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

[alsa-devel] Regression Intel Corporation Haswell-ULT HD Audio Controller
by Delcypher 05 May '14

05 May '14

Hi, I was told to report my bug here because the ALSA bug tracker is no longer active. I'm having an issue with " Intel Corporation Haswell-ULT HD Audio Controller" card which is inside the Dell XPS13 9333 developer edition [1]. I've tried two different distributions * Arch Linux (kernel 3.14) Output of alsa-utlils-alsa-info.sh [2] * Ubuntu 14.04 LTS (kernel 3.13) Output of alsa-utlils-alsa-info.sh [3] The card behaves correctly under Ubuntu but under Arch Linux I've observed the following issues. - The microphone does not work correctly. In alsamixer I select "Internal Mic" as my capture device but when I do $ arecord -d 5 test.wav $ aplay test.wav test.wav contains silence. I've also tried enabling/disabling "CAPTURE" on the Capture channel at the same time - I don't really understand why this channel is here (seems a little bizarre) but enabling/disabling "CAPTURE" on this channel doesn't seem to make any difference anyway. Under Ubuntu the "arecord... aplay" command works correctly. - There probably isn't a way to reproduce this but when I first installed Arch Linux the sound card was put in a state where the headphone jack would not work at all. Sound would continue playing out of the speakers when the headphone jack was inserted and no sound would come out of the headphones. I did notice that if the headphone jack was only partially inserted I would get some sound sent to one of the headphones. My intuition is that the card is probably being incorrectly initialised because if you take a look at the output of alsa-utils-alsa-info.sh and diff them. Two very big differences I noticed were that - The card name (from lscpi) is "Intel Corporation Haswell-ULT HD Audio Controller" under Ubuntu but under Arch Linux it is "Intel Corporation Device 0a0c" - The card codec is "Realtek ALC668" under Ubuntu but is "Realtek ALC3661" under Arch Linux. One interesting I did notice is that under Arch Linux it is possible to use the microphone from the Audacity program by selecting "HDA Intel PCH: ALC3661Analog (hw0:0) Internal mic:0" as the input (this is not the default). Unfortunately I know of no way to make other applications use this. Any thoughts on how to fix this? [1] http://www.dell.com/uk/business/p/xps-13-linux/pd [2] https://gist.github.com/delcypher/11367224 [3] https://gist.github.com/delcypher/11367302 Thanks, Dan.

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[alsa-devel] [PATCH 0/6] ASoC: Intel: rework the mfld platform driver
by Vinod Koul 05 May '14

05 May '14

This patch series reworks and reformats the existing platform driver we use. This is stepping support for seconds series of patches which will support Merrifield platform with DPCM based implementation Not much logical changes in code but modularize the existing code Vinod Koul (6): ASoC: Intel: remove FSF snail mail address ASoc: Intel: rename sst-mfld-platform.c ASoC: Intel: mark sst_set_stream_status as non static ASoC: Intel: split the pcm and compress to different files ASoC: Intel: remove unused sst-mfld platform dais ASoC: Intel: rename pcm dias to media dai sound/soc/intel/Makefile | 2 +- sound/soc/intel/sst-mfld-dsp.h | 4 - sound/soc/intel/sst-mfld-platform-compress.c | 228 ++++++++++++++++ ...sst-mfld-platform.c => sst-mfld-platform-pcm.c} | 278 ++------------------ sound/soc/intel/sst-mfld-platform.h | 9 +- 5 files changed, 254 insertions(+), 267 deletions(-) create mode 100644 sound/soc/intel/sst-mfld-platform-compress.c rename sound/soc/intel/{sst-mfld-platform.c => sst-mfld-platform-pcm.c} (62%)

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[alsa-devel] ALC1150 can't playback SND_PCM_FORMAT_FLOAT without dmix
by Madura Anushanga 04 May '14

