018:0457:1071.0004: input,hidraw3: I2C HID v1.00 Device [SIS0817:00 0457:1071] on i2c-SIS0817:00 [ 5.816775] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 5.816996] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 5.817204] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 5.817405] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 5.817616] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 5.818884] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 5.819109] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! [ 5.911351] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 5.911578] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 5.911787] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 5.911997] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 5.912201] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 5.912417] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 5.912634] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 5.915142] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! [ 6.042680] intel_rapl: Found RAPL domain package [ 6.042692] intel_rapl: Found RAPL domain core [ 6.247467] Bluetooth: BNEP (Ethernet Emulation) ver 1.3 [ 6.247489] Bluetooth: BNEP filters: protocol multicast [ 6.247515] Bluetooth: BNEP socket layer initialized [ 7.524852] brcmf_sdio_drivestrengthinit: No SDIO Drive strength init done for chip 43340 rev 2 pmurev 20 [ 7.525341] usbcore: registered new interface driver brcmfmac [ 7.722004] brcmf_c_preinit_dcmds: Firmware version = wl0: Jun 17 2014 11:48:43 version 6.10.190.49 (r485840) FWID 01-8744bcd2 [ 7.739335] brcmf_cfg80211_reg_notifier: not a ISO3166 code [ 7.899345] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 7.921532] audit: type=1006 audit(1450805770.516:2): pid=1018 uid=0 old-auid=4294967295 auid=1000 old-ses=4294967295 ses=1 res=1 [ 7.984230] audit: type=1006 audit(1450805770.580:3): pid=1091 uid=0 old-auid=4294967295 auid=1000 old-ses=4294967295 ses=2 res=1 [ 7.996410] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 8.151283] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 9.038571] IPv6: ADDRCONF(NETDEV_UP): wlan0: link is not ready [ 11.245916] Bluetooth: hci1: BCM: chip id 82 [ 11.245928] Bluetooth: hci1: BCM43341B0 (002.001.014) build 0000 [ 11.755123] Bluetooth: hci1: BCM (002.001.014) build 0164 [ 14.513027] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x0 fw_state 0x1 [ 14.513045] intel_sst_acpi 80860F28:00: fw download failed -16 [ 14.513067] intel_sst_acpi 80860F28:00: FW download fail -16 [ 14.513864] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.514676] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.515574] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.518058] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.518941] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.520293] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.525383] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.527248] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.528335] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 14.534111] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port

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[alsa-devel] [PATCH 0/6 v2] ALSA: dice: constrain PCM substreams to current sampling transfer frequency
by Takashi Sakamoto 11 Feb '16

11 Feb '16

Hi, This patchset is to update my former RFCv2 and for merge request to upstream. [alsa-devel][RFC][PATCH 00/10 v2] ALSA: dice: stabiliza packet streaming http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101897.h… Current ALSA Dice driver is written with somewhat ignoring actual hardware design to perform 'dynamic sample rate selection'. As a result, the driver includes some behaviors against users' expectation or functional misleading. This patchset purges such over-engineering. As a result, userspace applications can start PCM substreams just at current sampling transfer frequency. If users want to start at different sampling rate, they must set favorite rate in advance by tools. Currently, ffado-mixer with ffado-dbus-server[1] and hinawa-dice-common-cui[2] are available as such tool. For technical details, please refer to the post or our discussion in RFCv1: [alsa-devel] [RFC][PATCH 0/8] ALSA: dice: constrain PCM substreams to current sampling transfer frequency http://mailman.alsa-project.org/pipermail/alsa-devel/2015-November/100525.h… [1] as a part of FFADO utility http://subversion.ffado.org/ [2] in hinawa-utils repository https://github.com/takaswie/hinawa-utils Regards Takashi Sakamoto (6): ALSA: dice: limit to current sampling transfer frequency ALSA: dice: limit stream to current sampling transfer frequency. ALSA: dice: add MIDI ports according to current number of MIDI substreams ALSA: dice: get the number of MBLA data channel at opening PCM substream ALSA: dice: purge generating channel cache ALSA: dice: ensure phase lock before starting streaming sound/firewire/dice/dice-midi.c | 23 +++- sound/firewire/dice/dice-pcm.c | 199 +++++++++------------------------ sound/firewire/dice/dice-stream.c | 69 +++++++----- sound/firewire/dice/dice-transaction.c | 54 --------- sound/firewire/dice/dice.c | 67 +---------- sound/firewire/dice/dice.h | 8 -- 6 files changed, 116 insertions(+), 304 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] ASoC: sigmadsp: Fix missleading return value
by pascal.huerst＠gmail.com 10 Feb '16

