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

[alsa-devel] [PATCH v2] ASoC: arizona: Add support for SNDRV_PCM_RATE_KNOT
by Charles Keepax 05 Feb '16

05 Feb '16

The Arizona CODECs support several rates that do not have simple defines in ALSA. This patch adds support for SNDRV_PCM_RATE_KNOT so that users can open stream at these rates. As part of this we should always set constraints in arizona_startup, currently we only set the constraints if we already have a clock to limit rates to that family of sample rates. This patch updates this to set a constraint of all rates supported by the chip if we do not already know which family of rates to limit to. Finally we also reduce the list of rates supported in the constraints to only include those that are supported on current parts. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Thanks, for the pointer there Lars, sorry for missing what you were talking about the first time. It does indeed look much nicer this way. Changes since v1: - Use the mask field in the constaint list as suggested by Lars. Thanks, Charles sound/soc/codecs/arizona.c | 62 ++++++++++++---------------------------------- sound/soc/codecs/arizona.h | 2 ++ sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- 7 files changed, 23 insertions(+), 51 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 33143fe..a32cfb8 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1398,29 +1398,6 @@ static const int arizona_48k_bclk_rates[] = { 24576000, }; -static const unsigned int arizona_48k_rates[] = { - 12000, - 24000, - 48000, - 96000, - 192000, - 384000, - 768000, - 4000, - 8000, - 16000, - 32000, - 64000, - 128000, - 256000, - 512000, -}; - -static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = { - .count = ARRAY_SIZE(arizona_48k_rates), - .list = arizona_48k_rates, -}; - static const int arizona_44k1_bclk_rates[] = { -1, 44100, @@ -1443,22 +1420,7 @@ static const int arizona_44k1_bclk_rates[] = { 22579200, }; -static const unsigned int arizona_44k1_rates[] = { - 11025, - 22050, - 44100, - 88200, - 176400, - 352800, - 705600, -}; - -static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = { - .count = ARRAY_SIZE(arizona_44k1_rates), - .list = arizona_44k1_rates, -}; - -static int arizona_sr_vals[] = { +static const unsigned int arizona_sr_vals[] = { 0, 12000, 24000, @@ -1485,13 +1447,21 @@ static int arizona_sr_vals[] = { 512000, }; +#define ARIZONA_48K_RATE_MASK 0x0F003E +#define ARIZONA_44K1_RATE_MASK 0x003E00 +#define ARIZONA_RATE_MASK (ARIZONA_48K_RATE_MASK | ARIZONA_44K1_RATE_MASK) + +static const struct snd_pcm_hw_constraint_list arizona_constraint = { + .count = ARRAY_SIZE(arizona_sr_vals), + .list = arizona_sr_vals, +}; + static int arizona_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; - const struct snd_pcm_hw_constraint_list *constraint; unsigned int base_rate; if (!substream->runtime) @@ -1509,16 +1479,15 @@ static int arizona_startup(struct snd_pcm_substream *substream, } if (base_rate == 0) - return 0; - - if (base_rate % 8000) - constraint = &arizona_44k1_constraint; + dai_priv->constraint.mask = ARIZONA_RATE_MASK; + else if (base_rate % 8000) + dai_priv->constraint.mask = ARIZONA_44K1_RATE_MASK; else - constraint = &arizona_48k_constraint; + dai_priv->constraint.mask = ARIZONA_48K_RATE_MASK; return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - constraint); + &dai_priv->constraint); } static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, @@ -1911,6 +1880,7 @@ int arizona_init_dai(struct arizona_priv *priv, int id) struct arizona_dai_priv *dai_priv = &priv->dai[id]; dai_priv->clk = ARIZONA_CLK_SYSCLK; + dai_priv->constraint = arizona_constraint; return 0; } diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 0c64a57..1ea8e4e 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -68,6 +68,8 @@ struct wm_adsp; struct arizona_dai_priv { int clk; + + struct snd_pcm_hw_constraint_list constraint; }; struct arizona_priv { diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index b8d3cd0..576087b 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -923,7 +923,7 @@ static int cs47l24_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define CS47L24_RATES SNDRV_PCM_RATE_8000_192000 +#define CS47L24_RATES SNDRV_PCM_RATE_KNOT #define CS47L24_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index aa1e96a..a8b3e3f 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1738,7 +1738,7 @@ static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM5102_RATES SNDRV_PCM_RATE_8000_192000 +#define WM5102_RATES SNDRV_PCM_RATE_KNOT #define WM5102_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 302a8d3..83ba70f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2024,7 +2024,7 @@ static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM5110_RATES SNDRV_PCM_RATE_8000_192000 +#define WM5110_RATES SNDRV_PCM_RATE_KNOT #define WM5110_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index b4dba3a..52d766e 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -943,7 +943,7 @@ static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } } -#define WM8997_RATES SNDRV_PCM_RATE_8000_192000 +#define WM8997_RATES SNDRV_PCM_RATE_KNOT #define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 7719bc5..0123960 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1170,7 +1170,7 @@ static const struct snd_soc_dapm_route wm8998_dapm_routes[] = { { "DRC1 Signal Activity", NULL, "DRC1R" }, }; -#define WM8998_RATES SNDRV_PCM_RATE_8000_192000 +#define WM8998_RATES SNDRV_PCM_RATE_KNOT #define WM8998_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -- 2.1.4