04 May '14

Hi, MSI z87-GD65 GAMING motherboard's ALC1150 plays SND_PCM_FORMAT_FLOAT with noise[1] other non float formats work well. As soon as I changed /etc/asound.conf from pcm.!default{ type hw card 1 } to pcm.dmixed { type asym playback.pcm { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio slave { channels 2 # make 6 or 5.1 channel pcm { format S16_LE # S32_LE rate 48000 # can also be 44100 type hw card 1 # your card device 0 # your device subdevice 0 #important? } period_size 1024 buffer_size 8192 } bindings { 0 0 1 1 # Uncomment below if using 6 channel # 2 2 # 3 3 # 4 4 # 5 5 } } capture.pcm "hw:0" } pcm.!default { type plug slave.pcm "dmixed" } SND_PCM_FORMAT_FLOAT plays without a problem. 1. https://github.com/manushanga/vrok/blob/master/outs/alsa.cpp - Madura A.

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Re: [alsa-devel] [PATCH] ASoC: mc13783: Add devicetree support
by Lee Jones 03 May '14

03 May '14

> > > > This patch adds devicetree support for mc13783-codec. > > > > > > Applied, thanks. > > > > No Mark, no! > > What a problem, Lee? There's no problem with the patch, don't worry. Neither Mark, nor I can apply this patch independently, we need to create a shared branch somewhere, or we'll risk conflicts at merge time. -- Lee Jones Linaro STMicroelectronics Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

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[alsa-devel] [PATCH 13/15] ASoC: SND_S3C_DMA_LEGACY needs S3C24XX_DMA
by Xia Kaixu 02 May '14

02 May '14

From: Arnd Bergmann <arnd(a)arndb.de> SND_S3C_DMA_LEGACY can only be set on S3C24xx, which does not (yet) support the dmaengine framework, so samsung_dma_get_ops() fails to link if S3C24XX_DMA is disabled: sound/built-in.o: In function `dma_hw_params': :(.text+0x7f310): undefined reference to `s3c_dma_get_ops' sound/built-in.o: In function `s3c_ac97_trigger': :(.text+0x7f7f0): undefined reference to `s3c_dma_get_ops' sound/built-in.o: In function `s3c_ac97_mic_trigger': :(.text+0x7f884): undefined reference to `s3c_dma_get_ops' sound/built-in.o: In function `s3c2412_i2s_trigger': :(.text+0x80944): undefined reference to `s3c2410_dma_ctrl' This makes sure S3C24XX_DMA is always enabled when we need it, just like we do it for the same dependency in SND_S3C24XX_I2S, which has the same problem. Selecting "S3C2410_DMA" as we did before does not actually have the intended effect, since that one is only used on the s3c2410 and s3c2442 SoCs of the s3c24xx family, but not the others, so we can remove this from Kconfig. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Kukjin Kim <kgene.kim(a)samsung.com> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- sound/soc/samsung/Kconfig | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 99cc196..7b610a8 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,7 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C2410_DMA if ARCH_S3C24XX select S3C64XX_PL080 if ARCH_S3C64XX select SND_S3C_DMA if !ARCH_S3C24XX select SND_S3C_DMA_LEGACY if ARCH_S3C24XX @@ -15,11 +14,11 @@ config SND_S3C_DMA tristate config SND_S3C_DMA_LEGACY + select S3C24XX_DMA tristate config SND_S3C24XX_I2S tristate - select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate @@ -27,7 +26,6 @@ config SND_S3C_I2SV2_SOC config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC - select S3C2410_DMA config SND_SAMSUNG_PCM tristate @@ -83,7 +81,6 @@ config SND_SOC_SAMSUNG_SMDK_WM8994 config SND_SOC_SAMSUNG_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_SOC_SAMSUNG && MACH_SMDK2443 - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 @@ -94,7 +91,6 @@ config SND_SOC_SAMSUNG_SMDK2443_WM9710 config SND_SOC_SAMSUNG_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" depends on SND_SOC_SAMSUNG && ARCH_S3C24XX - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: add SND_SOC_BYTES_EXT
by Vinod Koul 02 May '14

