September 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: fsl: Do not force codecs selection by SND_SOC_FSL_ASOC_CARD
by Fabio Estevam 01 Oct '14

01 Oct '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> The wm8962 driver uses the input subsystem, but it is selected by SND_SOC_FSL_ASOC_CARD, which can be built with CONFIG_INPUT disabled, resulting in this link error: ERROR: "input_event" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! ERROR: "input_register_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! ERROR: "devm_input_allocate_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined! Do not force the selection of the codecs by SND_SOC_FSL_ASOC_CARD to avoid such problem. Reported-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 6164e78..081e406 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -288,9 +288,6 @@ config SND_SOC_FSL_ASOC_CARD select SND_SOC_FSL_ESAI select SND_SOC_FSL_SAI select SND_SOC_FSL_SSI - select SND_SOC_CS42XX8_I2C - select SND_SOC_SGTL5000 - select SND_SOC_WM8962 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 -- 1.9.1

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[alsa-devel] [BUG] Bad decibel data on C-Media USB sound card
by Allen M. 01 Oct '14

01 Oct '14

Raymond Yau wrote: >It affect your measurent if you are using the mic on your usb audio if the >automatic gain control is switch on Automatic gain control is now switched off. >Background noise affect the result if you are not using line in and line >out with a loopback cable I am using loopback, noise should not be an issue. >How many steps need to reduce amplitude of the recorded sine wave by >half when you play a sine wav through your usb audio and record it back >using audacity? Using recorded sine wav provided by 'speaker-test -t sin' amplitude of recorded sine wave reduced by half at step 77 output of 'amixer -c 2 set Headphone 77': Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 151 Mono: Front Left: Playback 77 [51%] [-13.94dB] [on] Front Right: Playback 77 [51%] [-13.94dB] [on] >You many need to change the name of the volume control to headphone >playback volume in alsamixertest Done. alsamixertest output: http://ix.io/ezd

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Re: [alsa-devel] [PATCH] ASoC:kirkwood: Don't raise an error when no DAI format
by Russell King - ARM Linux 01 Oct '14

01 Oct '14

On Sun, Sep 28, 2014 at 04:19:27PM +0200, Jean-Francois Moine wrote: > The two DAIs of the kirkwood controller have a unique PCM format. > > The simple-card sets the audio hardware definitions of all CPU DAIs. > The PCM format is defined only when it is present in the DT. > > This patch prevents the controller to raise an error when > the DT audio card definition by the simple card contains the PCM > format of one CPU DAI only. I think this is a silly idea - why should every driver have additional code to detect when it's called to do thing. Why doesn't the simple card code always pass the required format? Looking at other drivers, no one else does this; they all appear to require the proper format to be specified. What some drivers do (eg, omap-mcbsp.c) is to block set_fmt when the DAI is already in use - setting a flag "configured" in hw_params, and clearing it in shutdown. Maybe following this will solve your problem. In any case, random drivers doing stuff differently without reason is really not a good idea. -- FTTC broadband for 0.8mile line: currently at 9.5Mbps down 400kbps up according to speedtest.net.

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[alsa-devel] [GIT PULL] Another fix for v3.17
by Mark Brown 01 Oct '14

01 Oct '14

The following changes since commit 7d1311b93e58ed55f3a31cc8f94c4b8fe988a2b9: Linux 3.17-rc1 (2014-08-16 10:40:26 -0600) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.17-rc7 for you to fetch changes up to fe2a08b3bf1a6e35c00e18843bc19aa1778432c3: ASoC: ssm2602: do not hardcode type to SSM2602 (2014-09-30 13:27:19 +0100) ---------------------------------------------------------------- ASoC: Fix SSM2602 device identification Another fix for v3.17, fixing device identification after the split out of the I2C and SPI code. ---------------------------------------------------------------- Stefan Kristiansson (1): ASoC: ssm2602: do not hardcode type to SSM2602 sound/soc/codecs/ssm2602.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