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[alsa-devel] [PATCH] ASoC: adau17x1: Cache writes when core clock is disabled
by Andreas Irestål 05 Feb '16

05 Feb '16

In some configurations, the dai registers get written before the bias level is changed in the codec driver. This leads to a situation where an initial write to the serial port register gets ignored, and future writes may as well, since regmap thinks that the codec already holds the value. More specifically, configuring the codec as i2s master would in fact result in the codec running as slave, a situation where no i2s clocks are generated and hence no data is transferred. This change makes sure that regmap only caches writes when the core clock is disabled, and syncs regmap whenever enabling the core clock again. Signed-off-by: Andreas Irestål <andire(a)axis.com> --- sound/soc/codecs/adau1761.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 2f12477..e7136b1 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -456,13 +456,17 @@ static int adau1761_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: + regcache_cache_only(adau->regmap, false); regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) + regcache_sync(adau->regmap); break; case SND_SOC_BIAS_OFF: regmap_update_bits(adau->regmap, ADAU17X1_CLOCK_CONTROL, ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, 0); + regcache_cache_only(adau->regmap, true); break; } @@ -783,6 +787,10 @@ int adau1761_probe(struct device *dev, struct regmap *regmap, if (ret) return ret; + /* Enable cache only mode as we could miss writes before bias level + * reaches standby and the core clock is enabled */ + regcache_cache_only(regmap, true); + return snd_soc_register_codec(dev, &adau1761_codec_driver, dai_drv, 1); } EXPORT_SYMBOL_GPL(adau1761_probe); -- 2.1.4

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[alsa-devel] [PATCH v2 1/2] ASoC: fsl-asoc-card: add cs4271 and cs4272 support
by Felipe F. Tonello 05 Feb '16

05 Feb '16

add cs4271 and cs42727 support for fsl-asoc-card Signed-off-by: Felipe F. Tonello <eu(a)felipetonello.com> --- Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 9 +++++++++ sound/soc/fsl/Kconfig | 4 ++-- sound/soc/fsl/fsl-asoc-card.c | 7 +++++++ 3 files changed, 18 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index 4da41bf1888e..ceaef5126989 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -24,6 +24,9 @@ The compatible list for this generic sound card currently: "fsl,imx-audio-cs42888" + "fsl,imx-audio-cs427x" + (compatible with CS4271 and CS4272) + "fsl,imx-audio-wm8962" (compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt) @@ -63,6 +66,12 @@ Optional properties: - audio-asrc : The phandle of ASRC. It can be absent if there's no need to add ASRC support via DPCM. +Optional unless SSI is selected as a CPU DAI: + + - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) + + - mux-ext-port : The external port of the i.MX audio muxer + Example: sound-cs42888 { compatible = "fsl,imx-audio-cs42888"; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 14dfdee05fd5..35aabf9dc503 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -292,8 +292,8 @@ config SND_SOC_FSL_ASOC_CARD select SND_SOC_FSL_SSI help ALSA SoC Audio support with ASRC feature for Freescale SoCs that have - ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 - and SGTL5000. + ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888, + CS4271, CS4272 and SGTL5000. Say Y if you want to add support for Freescale Generic ASoC Sound Card. endif # SND_IMX_SOC diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 562b3bd22d9a..56b6b07d1922 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -28,6 +28,8 @@ #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" +#define CS427x_SYSCLK_MCLK 0 + #define RX 0 #define TX 1 @@ -528,6 +530,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) { + codec_dai_name = "cs4271-hifi"; + priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; @@ -681,6 +687,7 @@ fail: static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, + { .compatible = "fsl,imx-audio-cs427x", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, -- 2.7.0