10 Feb '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> Forwarding the return value of i2c_master_send, leads to errors later on, since i2c_master_send returns the number of bytes transmittet. Check for ret < 0 instead and return 0 otherwise. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/sigmadsp-i2c.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c index 21ca3a5..d374c18 100644 --- a/sound/soc/codecs/sigmadsp-i2c.c +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -31,7 +31,10 @@ static int sigmadsp_write_i2c(void *control_data, kfree(buf); - return ret; + if (ret < 0) + return ret; + + return 0; } static int sigmadsp_read_i2c(void *control_data, -- 2.4.3

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[alsa-devel] [PATCH] ASoC: arizona: fref must be limited in pseudo-fractional mode
by Richard Fitzgerald 10 Feb '16

10 Feb '16

When the FLL is in pseudo-fractional mode there is an additional limit on fref based on the fratio, to prevent aliasing around the Nyquist frequency. If fref exceeds this limit the refclk divider must be increased and the calculation tried again until a suitable combination of fref and fratio is found or we have to fall back to integer mode. This patch also adds some debug log prints around this code. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 43 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 42 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a32cfb8..913cfa8 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1899,6 +1899,25 @@ static struct { { 1000000, 13500000, 0, 1 }, }; +static const unsigned int pseudo_fref_max[ARIZONA_FLL_MAX_FRATIO] = { + 13500000, + 6144000, + 6144000, + 3072000, + 3072000, + 2822400, + 2822400, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 1536000, + 768000, +}; + static struct { unsigned int min; unsigned int max; @@ -2012,16 +2031,32 @@ static int arizona_calc_fratio(struct arizona_fll *fll, /* Adjust FRATIO/refdiv to avoid integer mode if possible */ refdiv = cfg->refdiv; + arizona_fll_dbg(fll, "pseudo: initial ratio=%u fref=%u refdiv=%u\n", + init_ratio, Fref, refdiv); + while (div <= ARIZONA_FLL_MAX_REFDIV) { for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { if ((ARIZONA_FLL_VCO_CORNER / 2) / - (fll->vco_mult * ratio) < Fref) + (fll->vco_mult * ratio) < Fref) { + arizona_fll_dbg(fll, "pseudo: hit VCO corner\n"); break; + } + + if (Fref > pseudo_fref_max[ratio - 1]) { + arizona_fll_dbg(fll, + "pseudo: exceeded max fref(%u) for ratio=%u\n", + pseudo_fref_max[ratio - 1], + ratio); + break; + } if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); return ratio; } } @@ -2030,6 +2065,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll, if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); return ratio; } } @@ -2038,6 +2076,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll, Fref /= 2; refdiv++; init_ratio = arizona_find_fratio(Fref, NULL); + arizona_fll_dbg(fll, + "pseudo: change fref=%u refdiv=%d(%d) ratio=%u\n", + Fref, refdiv, div, init_ratio); } arizona_fll_warn(fll, "Falling back to integer mode operation\n"); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: sigmadsp: Fix missleading return value
by pascal.huerst＠gmail.com 10 Feb '16

10 Feb '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> Forwarding the return value of i2c_master_send, leads to errors later on, since i2c_master_send returns the number of bytes transmittet. Check for ret < 0 instead and return 0 otherwise. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/sigmadsp-i2c.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c index 21ca3a5..d374c18 100644 --- a/sound/soc/codecs/sigmadsp-i2c.c +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -31,7 +31,10 @@ static int sigmadsp_write_i2c(void *control_data, kfree(buf); - return ret; + if (ret < 0) + return ret; + + return 0; } static int sigmadsp_read_i2c(void *control_data, -- 2.4.3

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[alsa-devel] topic/hda-mst branch merged to for-next
by Takashi Iwai 10 Feb '16

10 Feb '16

JFYI, I merged topic/hda-mst branch to for-next branch for wider testing. It means that this will be included in 4.6. I'll keep topic/hda-mst branch for a while, but for future patches, it might be better to based on for-next branch. Takashi

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[alsa-devel] [PATCH] ASoC: cs4271: add regulator consumer support
by pascal.huerst＠gmail.com 10 Feb '16