02 May '14

we need _EXT version for SND_SOC_BYTES so that DSPs can use this to pass data for DSP modules Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc.h | 13 +++++++++++++ sound/soc/soc-core.c | 13 +++++++++++++ 2 files changed, 26 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..63202cf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -266,6 +266,13 @@ {.base = xbase, .num_regs = xregs, \ .mask = xmask }) } +#define SND_SOC_BYTES_EXT(xname, xcount, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_bytes_info_ext, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_bytes_ext) \ + {.max = xcount} } + #define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \ xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -540,6 +547,8 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol); int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, @@ -1090,6 +1099,10 @@ struct soc_bytes { u32 mask; }; +struct soc_bytes_ext { + int max; +}; + /* multi register control */ struct soc_mreg_control { long min, max; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 051c006..48e4bba 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3297,6 +3297,19 @@ out: } EXPORT_SYMBOL_GPL(snd_soc_bytes_put); +int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + + ucontrol->type = SNDRV_CTL_ELEM_TYPE_BYTES; + ucontrol->count = params->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); + /** * snd_soc_info_xr_sx - signed multi register info callback * @kcontrol: mreg control -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: Intel: Fix block allocation so we only allocate blocks once.
by Liam Girdwood 02 May '14

02 May '14

Make sure we dont alloc blocks twice with requests spanning more than one block. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-firmware.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index f768710..c4e7126 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -376,10 +376,6 @@ static int block_alloc_fixed(struct sst_module *module, if (err < 0) return -ENOMEM; - /* add block */ - block->data_type = data->data_type; - list_move(&block->list, &dsp->used_block_list); - list_add(&block->module_list, &module->block_list); return 0; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Remove IS_ENABLED for INPUT in CS42L52 and WM8962
by Brian Austin 02 May '14

02 May '14

Now that INPUT is required for the CS42L52 and WM8962 we can remove the IS_ENABLED(INPUT) check in the drivers. Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 12 ------------ sound/soc/codecs/wm8962.c | 12 ------------ 2 files changed, 24 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 8a7bde4..c0581c5 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -50,11 +50,9 @@ struct cs42l52_private { u8 mclksel; u32 mclk; u8 flags; -#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; -#endif }; static const struct reg_default cs42l52_reg_defaults[] = { @@ -962,7 +960,6 @@ static int cs42l52_resume(struct snd_soc_codec *codec) return 0; } -#if IS_ENABLED(CONFIG_INPUT) static int beep_rates[] = { 261, 522, 585, 667, 706, 774, 889, 1000, 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 @@ -1096,15 +1093,6 @@ static void cs42l52_free_beep(struct snd_soc_codec *codec) snd_soc_update_bits(codec, CS42L52_BEEP_TONE_CTL, CS42L52_BEEP_EN_MASK, 0); } -#else -static void cs42l52_init_beep(struct snd_soc_codec *codec) -{ -} - -static void cs42l52_free_beep(struct snd_soc_codec *codec) -{ -} -#endif static int cs42l52_probe(struct snd_soc_codec *codec) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 37986c8..b77e4c3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -74,11 +74,9 @@ struct wm8962_priv { struct regulator_bulk_data supplies[WM8962_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8962_NUM_SUPPLIES]; -#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; -#endif #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; @@ -3145,7 +3143,6 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(wm8962_mic_detect); -#if IS_ENABLED(CONFIG_INPUT) static int beep_rates[] = { 500, 1000, 2000, 4000, }; @@ -3277,15 +3274,6 @@ static void wm8962_free_beep(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8962_BEEP_GENERATOR_1, WM8962_BEEP_ENA,0); } -#else -static void wm8962_init_beep(struct snd_soc_codec *codec) -{ -} - -static void wm8962_free_beep(struct snd_soc_codec *codec) -{ -} -#endif static void wm8962_set_gpio_mode(struct wm8962_priv *wm8962, int gpio) { -- 1.7.9.5