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[alsa-devel] [PATCH 0/4] ALSA: firewire-tascam: add support for FW-1804
by Takashi Sakamoto 05 Feb '16

05 Feb '16

Hi, This patchset adds support for FW-1804. Additionally it fix some bugs. Nothing else ;) This commit also adds no functionality to control internal multiplexer of supported models, following to basic design of ALSA firewire stack. It should be implemented in userspace. Currently, there's no udev rules for fw character devices corresponding to models supported by this module. Thus, permission of the devices may be 0600. When userspace applications touches the devices, they needs effective UID or GID as 0. This is not better environment for users. Further works are required for related softwares. Regards Takashi Sakamoto (4): ALSA: firewire-tascam: fix NULL pointer dereference when model identification fails ALSA: firewire-tascam: add support for FW-1804 ALSA: firewire-tascam: remove a flag for controller ALSA: firewire-tascam: remove needless member for control and status message sound/firewire/tascam/tascam-transaction.c | 6 ++++++ sound/firewire/tascam/tascam.c | 12 +++++++++--- sound/firewire/tascam/tascam.h | 4 ---- 3 files changed, 15 insertions(+), 7 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] ALSA: hda - Fix static checker warning in patch_hdmi.c
by David Henningsson 05 Feb '16

05 Feb '16

The static checker warning is: sound/pci/hda/patch_hdmi.c:460 hdmi_eld_ctl_get() error: __memcpy() 'eld->eld_buffer' too small (256 vs 512) I have a hard time figuring out if this can ever cause an information leak (I don't think so), but nonetheless it does not hurt to increase the robustness of the code. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1f52b55..2191e235 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -448,7 +448,8 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, eld = &per_pin->sink_eld; mutex_lock(&per_pin->lock); - if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data)) { + if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) || + eld->eld_size > ELD_MAX_SIZE) { mutex_unlock(&per_pin->lock); snd_BUG(); return -EINVAL; -- 1.9.1

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Re: [alsa-devel] ALSA: hda - hdmi: Do not expose eld data when eld is invalid
by Dan Carpenter 05 Feb '16

05 Feb '16

Hello David Henningsson, The patch 68e03de98507: "ALSA: hda - hdmi: Do not expose eld data when eld is invalid" from Feb 19, 2013, leads to the following static checker warning: sound/pci/hda/patch_hdmi.c:460 hdmi_eld_ctl_get() error: __memcpy() 'eld->eld_buffer' too small (256 vs 512) sound/pci/hda/patch_hdmi.c 437 static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, 438 struct snd_ctl_elem_value *ucontrol) 439 { 440 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 441 struct hdmi_spec *spec = codec->spec; 442 struct hdmi_spec_per_pin *per_pin; 443 struct hdmi_eld *eld; 444 int pin_idx; 445 446 pin_idx = kcontrol->private_value; 447 per_pin = get_pin(spec, pin_idx); 448 eld = &per_pin->sink_eld; 449 450 mutex_lock(&per_pin->lock); 451 if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data)) { In the original code we always copied ELD_MAX_SIZE (256) bytes. Now it looks like we can copy up to 512 bytes which seems like an information leak. I don't know where eld->eld_size is set so I can't say if this is a real issue or not. Perhaps it's always a valid value. 452 mutex_unlock(&per_pin->lock); 453 snd_BUG(); 454 return -EINVAL; 455 } 456 457 memset(ucontrol->value.bytes.data, 0, 458 ARRAY_SIZE(ucontrol->value.bytes.data)); 459 if (eld->eld_valid) 460 memcpy(ucontrol->value.bytes.data, eld->eld_buffer, 461 eld->eld_size); 462 mutex_unlock(&per_pin->lock); 463 464 return 0; 465 } regards, dan carpenter

1 0

[alsa-devel] linux-api scope (Re: [PATCH v2 11/22] media: dvb-frontend invoke enable/disable_source handlers)
by Andy Lutomirski 04 Feb '16

04 Feb '16

[cc list heavily trimmed] On Wed, Feb 3, 2016 at 8:03 PM, Shuah Khan <shuahkh(a)osg.samsung.com> wrote: > Change dvb frontend to check if tuner is free when > device opened in RW mode. Call to enable_source > handler either returns with an active pipeline to > tuner or error if tuner is busy. Tuner is released > when frontend is released calling the disable_source > handler. As an actual subscriber to linux-api, I prefer for the linux-api list to be lowish-volume and mostly limited to API-related things. Is this API related? Do people think that these series should be sent to linux-api? Thanks, Andy