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[alsa-devel] Radeon unconnected HDMI eats samples at 280 kHz
by David Henningsson 30 Sep '14

30 Sep '14

Hi, While investigating some interesting debug output from PulseAudio, I tried to figure out the cause. From what I can tell, my Radeon seems to ask for new samples at a very high rate, which I estimate to be around 280 kHz. My radeon card has DVI, HDMI and VGA connectors, and the only thing connected is my screen over DVI. I'm currently running the 3.13 kernel with updated hda directory from Takashi's tree, but I think it's been this way for a long time. Note that if a screen is connected to the HDMI card, the problem disappears and sample rates are normal. In short, do you think this is a driver bug, or just something we have to live with as some sort of hw anomaly? Since nothing is connected, it does not really hurt, except PulseAudio gets confused (in a way that could potentially cause problems for low-latency output, should something be connected later on). For reference, here's lspci of the HDMI card: 01:00.0 VGA compatible controller [0300]: Advanced Micro Devices, Inc. [AMD/ATI] RV710 [Radeon HD 4550] [1002:9540] (prog-if 00 [VGA controller]) Subsystem: PC Partner Limited / Sapphire Technology Device [174b:e106] 01:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] RV710/730 HDMI Audio [Radeon HD 4000 series] [1002:aa38] Subsystem: PC Partner Limited / Sapphire Technology Device [174b:aa38] -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH v3 1/2] ASoC: fsl_ssi: Remove unneeded 'i2s-slave' property
by Fabio Estevam 30 Sep '14

30 Sep '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> There is no need to use 'i2s-slave' property, since master/slave configuration are passed via machine layer. This change does not break existing users because they do check for slave mode inside sound/soc/fsl/mpc8610_hpcd.c/p1022_ds.c/p1022_rdk.c Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Acked-by: Timur Tabi <timur(a)tabi.org> --- Changes since v2: - Added Timur's Ack and CC the alsa-list Changes since v1: - None sound/soc/fsl/fsl_ssi.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2fc3e66..0e7ce50 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1285,9 +1285,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (sprop) { if (!strcmp(sprop, "ac97-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97; - else if (!strcmp(sprop, "i2s-slave")) - ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_CBM_CFM; } ssi_private->use_dma = !of_property_read_bool(np, -- 1.9.1

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[alsa-devel] [PATCH v2 1/3] ASoC: ssm2602: do not hardcode type to SSM2602
by Stefan Kristiansson 30 Sep '14

30 Sep '14

The correct type (SSM2602/SSM2603/SSM2604) is passed down from the ssm2602_spi_probe()/ssm2602_spi_probe() functions, so use that instead of hardcoding it to SSM2602 in ssm2602_probe(). Signed-off-by: Stefan Kristiansson <stefan.kristiansson(a)saunalahti.fi> --- Changes in v2: - None, new in series. --- sound/soc/codecs/ssm2602.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 484b3bb..4021cd4 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -647,7 +647,7 @@ int ssm2602_probe(struct device *dev, enum ssm2602_type type, return -ENOMEM; dev_set_drvdata(dev, ssm2602); - ssm2602->type = SSM2602; + ssm2602->type = type; ssm2602->regmap = regmap; return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602, -- 1.9.1

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[alsa-devel] [PATCH v5 1/6] ALSA: ctxfi: changed void * to struct hw *
by Sudip Mukherjee 30 Sep '14

