September 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: nuc900: Fix a loop timeout test
by Dan Carpenter 11 Dec '17

11 Dec '17

We should be finishing the loop with timeout set to zero but because this is a post-op we exit with timeout == -1. Fixes: 1082e2703a2d ("ASoC: NUC900/audio: add nuc900 audio driver support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index b6615affe571..fde974d52bb2 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -67,7 +67,7 @@ static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97, /* polling the AC_R_FINISH */ while (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_R_FINISH) - && timeout--) + && --timeout) mdelay(1); if (!timeout) { @@ -121,7 +121,7 @@ static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg, /* polling the AC_W_FINISH */ while ((AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_W_FINISH) - && timeout--) + && --timeout) mdelay(1); if (!timeout)

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[alsa-devel] [PATCH] ASoC: fsl_asrc: protect macro argument
by Stefan Agner 06 Dec '17

06 Dec '17

Protect macro argument with parentheses to avoid ambiguity. This fixes a warning seen with clang: warning: logical not is only applied to the left hand side of this comparison Signed-off-by: Stefan Agner <stefan(a)agner.ch> --- sound/soc/fsl/fsl_asrc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 0f163abe4ba3..ec33dab4b909 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -57,7 +57,7 @@ #define REG_ASRDOC 0x74 #define REG_ASRDI(i) (REG_ASRDIA + (i << 3)) #define REG_ASRDO(i) (REG_ASRDOA + (i << 3)) -#define REG_ASRDx(x, i) (x == IN ? REG_ASRDI(i) : REG_ASRDO(i)) +#define REG_ASRDx(x, i) ((x) == IN ? REG_ASRDI(i) : REG_ASRDO(i)) #define REG_ASRIDRHA 0x80 #define REG_ASRIDRLA 0x84 -- 2.13.3

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[alsa-devel] [PATCH] ASoC: fsl_ssi: Override bit clock rate based on slot number
by Nicolin Chen 26 Nov '17

26 Nov '17

The set_sysclk() now is used to override the output bit clock rate. But this is not a common way to implement a set_dai_sysclk(). And this creates a problem when a general machine driver (simple-card for example) tries to do set_dai_sysclk() by passing an input clock rate for the baud clock instead of setting the bit clock rate as fsl_ssi driver expected. So this patch solves this problem by firstly removing set_sysclk() since the hw_params() can calculate the bit clock rate. Secondly, in order not to break those TDM use cases which previously might have been using set_sysclk() to override the bit clock rate, this patch changes the driver to override it based on the slot number. The patch also removes an obsolete comment of the dir parameter. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 26 ++++++++------------------ 1 file changed, 8 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 64598d1..3657c88 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -197,12 +197,12 @@ struct fsl_ssi_soc_data { * @use_dma: DMA is used or FIQ with stream filter * @use_dual_fifo: DMA with support for both FIFOs used * @fifo_deph: Depth of the SSI FIFOs + * @slots: number of slots * @rxtx_reg_val: Specific register settings for receive/transmit configuration * * @clk: SSI clock * @baudclk: SSI baud clock for master mode * @baudclk_streams: Active streams that are using baudclk - * @bitclk_freq: bitclock frequency set by .set_dai_sysclk * * @dma_params_tx: DMA transmit parameters * @dma_params_rx: DMA receive parameters @@ -233,12 +233,12 @@ struct fsl_ssi_private { bool use_dual_fifo; bool has_ipg_clk_name; unsigned int fifo_depth; + unsigned int slots; struct fsl_ssi_rxtx_reg_val rxtx_reg_val; struct clk *clk; struct clk *baudclk; unsigned int baudclk_streams; - unsigned int bitclk_freq; /* regcache for volatile regs */ u32 regcache_sfcsr; @@ -700,8 +700,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, * Note: This function can be only called when using SSI as DAI master * * Quick instruction for parameters: - * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels - * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + * freq: Output BCLK frequency = samplerate * 32 (fixed) * slots (or channels) */ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai, @@ -716,9 +715,9 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, unsigned int freq; bool baudclk_is_used; - /* Prefer the explicitly set bitclock frequency */ - if (ssi_private->bitclk_freq) - freq = ssi_private->bitclk_freq; + /* Generate bit clock based on the slot or channel number */ + if (ssi_private->slots) + freq = ssi_private->slots * 32 * params_rate(hw_params); else freq = params_channels(hw_params) * 32 * params_rate(hw_params); @@ -805,16 +804,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, return 0; } -static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); - - ssi_private->bitclk_freq = freq; - - return 0; -} - /** * fsl_ssi_hw_params - program the sample size * @@ -1121,6 +1110,8 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN, val); + ssi_private->slots = slots; + return 0; } @@ -1191,7 +1182,6 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, .set_fmt = fsl_ssi_set_dai_fmt, - .set_sysclk = fsl_ssi_set_dai_sysclk, .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; -- 2.7.4