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[alsa-devel] [PATCH v3 1/3] ALSA: mts64: whitespace fixes
by Sudip Mukherjee 04 Feb '16

04 Feb '16

To improve the readability and grep-ability of the code trailing whitespace, multiple blank lines, space before start of a line and a required space before opening brace have been removed. This patch takes care of these issues related to whitespace changes and there was no other change. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- v3: changed commit message v2: combined 3 patches of v1 series into this patch. sound/drivers/mts64.c | 112 ++++++++++++++++++++++++-------------------------- 1 file changed, 54 insertions(+), 58 deletions(-) diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 2a008a9..f7d938d 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -1,14 +1,14 @@ -/* +/* * ALSA Driver for Ego Systems Inc. (ESI) Miditerminal 4140 * Copyright (c) 2006 by Matthias König <mk(a)phasorlab.de> * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * @@ -38,7 +38,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; -static struct platform_device *platform_devices[SNDRV_CARDS]; +static struct platform_device *platform_devices[SNDRV_CARDS]; static int device_count; module_param_array(index, int, NULL, S_IRUGO); @@ -92,7 +92,7 @@ static int snd_mts64_create(struct snd_card *card, *rchip = NULL; mts = kzalloc(sizeof(struct mts64), GFP_KERNEL); - if (mts == NULL) + if (mts == NULL) return -ENOMEM; /* Init chip specific data */ @@ -118,9 +118,9 @@ static int snd_mts64_create(struct snd_card *card, /* Control Bits */ #define MTS64_CTL_READOUT 0x08 /* enable readout */ -#define MTS64_CTL_WRITE_CMD 0x06 -#define MTS64_CTL_WRITE_DATA 0x02 -#define MTS64_CTL_STROBE 0x01 +#define MTS64_CTL_WRITE_CMD 0x06 +#define MTS64_CTL_WRITE_DATA 0x02 +#define MTS64_CTL_STROBE 0x01 /* Command */ #define MTS64_CMD_RESET 0xfe @@ -130,7 +130,7 @@ static int snd_mts64_create(struct snd_card *card, #define MTS64_CMD_SMPTE_STOP 0xef #define MTS64_CMD_SMPTE_FPS_24 0xe3 #define MTS64_CMD_SMPTE_FPS_25 0xe2 -#define MTS64_CMD_SMPTE_FPS_2997 0xe4 +#define MTS64_CMD_SMPTE_FPS_2997 0xe4 #define MTS64_CMD_SMPTE_FPS_30D 0xe1 #define MTS64_CMD_SMPTE_FPS_30 0xe0 #define MTS64_CMD_COM_OPEN 0xf8 /* setting the communication mode */ @@ -159,7 +159,6 @@ static void mts64_write_command(struct parport *p, u8 c); static void mts64_write_data(struct parport *p, u8 c); static void mts64_write_midi(struct mts64 *mts, u8 c, int midiport); - /* Enables the readout procedure * * Before we can read a midi byte from the device, we have to set @@ -171,10 +170,10 @@ static void mts64_enable_readout(struct parport *p) c = parport_read_control(p); c |= MTS64_CTL_READOUT; - parport_write_control(p, c); + parport_write_control(p, c); } -/* Disables readout +/* Disables readout * * Readout is disabled by clearing bit 3 of control */ @@ -201,9 +200,9 @@ static int mts64_device_ready(struct parport *p) for (i = 0; i < 0xffff; ++i) { c = parport_read_status(p); c &= MTS64_STAT_BSY; - if (c != 0) + if (c != 0) return 1; - } + } return 0; } @@ -234,7 +233,7 @@ static int mts64_device_init(struct parport *p) return -EIO; } -/* +/* * Opens the device (set communication mode) */ static int mts64_device_open(struct mts64 *mts) @@ -248,7 +247,7 @@ static int mts64_device_open(struct mts64 *mts) return 0; } -/* +/* * Close device (clear communication mode) */ static int mts64_device_close(struct mts64 *mts) @@ -265,14 +264,14 @@ static int mts64_device_close(struct mts64 *mts) } /* map hardware port to substream number - * + * * When reading a byte from the device, the device tells us * on what port the byte is. This HW port has to be mapped to * the midiport (substream number). * substream 0-3 are Midiports 1-4 * substream 4 is SMPTE Timecode * The mapping is done by the table: - * HW | 0 | 1 | 2 | 3 | 4 + * HW | 0 | 1 | 2 | 3 | 4 * SW | 0 | 1 | 4 | 2 | 3 */ static u8 mts64_map_midi_input(u8 c) @@ -282,10 +281,9 @@ static u8 mts64_map_midi_input(u8 c) return map[c]; } - /* Probe parport for device * - * Do we have a Miditerminal 4140 on parport? + * Do we have a Miditerminal 4140 on parport? * Returns: * 0 device found * -ENODEV no device @@ -298,13 +296,13 @@ static int mts64_probe(struct parport *p) mts64_write_command(p, MTS64_CMD_PROBE); msleep(50); - + c = mts64_read(p); c &= 0x00ff; - if (c != MTS64_CMD_PROBE) + if (c != MTS64_CMD_PROBE) return -ENODEV; - else + else return 0; } @@ -330,7 +328,7 @@ static u16 mts64_read(struct parport *p) /* Read a byte from device * * Note, that readout mode has to be enabled. - * readout procedure is as follows: + * readout procedure is as follows: * - Write number of the Bit to read to DATA * - Read STATUS * - Bit 5 of STATUS indicates if Bit is set @@ -348,10 +346,10 @@ static u8 mts64_read_char(struct parport *p) parport_write_data(p, i); c >>= 1; status = parport_read_status(p); - if (status & MTS64_STAT_BIT_SET) + if (status & MTS64_STAT_BIT_SET) c |= 0x80; } - + return c; } @@ -369,9 +367,9 @@ static void mts64_smpte_start(struct parport *p, u8 seconds, u8 frames, u8 idx) { - static u8 