30 Sep '14

in the code we have void *hw and while using we are always typecasting it to (struct hw *). it is better to use void type of pointer when we store different types of pointer , but in this code we are only having struct hw. So changed all the relevant reference of void *hw to struct hw *hw, without any modification of the existing code logic. the next patch of the series will remove the typecasting which is not required now. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/ctxfi/ctamixer.c | 4 ++-- sound/pci/ctxfi/ctamixer.h | 4 ++-- sound/pci/ctxfi/ctatc.c | 4 ++-- sound/pci/ctxfi/ctatc.h | 2 +- sound/pci/ctxfi/ctdaio.c | 4 ++-- sound/pci/ctxfi/ctdaio.h | 6 +++--- sound/pci/ctxfi/ctresource.c | 5 +++-- sound/pci/ctxfi/ctresource.h | 9 +++++---- sound/pci/ctxfi/ctsrc.c | 4 ++-- sound/pci/ctxfi/ctsrc.h | 4 ++-- 10 files changed, 24 insertions(+), 22 deletions(-) diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index fed6e6a..4671cbe7 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -296,7 +296,7 @@ static int put_amixer_rsc(struct amixer_mgr *mgr, struct amixer *amixer) return 0; } -int amixer_mgr_create(void *hw, struct amixer_mgr **ramixer_mgr) +int amixer_mgr_create(struct hw *hw, struct amixer_mgr **ramixer_mgr) { int err; struct amixer_mgr *amixer_mgr; @@ -449,7 +449,7 @@ static int put_sum_rsc(struct sum_mgr *mgr, struct sum *sum) return 0; } -int sum_mgr_create(void *hw, struct sum_mgr **rsum_mgr) +int sum_mgr_create(struct hw *hw, struct sum_mgr **rsum_mgr) { int err; struct sum_mgr *sum_mgr; diff --git a/sound/pci/ctxfi/ctamixer.h b/sound/pci/ctxfi/ctamixer.h index cc49e5a..6fa5eff 100644 --- a/sound/pci/ctxfi/ctamixer.h +++ b/sound/pci/ctxfi/ctamixer.h @@ -45,7 +45,7 @@ struct sum_mgr { }; /* Constructor and destructor of daio resource manager */ -int sum_mgr_create(void *hw, struct sum_mgr **rsum_mgr); +int sum_mgr_create(struct hw *hw, struct sum_mgr **rsum_mgr); int sum_mgr_destroy(struct sum_mgr *sum_mgr); /* Define the descriptor of a amixer resource */ @@ -90,7 +90,7 @@ struct amixer_mgr { }; /* Constructor and destructor of amixer resource manager */ -int amixer_mgr_create(void *hw, struct amixer_mgr **ramixer_mgr); +int amixer_mgr_create(struct hw *hw, struct amixer_mgr **ramixer_mgr); int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr); #endif /* CTAMIXER_H */ diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index d92a08c..04e54cc 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -106,11 +106,11 @@ static struct { .public_name = "Mixer"} }; -typedef int (*create_t)(void *, void **); +typedef int (*create_t)(struct hw *, void **); typedef int (*destroy_t)(void *); static struct { - int (*create)(void *hw, void **rmgr); + int (*create)(struct hw *hw, void **rmgr); int (*destroy)(void *mgr); } rsc_mgr_funcs[NUM_RSCTYP] = { [SRC] = { .create = (create_t)src_mgr_create, diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index 5f11ca2..5641334 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -131,7 +131,7 @@ struct ct_atc { /* Don't touch! Used for internal object. */ void *rsc_mgrs[NUM_RSCTYP]; /* chip resource managers */ void *mixer; /* internal mixer object */ - void *hw; /* chip specific hardware access object */ + struct hw *hw; /* chip specific hardware access object */ void **daios; /* digital audio io resources */ void **pcm; /* SUMs for collecting all pcm stream */ void **srcs; /* Sample Rate Converters for input signal */ diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 6f0654e..7541641 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -331,7 +331,7 @@ static struct dai_rsc_ops dai_ops = { static int daio_rsc_init(struct daio *daio, const struct daio_desc *desc, - void *hw) + struct hw *hw) { int err; unsigned int idx_l, idx_r; @@ -692,7 +692,7 @@ static int daio_mgr_commit_write(struct daio_mgr *mgr) return 0; } -int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr) +int daio_mgr_create(struct hw *hw, struct daio_mgr **rdaio_mgr) { int err, i; struct daio_mgr *daio_mgr; diff --git a/sound/pci/ctxfi/ctdaio.h b/sound/pci/ctxfi/ctdaio.h index 85ccb6e..e4817de 100644 --- a/sound/pci/ctxfi/ctdaio.h +++ b/sound/pci/ctxfi/ctdaio.h @@ -53,14 +53,14 @@ struct dao { struct dao_rsc_ops *ops; /* DAO specific operations */ struct imapper **imappers; struct daio_mgr *mgr; - void *hw; + struct hw *hw; void *ctrl_blk; }; struct dai { struct daio daio; struct dai_rsc_ops *ops; /* DAI specific operations */ - void *hw; + struct hw *hw; void *ctrl_blk; }; @@ -117,7 +117,7 @@ struct daio_mgr { }; /* Constructor and destructor of daio resource manager */ -int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr); +int daio_mgr_create(struct hw *hw, struct daio_mgr **rdaio_mgr); int daio_mgr_destroy(struct daio_mgr *daio_mgr); #endif /* CTDAIO_H */ diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index e49d2be..f14cbea 100644 --- a/sound/pci/ctxfi/ctresource.c +++ b/sound/pci/ctxfi/ctresource.c @@ -134,7 +134,8 @@ static struct rsc_ops rsc_generic_ops = { .next_conj = rsc_next_conj, }; -int rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, void *hw) +int +rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, struct hw *hw) { int err = 0; @@ -206,7 +207,7 @@ int rsc_uninit(struct rsc *rsc) } int rsc_mgr_init(struct rsc_mgr *mgr, enum RSCTYP type, - unsigned int amount, void *hw_obj) + unsigned int amount, struct hw *hw_obj) { int err = 0; struct hw *hw = hw_obj; diff --git a/sound/pci/ctxfi/ctresource.h b/sound/pci/ctxfi/ctresource.h index 0838c2e..9b746c3 100644 --- a/sound/pci/ctxfi/ctresource.h +++ b/sound/pci/ctxfi/ctresource.h @@ -38,7 +38,7 @@ struct rsc { u32 conj:12; /* Current conjugate index */ u32 msr:4; /* The Master Sample Rate a resource working on */ void *ctrl_blk; /* Chip specific control info block for a resource */ - void *hw; /* Chip specific object for hardware access means */ + struct hw *hw; /* Chip specific object for hardware access means */ struct rsc_ops *ops; /* Generic resource operations */ }; @@ -50,7 +50,8 @@ struct rsc_ops { int (*output_slot)(const struct rsc *rsc); }; -int rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, void *hw); +int +rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, struct hw *hw); int rsc_uninit(struct rsc *rsc); struct rsc_mgr { @@ -59,12 +60,12 @@ struct rsc_mgr { unsigned int avail; /* The amount of currently available resources */ unsigned char *rscs; /* The bit-map for resource allocation */ void *ctrl_blk; /* Chip specific control info block */ - void *hw; /* Chip specific object for hardware access */ + struct hw *hw; /* Chip specific object for hardware access */ }; /* Resource management is based on bit-map mechanism */ int rsc_mgr_init(struct rsc_mgr *mgr, enum RSCTYP type, - unsigned int amount, void *hw); + unsigned int amount, struct hw *hw); int rsc_mgr_uninit(struct rsc_mgr *mgr); int mgr_get_resource(struct rsc_mgr *mgr, unsigned int n, unsigned int *ridx); int mgr_put_resource(struct rsc_mgr *mgr, unsigned int n, unsigned int idx); diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index 19df9b4..342008f 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -543,7 +543,7 @@ static int src_mgr_commit_write(struct src_mgr *mgr) return 0; } -int src_mgr_create(void *hw, struct src_mgr **rsrc_mgr) +int src_mgr_create(struct hw *hw, struct src_mgr **rsrc_mgr) { int err, i; struct src_mgr *src_mgr; @@ -825,7 +825,7 @@ static int srcimp_imap_delete(struct srcimp_mgr *mgr, struct imapper *entry) return err; } -int srcimp_mgr_create(void *hw, struct srcimp_mgr **rsrcimp_mgr) +int srcimp_mgr_create(struct hw *hw, struct srcimp_mgr **rsrcimp_mgr) { int err; struct srcimp_mgr *srcimp_mgr; diff --git a/sound/pci/ctxfi/ctsrc.h b/sound/pci/ctxfi/ctsrc.h index 259366a..6d95afb 100644 --- a/sound/pci/ctxfi/ctsrc.h +++ b/sound/pci/ctxfi/ctsrc.h @@ -140,10 +140,10 @@ struct srcimp_mgr { }; /* Constructor and destructor of SRC resource manager */ -int src_mgr_create(void *hw, struct src_mgr **rsrc_mgr); +int src_mgr_create(struct hw *hw, struct src_mgr **rsrc_mgr); int src_mgr_destroy(struct src_mgr *src_mgr); /* Constructor and destructor of SRCIMP resource manager */ -int srcimp_mgr_create(void *hw, struct srcimp_mgr **rsrc_mgr); +int srcimp_mgr_create(struct hw *hw, struct srcimp_mgr **rsrc_mgr); int srcimp_mgr_destroy(struct srcimp_mgr *srcimp_mgr); #endif /* CTSRC_H */ -- 1.8.1.2