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[alsa-devel] LADSPA ALSA plugin
by Robert Bielik 26 Nov '17

26 Nov '17

Hi all, Started experimenting with using LADSPA plugins with the ALSA LADSPA PCM plugin, and it works nicely. However, it seems only the PCM is interface is exposed from the plugin. Question is if there is a way to expose the LADSPA plugin parameters directly to alsamixer ? Regards /Rob

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[alsa-devel] [PATCH] ALSA: line6: add support for POD HD DESKTOP
by Hans P. Möller Ebner 20 Nov '17

20 Nov '17

Add support for the Line6 POD HD500X multi effect processor for playback and capture (in/out audio) through USB. Signed-off-by: Hans P. Moller <hmoller(a)uc.cl> --- sound/usb/line6/podhd.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 956f847..ba4f71f 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -39,7 +39,8 @@ enum { LINE6_PODHD500_1, LINE6_PODX3, LINE6_PODX3LIVE, - LINE6_PODHD500X + LINE6_PODHD500X, + LINE6_PODHDDESKTOP }; struct usb_line6_podhd { @@ -377,6 +378,7 @@ static const struct usb_device_id podhd_id_table[] = { { LINE6_IF_NUM(0x414A, 0), .driver_info = LINE6_PODX3 }, { LINE6_IF_NUM(0x414B, 0), .driver_info = LINE6_PODX3LIVE }, { LINE6_IF_NUM(0x4159, 0), .driver_info = LINE6_PODHD500X }, + { LINE6_IF_NUM(0x4156, 0), .driver_info = LINE6_PODHDDESKTOP }, {} }; @@ -463,6 +465,18 @@ static const struct line6_properties podhd_properties_table[] = { .ep_audio_r = 0x86, .ep_audio_w = 0x02, }, + [LINE6_PODHDDESKTOP] = { + .id = "PODHDDESKTOP", + .name = "POD HDDESKTOP", + .capabilities = LINE6_CAP_CONTROL + | LINE6_CAP_PCM | LINE6_CAP_HWMON, + .altsetting = 1, + .ep_ctrl_r = 0x81, + .ep_ctrl_w = 0x01, + .ctrl_if = 1, + .ep_audio_r = 0x86, + .ep_audio_w = 0x02, + }, }; /* -- 2.7.4

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[alsa-devel] [PATCH] ALSA: via82xx: Use common error handling code in snd_via82xx_create()
by SF Markus Elfring 19 Nov '17

19 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 19:37:26 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/via82xx_modem.c | 33 +++++++++++++++++++-------------- 1 file changed, 19 insertions(+), 14 deletions(-) diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 55f79b2599e7..8659776887b2 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1110,8 +1110,8 @@ static int snd_via82xx_create(struct snd_card *card, return err; if ((chip = kzalloc(sizeof(*chip), GFP_KERNEL)) == NULL) { - pci_disable_device(pci); - return -ENOMEM; + err = -ENOMEM; + goto disable_device; } spin_lock_init(&chip->reg_lock); @@ -1121,30 +1121,27 @@ static int snd_via82xx_create(struct snd_card *card, if ((err = pci_request_regions(pci, card->driver)) < 0) { kfree(chip); - pci_disable_device(pci); - return err; + goto disable_device; } chip->port = pci_resource_start(pci, 0); if (request_irq(pci->irq, snd_via82xx_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); - snd_via82xx_free(chip); - return -EBUSY; + err = -EBUSY; + goto free_chip; } chip->irq = pci->irq; if (ac97_clock >= 8000 && ac97_clock <= 48000) chip->ac97_clock = ac97_clock; synchronize_irq(chip->irq); - if ((err = snd_via82xx_chip_init(chip)) < 0) { - snd_via82xx_free(chip); - return err; - } + err = snd_via82xx_chip_init(chip); + if (err < 0) + goto free_chip; - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_via82xx_free(chip); - return err; - } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto free_chip; /* The 8233 ac97 controller does not implement the master bit * in the pci command register. IMHO this is a violation of the PCI spec. @@ -1153,6 +1150,14 @@ static int snd_via82xx_create(struct snd_card *card, *r_via = chip; return 0; + +disable_device: + pci_disable_device(pci); + goto exit; +free_chip: + snd_via82xx_free(chip); +exit: + return err; } -- 2.14.0