fps[5] = { MTS64_CMD_SMPTE_FPS_24, + static u8 fps[5] = { MTS64_CMD_SMPTE_FPS_24, MTS64_CMD_SMPTE_FPS_25, - MTS64_CMD_SMPTE_FPS_2997, + MTS64_CMD_SMPTE_FPS_2997, MTS64_CMD_SMPTE_FPS_30D, MTS64_CMD_SMPTE_FPS_30 }; @@ -405,7 +403,7 @@ static void mts64_write_command(struct parport *p, u8 c) parport_write_control(p, MTS64_CTL_WRITE_CMD); } -/* Write a data byte to device +/* Write a data byte to device */ static void mts64_write_data(struct parport *p, u8 c) { @@ -652,7 +650,6 @@ static struct snd_kcontrol_new mts64_ctl_smpte_fps = { .put = snd_mts64_ctl_smpte_fps_put }; - static int snd_mts64_ctl_create(struct snd_card *card, struct mts64 *mts) { @@ -669,7 +666,7 @@ static int snd_mts64_ctl_create(struct snd_card *card, for (i = 0; control[i]; ++i) { err = snd_ctl_add(card, snd_ctl_new1(control[i], mts)); if (err < 0) { - snd_printd("Cannot create control: %s\n", + snd_printd("Cannot create control: %s\n", control[i]->name); return err; } @@ -688,8 +685,8 @@ static int snd_mts64_rawmidi_open(struct snd_rawmidi_substream *substream) struct mts64 *mts = substream->rmidi->private_data; if (mts->open_count == 0) { - /* We don't need a spinlock here, because this is just called - if the device has not been opened before. + /* We don't need a spinlock here, because this is just called + if the device has not been opened before. So there aren't any IRQs from the device */ mts64_device_open(mts); @@ -746,8 +743,8 @@ static void snd_mts64_rawmidi_input_trigger(struct snd_rawmidi_substream *substr if (up) mts->mode[substream->number] |= MTS64_MODE_INPUT_TRIGGERED; else - mts->mode[substream->number] &= ~MTS64_MODE_INPUT_TRIGGERED; - + mts->mode[substream->number] &= ~MTS64_MODE_INPUT_TRIGGERED; + spin_unlock_irqrestore(&mts->lock, flags); } @@ -771,42 +768,42 @@ static int snd_mts64_rawmidi_create(struct snd_card *card) struct snd_rawmidi_substream *substream; struct list_head *list; int err; - - err = snd_rawmidi_new(card, CARD_NAME, 0, - MTS64_NUM_OUTPUT_PORTS, - MTS64_NUM_INPUT_PORTS, + + err = snd_rawmidi_new(card, CARD_NAME, 0, + MTS64_NUM_OUTPUT_PORTS, + MTS64_NUM_INPUT_PORTS, &rmidi); - if (err < 0) + if (err < 0) return err; rmidi->private_data = mts; strcpy(rmidi->name, CARD_NAME); rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | - SNDRV_RAWMIDI_INFO_INPUT | - SNDRV_RAWMIDI_INFO_DUPLEX; + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; mts->rmidi = rmidi; /* register rawmidi ops */ - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mts64_rawmidi_output_ops); - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mts64_rawmidi_input_ops); /* name substreams */ /* output */ - list_for_each(list, + list_for_each(list, &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) { substream = list_entry(list, struct snd_rawmidi_substream, list); sprintf(substream->name, "Miditerminal %d", substream->number+1); } /* input */ - list_for_each(list, + list_for_each(list, &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams) { substream = list_entry(list, struct snd_rawmidi_substream, list); mts->midi_input_substream[substream->number] = substream; - switch(substream->number) { + switch (substream->number) { case MTS64_SMPTE_SUBSTREAM: strcpy(substream->name, "Miditerminal SMPTE"); break; @@ -840,7 +837,7 @@ static void snd_mts64_interrupt(void *private) if (status & MTS64_STAT_PORT) { mts->current_midi_input_port = mts64_map_midi_input(data); } else { - if (mts->current_midi_input_port == -1) + if (mts->current_midi_input_port == -1) goto __out; substream = mts->midi_input_substream[mts->current_midi_input_port]; if (mts->mode[substream->number] & MTS64_MODE_INPUT_TRIGGERED) @@ -860,7 +857,7 @@ static int snd_mts64_probe_port(struct parport *p) 0, NULL); if (!pardev) return -EIO; - + if (parport_claim(pardev)) { parport_unregister_device(pardev); return -EIO; @@ -944,7 +941,7 @@ static int snd_mts64_probe(struct platform_device *pdev) if (dev >= SNDRV_CARDS) return -ENODEV; - if (!enable[dev]) + if (!enable[dev]) return -ENOENT; if ((err = snd_mts64_probe_port(p)) < 0) return err; @@ -957,7 +954,7 @@ static int snd_mts64_probe(struct platform_device *pdev) } strcpy(card->driver, DRIVER_NAME); strcpy(card->shortname, "ESI " CARD_NAME); - sprintf(card->longname, "%s at 0x%lx, irq %i", + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, p->base, p->irq); pardev = parport_register_device(p, /* port */ @@ -980,7 +977,7 @@ static int snd_mts64_probe(struct platform_device *pdev) } card->private_data = mts; card->private_free = snd_mts64_card_private_free; - + if ((err = snd_mts64_rawmidi_create(card)) < 0) { snd_printd("Creating Rawmidi component failed\n"); goto __err; @@ -1024,7 +1021,6 @@ static int snd_mts64_remove(struct platform_device *pdev) return 0; } - static struct platform_driver snd_mts64_driver = { .probe = snd_mts64_probe, .remove = snd_mts64_remove, @@ -1045,7 +1041,7 @@ static void snd_mts64_unregister_all(void) platform_device_unregister(platform_devices[i]); platform_devices[i] = NULL; } - } + } platform_driver_unregister(&snd_mts64_driver); parport_unregister_driver(&mts64_parport_driver); } -- 1.9.1