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[alsa-devel] [GIT PULL] ASoC fixes for v3.17
by Mark Brown 30 Sep '14

30 Sep '14

The following changes since commit 0f33be009b89d2268e94194dc4fd01a7851b6d51: Linux 3.17-rc6 (2014-09-21 15:43:02 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.17-rc6 for you to fetch changes up to 82b925c405444b760e743b55a9ad498cb3551afe: Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/compress', 'asoc/fix/core', 'asoc/fix/fsl-ssi' and 'asoc/fix/rt286' into asoc-linus (2014-09-28 12:25:12 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.17 A few small driver specific fixes and a couple of error handling fixes in the core. ---------------------------------------------------------------- Bard Liao (2): ASoC: rt286: Correct default value ASoC: rt286: Fix sync function Bo Shen (1): MAINTAINERS: add atmel audio alsa driver maintainer entry Mark Brown (1): Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/compress', 'asoc/fix/core', 'asoc/fix/fsl-ssi' and 'asoc/fix/rt286' into asoc-linus Michael Trimarchi (1): ASoC: fsl_ssi: fix kernel panic in probe function Qiao Zhou (1): ASoC: soc-compress: fix double unlock of fe card mutex Xiubo Li (1): ASoC: core: fix possible ZERO_SIZE_PTR pointer dereferencing error. MAINTAINERS | 6 ++++++ sound/soc/codecs/rt286.c | 7 +++---- sound/soc/fsl/fsl_ssi.c | 12 +++++++----- sound/soc/soc-compress.c | 6 ++---- sound/soc/soc-core.c | 2 +- 5 files changed, 19 insertions(+), 14 deletions(-)

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[alsa-devel] [BUG] Bad decibel data on C-Media USB sound card
by Allen M. 30 Sep '14

30 Sep '14

Raymond Yau wrote: >Do the automatic gain has any side effect when it normalise the volume? I am unsure of how to go about testing that, 'alsamixer' does not appear to allow me to modify automatic gain. >Do you mean the sound card does not has 7248 steps? I am not familiar with how to properly check, however dbverify indicates 'Element volume range is 0..151 (-28.37..-0.06 dB).' Instructions for dbmeasure on the wiki indicates the need for a Line In, the sound card only provides Line Out and Mic In. Will it automatically make use of 'Mic In' if 'Line In' is absent? Or do I even need to use it? >You need to find out the value of c->res >As you complain dB range -28.37 to -0.06 are not correct Quick testing done with dbverify: http://ix.io/eyi >You need to calibrate the dB range Volume Steps <=3 are mute while volume step 4 is audible, based on dbverify testing. This 'seems' to indicate that the minimum volume is somewhere between -27.81 dB to -27.63 dB? Additional info logs: Output of 'amixer -c2 contents': http://ix.io/eyq Alsa info script output: http://ix.io/eyn

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