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[alsa-devel] [PATCH 0/7] ALSA: Fine-tuning for several function implementations
by SF Markus Elfring 14 Nov '17

14 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 24 Jan 2017 11:02:34 +0100 Some update suggestions were taken into account from static source code analysis. Markus Elfring (7): seq: Delete unnecessary checks in snd_seq_ioctl_query_subs() seq: Delete unnecessary checks in snd_seq_ioctl_get_subscription() seq: Delete unnecessary checks in snd_seq_ioctl_unsubscribe_port() seq: Delete unnecessary checks in snd_seq_ioctl_subscribe_port() seq: Adjust 16 function calls together with a variable assignment dmasound_core: Move two assignments for the variable "ret" in state_open() dmasound_core: Adjust six function calls together with a variable assignment sound/core/seq/seq_clientmgr.c | 163 ++++++++++++++++++++----------------- sound/oss/dmasound/dmasound_core.c | 33 +++++--- 2 files changed, 109 insertions(+), 87 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH 1/1] ASoC: bcm2835-i2s: Support 384k sample rate
by Clive Messer 10 Nov '17

10 Nov '17

Add support for sample rates up to 384k to bcm2835-i2s driver, with SNDRV_PCM_RATE_CONTINUOUS and specifying 8k min and 384k max, for both playback and capture. Signed-off-by: Clive Messer <clive.messer(a)digitaldreamtime.co.uk> --- sound/soc/bcm/bcm2835-i2s.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 6ba2049..4a67db2 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -578,7 +578,9 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE @@ -586,7 +588,9 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE -- 2.5.5

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[alsa-devel] [PATCH RESEND1 00/12] ALSA: vsnd: Add Xen para-virtualized frontend driver
by Oleksandr Andrushchenko 02 Nov '17

02 Nov '17

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> This patch series adds support for Xen [1] para-virtualized sound frontend driver. It implements the protocol from include/xen/interface/io/sndif.h with the following limitations: - mute/unmute is not supported - get/set volume is not supported Volume control is not supported for the reason that most of the use-cases (at the moment) are based on scenarious where unprivileged OS (e.g. Android, AGL etc) use software mixers. Both capture and playback are supported. Thank you, Oleksandr Resending because of rebase onto [2] + added missing patch [1] https://xenproject.org/ [2] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/log/?h=for-… Oleksandr Andrushchenko (12): ALSA: vsnd: Introduce Xen para-virtualized sound frontend driver ALSA: vsnd: Implement driver's probe/remove ALSA: vsnd: Implement Xen bus state handling ALSA: vsnd: Read sound driver configuration from Xen store ALSA: vsnd: Implement Xen event channel handling ALSA: vsnd: Implement handling of shared buffers ALSA: vsnd: Introduce ALSA virtual sound driver ALSA: vsnd: Initialize virtul sound card ALSA: vsnd: Add timer for period interrupt emulation ALSA: vsnd: Implement ALSA PCM operations ALSA: vsnd: Implement communication with backend ALSA: vsnd: Introduce Kconfig option to enable Xen PV sound sound/drivers/Kconfig | 12 + sound/drivers/Makefile | 2 + sound/drivers/xen-front.c | 2107 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 2121 insertions(+) create mode 100644 sound/drivers/xen-front.c -- 2.7.4

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[alsa-devel] [PATCH 0/2] Enable TDM slot and fix SSP params for kabylake machine driver
by Harsha Priya 31 Oct '17

31 Oct '17

This patch series includes 2 patches for kabylake machine driver using max98927 and rt5663 codecs. The changes are: 1. Enable 4 tdm slots for IV feedback and 2 tdm slots for playback on max98927 2. Fix setting of SSP parameters to be based on FE and BE dai links. Naveen M (2): ASoC: Intel: Enable tdm slots for max98927 ASoC: Intel: Fix setting of SSP parameters in Kabylake machine driver sound/soc/intel/boards/kbl_rt5663_max98927.c | 55 ++++++++++++++++++++++++++-- 1 file changed, 52 insertions(+), 3 deletions(-) -- 1.9.1

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