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[alsa-devel] sound: use-after-free in snd_timer_start1
by Dmitry Vyukov 04 Feb '16

04 Feb '16

Hello, The following program triggers use-after-free in snd_timer_start1: // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> int main() { syscall(SYS_mmap, 0x20000000ul, 0x664000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); int fd = syscall(SYS_open, "/dev/snd/timer", 0x40000ul, 0, 0, 0); *(uint32_t*)0x20653000 = (uint32_t)0x1; *(uint32_t*)0x20653004 = (uint32_t)0xfffffffffffffffe; *(uint32_t*)0x20653008 = (uint32_t)0x8; *(uint32_t*)0x2065300c = (uint32_t)0x0; *(uint32_t*)0x20653010 = (uint32_t)0x0; *(uint8_t*)0x20653014 = (uint8_t)0x0; *(uint8_t*)0x20653015 = (uint8_t)0x0; *(uint8_t*)0x20653016 = (uint8_t)0x0; *(uint8_t*)0x20653017 = (uint8_t)0x0; *(uint8_t*)0x20653018 = (uint8_t)0x0; *(uint8_t*)0x20653019 = (uint8_t)0x0; *(uint8_t*)0x2065301a = (uint8_t)0x0; *(uint8_t*)0x2065301b = (uint8_t)0x0; *(uint8_t*)0x2065301c = (uint8_t)0x0; *(uint8_t*)0x2065301d = (uint8_t)0x0; *(uint8_t*)0x2065301e = (uint8_t)0x0; *(uint8_t*)0x2065301f = (uint8_t)0x0; *(uint8_t*)0x20653020 = (uint8_t)0x0; *(uint8_t*)0x20653021 = (uint8_t)0x0; *(uint8_t*)0x20653022 = (uint8_t)0x0; *(uint8_t*)0x20653023 = (uint8_t)0x0; *(uint8_t*)0x20653024 = (uint8_t)0x0; *(uint8_t*)0x20653025 = (uint8_t)0x0; *(uint8_t*)0x20653026 = (uint8_t)0x0; *(uint8_t*)0x20653027 = (uint8_t)0x0; *(uint8_t*)0x20653028 = (uint8_t)0x0; *(uint8_t*)0x20653029 = (uint8_t)0x0; *(uint8_t*)0x2065302a = (uint8_t)0x0; *(uint8_t*)0x2065302b = (uint8_t)0x0; *(uint8_t*)0x2065302c = (uint8_t)0x0; *(uint8_t*)0x2065302d = (uint8_t)0x0; *(uint8_t*)0x2065302e = (uint8_t)0x0; *(uint8_t*)0x2065302f = (uint8_t)0x0; *(uint8_t*)0x20653030 = (uint8_t)0x0; *(uint8_t*)0x20653031 = (uint8_t)0x0; *(uint8_t*)0x20653032 = (uint8_t)0x0; *(uint8_t*)0x20653033 = (uint8_t)0x0; syscall(SYS_ioctl, fd, 0x40345410ul, 0x20653000ul, 0, 0, 0); syscall(SYS_ioctl, fd, 0x54a0ul, 0, 0, 0, 0); return 0; } ================================================================== BUG: KASAN: use-after-free in __list_add+0x233/0x22016/02/03 13:44:32 executing program 0: Read of size 8 by task syz-executor/27661 INFO: Allocated in snd_timer_instance_new+0x52/0x3a0 age=7 cpu=0 pid=27660 [< inline >] slab_alloc_node mm/slub.c:2562 [< inline >] slab_alloc mm/slub.c:2604 [< none >] kmem_cache_alloc_trace+0x25c/0x300 mm/slub.c:2621 [< inline >] kmalloc include/linux/slab.h:463 [< inline >] kzalloc include/linux/slab.h:607 [< none >] snd_timer_instance_new+0x52/0x3a0 sound/core/timer.c:106 [< none >] snd_timer_open+0x522/0xd20 sound/core/timer.c:289 [< inline >] snd_timer_user_tselect sound/core/timer.c:1598 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1874 [< none >] snd_timer_user_ioctl+0x8db/0x25f0 sound/core/timer.c:1904 [< inline >] vfs_ioctl fs/ioctl.c:43 [< none >] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [< none >] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [< none >] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 INFO: Freed in snd_timer_close+0x3ee/0x750 age=9 cpu=3 pid=27661 [< inline >] slab_free mm/slub.c:2835 [< none >] kfree+0x2ac/0x2c0 mm/slub.c:3664 [< none >] snd_timer_close+0x3ee/0x750 sound/core/timer.c:375 [< inline >] snd_timer_user_tselect sound/core/timer.c:1588 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1874 [< none >] snd_timer_user_ioctl+0x7b6/0x25f0 sound/core/timer.c:1904 [< inline >] vfs_ioctl fs/ioctl.c:43 [< none >] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [< none >] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [< none >] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 Call Trace: [< inline >] kasan_report mm/kasan/report.c:274 [<ffffffff8176640e>] __asan_report_load8_noabort+0x3e/0x40 mm/kasan/report.c:295 [<ffffffff82c4a7e3>] __list_add+0x233/0x250 lib/list_debug.c:39 [< inline >] list_move_tail include/linux/list.h:77 [<ffffffff85240fdc>] snd_timer_start1+0x6c/0x2c0 sound/core/timer.c:432 [<ffffffff85241a7b>] snd_timer_start+0x20b/0x2b0 sound/core/timer.c:499 [< inline >] snd_timer_user_start sound/core/timer.c:1799 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1883 [<ffffffff85245fb7>] snd_timer_user_ioctl+0x767/0x25f0 sound/core/timer.c:1904 [< inline >] vfs_ioctl fs/ioctl.c:43 [<ffffffff817fa9ec>] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [<ffffffff817fb89f>] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [<ffffffff8665ebb6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 ================================================================== Sometimes it manifests as: ------------[ cut here ]------------ WARNING: CPU: 1 PID: 6548 at lib/list_debug.c:42 __list_add+0x1db/0x250() list_add corruption. prev->next should be next (ffff8800697c10e8), but was ffff880030441fd0. (prev=ffff880030441fd0). Modules linked in: CPU: 1 PID: 6548 Comm: a.out Not tainted 4.5.0-rc2+ #309 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 00000000ffffffff ffff88002fa17918 ffffffff82be2c8d ffff88002fa17988 ffff88002fd62f80 ffffffff86acf380 ffff88002fa17958 ffffffff81355139 ffffffff82c4a78b ffffed0005f42f2d ffffffff86acf380 000000000000002a Call Trace: [< inline >] __dump_stack lib/dump_stack.c:15 [<ffffffff82be2c8d>] dump_stack+0x6f/0xa2 lib/dump_stack.c:50 [<ffffffff81355139>] warn_slowpath_common+0xd9/0x140 kernel/panic.c:482 [<ffffffff81355249>] warn_slowpath_fmt+0xa9/0xd0 kernel/panic.c:494 [<ffffffff82c4a78b>] __list_add+0x1db/0x250 lib/list_debug.c:39 [< inline >] list_move_tail include/linux/list.h:77 [<ffffffff85240fdc>] snd_timer_start1+0x6c/0x2c0 sound/core/timer.c:432 [<ffffffff85241a7b>] snd_timer_start+0x20b/0x2b0 sound/core/timer.c:499 [< inline >] snd_timer_user_start sound/core/timer.c:1799 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1883 [<ffffffff85245fb7>] snd_timer_user_ioctl+0x767/0x25f0 sound/core/timer.c:1904 [< inline >] vfs_ioctl fs/ioctl.c:43 [<ffffffff817fa9ec>] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [<ffffffff817fb89f>] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [<ffffffff8665ebb6>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 ---[ end trace 77da812dd399e78b ]--- On commit 34229b277480f46c1e9a19f027f30b074512e68b + recent fixes from Takashi.