10 Feb '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> The cs4271 has three power domains: vd, vl and va. Enable them all, as long as the codec is in use. While at it, factored out the reset code into its own function. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- Documentation/devicetree/bindings/sound/cs4271.txt | 7 +++ sound/soc/codecs/cs4271.c | 69 +++++++++++++++++++--- 2 files changed, 68 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index e2cd1d7..6e699ce 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -33,12 +33,19 @@ Optional properties: Note that this is not needed in case the clocks are stable throughout the entire runtime of the codec. + - vd-supply: Digital power + - vl-supply: Logic power + - va-supply: Analog Power + Examples: codec_i2c: cs4271@10 { compatible = "cirrus,cs4271"; reg = <0x10>; reset-gpio = <&gpio 23 0>; + vd-supply = <&vdd_3v3_reg>; + vl-supply = <&vdd_3v3_reg>; + va-supply = <&vdd_3v3_reg>; }; codec_spi: cs4271@0 { diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e770ee6..0c0010b 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/tlv.h> @@ -157,6 +158,10 @@ static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) return reg == CS4271_CHIPID; } +static const char * const supply_names[] = { + "vd", "vl", "va" +}; + struct cs4271_private { unsigned int mclk; bool master; @@ -170,6 +175,7 @@ struct cs4271_private { int gpio_disable; /* enable soft reset workaround */ bool enable_soft_reset; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget cs4271_dapm_widgets[] = { @@ -487,6 +493,20 @@ static struct snd_soc_dai_driver cs4271_dai = { .symmetric_rates = 1, }; +static int cs4271_reset(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(cs4271->gpio_nreset)) { + gpio_set_value(cs4271->gpio_nreset, 0); + mdelay(1); + gpio_set_value(cs4271->gpio_nreset, 1); + mdelay(1); + } + + return 0; +} + #ifdef CONFIG_PM static int cs4271_soc_suspend(struct snd_soc_codec *codec) { @@ -499,6 +519,9 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec) if (ret < 0) return ret; + regcache_mark_dirty(cs4271->regmap); + regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); + return 0; } @@ -507,6 +530,16 @@ static int cs4271_soc_resume(struct snd_soc_codec *codec) int ret; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + + /* Do a proper reset after power up */ + cs4271_reset(codec); + /* Restore codec state */ ret = regcache_sync(cs4271->regmap); if (ret < 0) @@ -553,19 +586,24 @@ static int cs4271_codec_probe(struct snd_soc_codec *codec) } #endif + ret = regulator_bulk_enable(ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + if (cs4271plat) { amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } - if (gpio_is_valid(cs4271->gpio_nreset)) { - /* Reset codec */ - gpio_direction_output(cs4271->gpio_nreset, 0); - mdelay(1); - gpio_set_value(cs4271->gpio_nreset, 1); - /* Give the codec time to wake up */ - mdelay(1); - } + /* Reset codec */ + cs4271_reset(codec); + + ret = regcache_sync(cs4271->regmap); + if (ret < 0) + return ret; ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, @@ -595,6 +633,9 @@ static int cs4271_codec_remove(struct snd_soc_codec *codec) /* Set codec to the reset state */ gpio_set_value(cs4271->gpio_nreset, 0); + regcache_mark_dirty(cs4271->regmap); + regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); + return 0; }; @@ -617,6 +658,7 @@ static int cs4271_common_probe(struct device *dev, { struct cs4271_platform_data *cs4271plat = dev->platform_data; struct cs4271_private *cs4271; + int i, ret; cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL); if (!cs4271) @@ -638,6 +680,17 @@ static int cs4271_common_probe(struct device *dev, return ret; } + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + cs4271->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + + if (ret < 0) { + dev_err(dev, "Failed to get regulators: %d\n", ret); + return ret; + } + *c = cs4271; return 0; } -- 2.4.3

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[alsa-devel] [PATCH] MAINTAINERS: add msm mailing list to sound/soc/qcom
by Srinivas Kandagatla 10 Feb '16

10 Feb '16

This patch adds msm mailing list to sound/soc/qcom so that get_maintainers would return correct list to CC to. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 7f1fa4f..213be2a 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5820,6 +5820,7 @@ F: include/uapi/linux/ipmi* QCOM AUDIO (ASoC) DRIVERS M: Patrick Lai <plai(a)codeaurora.org> M: Banajit Goswami <bgoswami(a)codeaurora.org> +L: linux-arm-msm(a)vger.kernel.org L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/qcom/ -- 1.9.1

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[alsa-devel] Should snd_card_free() check for null pointer?
by Jerome Marchand 10 Feb '16

10 Feb '16

Hi, Before commit f24640648186b (ALSA: Use standard device refcount for card accounting), snd_card_free() would return -EINVAL on a null pointer. Now it ends up in a null pointer dereference. There is at least one driver that can call snd_card_free() with null argument: saa7134_alsa. It can easily be triggered by just inserting and removing the module (no need to have the hardware). I don't think that is a rule, but it seems that the standard behavior of *_free() functions is to check for null pointer. What do you think? Thanks, Jerome

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[alsa-devel] [PATCH] ASoC: rt5645: add "coreboot" to dmi_system_id list
by Bard Liao 09 Feb '16

09 Feb '16

There are some Google projects with manufacturer name different from "GOOGLE". So we need to add DMI list for those projects. However some OEMs / vendors want to keep secret before product launch. So, we can't use DMI_SYS_VENDOR for them. Fortunately, the BIOS vendor field is the same for all chrome os projects. It allows us to match chrome projects by DMI_BIOS_VENDOR = "coreboot". Signed-off-by: John Lin <john.lin(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 93e8c90..8521b22 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3552,6 +3552,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"), }, }, + { + .ident = "Google Chrome", + .matches = { + DMI_MATCH(DMI_BIOS_VENDOR, "coreboot"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

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