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[alsa-devel] [PATCH v2] ALSA: usb-audio: work around corrupted TEAC UD-H01 feedback data
by Daniel Mack 02 May '14

02 May '14

From: Clemens Ladisch <clemens(a)ladisch.de> The TEAC UD-H01 firmware sends wrong feedback frequency values, thus causing the PC to send the samples at a wrong rate, which results in clicks and crackles in the output. Add a workaround to detect and fix the corruption. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> [mick37(a)gmx.de: use sender->udh01_fb_quirk rather than ep->udh01_fb_quirk in snd_usb_handle_sync_urb()] Reported-and-tested-by: Mick <mick37(a)gmx.de> Reported-and-tested-by: Andrea Messa <andr.messa(a)tiscali.it> Cc: <stable(a)vger.kernel.org> --- Clemens, I hope you're ok that I resend this patch. With Mick's minor amendment (use sender->udh01_fb_quirk rather than ep->udh01_fb_quirk), Andrea Messa also reported success with the same hardware model. Thanks, Daniel sound/usb/card.h | 1 + sound/usb/endpoint.c | 15 ++++++++++++++- 2 files changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 9867ab8..97acb90 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -92,6 +92,7 @@ struct snd_usb_endpoint { unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int syncmaxsize; /* sync endpoint packet size */ unsigned int fill_max:1; /* fill max packet size always */ + unsigned int udh01_fb_quirk:1; /* corrupted feedback data */ unsigned int datainterval; /* log_2 of data packet interval */ unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ unsigned char silence_value; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e70a87e..289f582 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -471,6 +471,10 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, ep->syncinterval = 3; ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); + + if (chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ && + ep->syncmaxsize == 4) + ep->udh01_fb_quirk = 1; } list_add_tail(&ep->list, &chip->ep_list); @@ -1105,7 +1109,16 @@ void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, if (f == 0) return; - if (unlikely(ep->freqshift == INT_MIN)) { + if (unlikely(sender->udh01_fb_quirk)) { + /* + * The TEAC UD-H01 firmware sometimes changes the feedback value + * by +/- 0x1.0000. + */ + if (f < ep->freqn - 0x8000) + f += 0x10000; + else if (f > ep->freqn + 0x8000) + f -= 0x10000; + } else if (unlikely(ep->freqshift == INT_MIN)) { /* * The first time we see a feedback value, determine its format * by shifting it left or right until it matches the nominal -- 1.9.0

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Re: [alsa-devel] [PATCH] ASoC: fsl: select SND_SOC_IMX_PCM_DMA where needed
by Arnd Bergmann 01 May '14

01 May '14

On Tuesday 29 April 2014 07:07:33 Alexander Shiyan wrote: > Mon, 28 Apr 2014 23:12:14 +0200 от Arnd Bergmann <arnd(a)arndb.de>: > > On Tuesday 29 April 2014 00:35:41 Alexander Shiyan wrote: > > > > > > So, I don't understand why this error happen, as well as I can not > > > reproduce this... > > > > It's probably CONFIG_SND_SOC_IMX_PCM_DMA=m and CONFIG_SND_SOC_FSL_SSI=y then. > > What is the intended behavior in this case? Should CONFIG_SND_SOC_FSL_SSI > > be forced to be a module as well? > > Hmm, yes... > I thought that I had already solved a similar problem for the earlier > version of the patch ... How about this? Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index fb26345..af0bb92 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -9,6 +9,7 @@ config SND_SOC_FSL_SAI config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module support" + depends on m || SND_SOC_IMX_PCM_DMA != m help Say Y if you want to add Synchronous Serial Interface (SSI) support for the Freescale CPUs. @@ -18,6 +19,7 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate "Sony/Philips Digital Interface module support" select REGMAP_MMIO + depends on m || SND_SOC_IMX_PCM_DMA != m help Say Y if you want to add Sony/Philips Digital Interface (SPDIF) support for the Freescale CPUs.

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