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[alsa-devel] [PATCH 00/11] ASoC: Intel: Skylake: Driver fixes
by Vinod Koul 04 Feb '16

04 Feb '16

Now that 4.5-rc1 has audio functional with driver, time to fix some corner cases in driver. This series tries to fix: o resource leakage on error cases while allocation and freeup o wrap of delay reporting o fix of various corner cases in topology and support of multiple sinks/source ona pipe and without a mixer termination o tlv buffer overwrite as suggested by Takashi o removal of autosuspend delay o 2 ch support on rt286 machine Although patches are 11, the diffstat is only +59, -29 Dharageswari.R (1): ASoC: Intel: Skylake: Fix pipe memory allocation leak Guneshwor Singh (2): ASoC: Intel: Skylake: Fix delay wrap condition ASoC: Intel: Skylake: Fix mcps freeup after module unbind failure Jeeja KP (6): ASoC: Intel: Skylake: Fix return of skl_get_queue_index ASoC: Intel: Skylake: Fix the module state check condition ASoC: Intel: Skylake: Fix not to stop sink pipe in pga pmd event ASoC: Intel: Skylake: Fix bind of source with multiple sinks ASoC: Intel: Skylake: Add missing PRE/POST_PMU handlers for vmixer ASoC: Intel: Skylake: Fix stereo DMIC record Vinod Koul (2): ASoC: Intel: Skylake: Fix the memory overwrite of tlv buffer ASoC: Intel: Skylake: Remove autosuspend delay sound/soc/intel/boards/skl_rt286.c | 5 ++- sound/soc/intel/skylake/skl-messages.c | 6 +-- sound/soc/intel/skylake/skl-pcm.c | 1 + sound/soc/intel/skylake/skl-topology.c | 74 +++++++++++++++++++++++----------- sound/soc/intel/skylake/skl.c | 2 - 5 files changed, 59 insertions(+), 29 deletions(-) -- 1.9.1

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