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

[alsa-devel] TLV320AIC3106 + DM3730 soc doens't capture audio (no BLCK, no WLCK)
by Sergio Serrano 17 Nov '14

17 Nov '14

Dear members, Please, let me explain my setup: I'm using DM3730 soc <https://www.isee.biz/products/igep-processor-boards/igepv2-dm3730> with McBSP3 and I2C2 to drive audio and control to the TLV320AIC3106 codec (which it is acting as master). Currently I'm using 2.6.37 kernel version <http://git.isee.biz/?p=pub/scm/linux-omap-2.6.git;a=tree>. For reference I've used BBB Audio Cape <http://elinux.org/CircuitCo:Audio_Cape_RevA> as base design. Right now I've got playback working as expected (codec generates BCLK and WCLK). Whenever I try to record using "arecor,d -f cd -Dhw:1,0" it's only showing wave header but no audio is captured at all. I can not see any clocks running on the codec. How can I debug or go further with this issue? Any thoughts will be appreciated!! BTW this is the init code: static int TLV320AIC3106_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; printk(KERN_ERR "+tlv_hw_params(%p,%p) %p, %p, %p\n", substream, params, rtd, codec_dai, cpu_dai); // Set codec DAI configuration ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) { printk(KERN_ERR "tlv_hw_params: can't set codec DAI configuration\n"); return ret; } // Set cpu DAI configuration ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) { printk(KERN_ERR "tlv_hw_params: can't set cpu DAI configuration\n"); return ret; } // Set the codec system clock for DAC and ADC ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "tlv_hw_params: can't set codec system clock\n"); return ret; } printk(KERN_ERR "-tlv_hw_params\n"); return 0; } BR, Sergio

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[alsa-devel] Pop and offsets at start of audio playback/record for SGTL5000 on i.MX28
by Craig McQueen 17 Nov '14

17 Nov '14

I'm testing the 3.14.19 kernel for i.MX28 EVK, which has an SGTL5000 CODEC. I've also tested on the 3.18-rc4 kernel and confirmed this issue still occurs. Playback When doing audio playback, I notice that the audio "fades in" over the first (approximately) 500 ms of playback. This is very noticeable and not ideal in certain applications (e.g. audio notifications that are of short duration). However, if audio playback starts while audio recording is already in-progress, then there is no fade-in; instead there is a loud pop. I assume the fade-in is there to mask the pop. The pop is very unpleasant through headphones. (However, if I do audio playback a _second_ time while audio recording is still in-progress, then there is no loud pop the second time.) I didn't think either the pop or the fade-in occurred on the 2.6.35 kernel. How feasible is it to improve the driver to avoid or reduce the pop, and remove the necessity for a fade-in? Record When recording, I notice there is a low-frequency ramping offset at the start of a recording (after the 400-500 ms of previous audio) of approx 600-700 ms duration. This also reduces the audio quality at the start of the recording. It could reduce the quality of captured audio depending on the application usage. I am getting distortion (clipping I assume) of my audio input during this time. It could cause a pop on playback (local playback or e.g. VoIP transmission to a remote device). It could reduce the audio quality through an audio CODEC. How feasible is it to improve the driver to avoid a ramping offset at the start of recording? -- Craig McQueen

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[alsa-devel] What tree to base patches on?
by Johan Aires Rastén 16 Nov '14

16 Nov '14

If I want to send you a patch for snd_usb_audio, is there a git repo + branch or something else which I should base it on?

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[alsa-devel] [PATCH v2] ASoC: Fix kerneldoc errors
by Takashi Iwai 16 Nov '14

16 Nov '14

Lots of typos and missing parameter descriptions. Some private struct fields are moved to the end of struct and put markers to make kerneldoc happy (and it make even easier for human readers). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc.h | 16 +++++++++++----- sound/soc/soc-core.c | 21 ++++++++++++++------- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-devres.c | 2 +- sound/soc/soc-jack.c | 2 +- 5 files changed, 28 insertions(+), 15 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130037a0..10129be3cbb1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -588,10 +588,11 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, * @invert: if non-zero then pin is enabled when status is not reported */ struct snd_soc_jack_pin { - struct list_head list; const char *pin; int mask; bool invert; + /* private: */ + struct list_head list; }; /** @@ -602,13 +603,13 @@ struct snd_soc_jack_pin { * @jack_type: type of jack that is expected for this voltage * @debounce_time: debounce_time for jack, codec driver should wait for this * duration before reading the adc for voltages - * @:list: list container */ struct snd_soc_jack_zone { unsigned int min_mv; unsigned int max_mv; unsigned int jack_type; unsigned int debounce_time; + /* private: */ struct list_head list; }; @@ -618,16 +619,19 @@ struct snd_soc_jack_zone { * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device - * @gpiod_dev GPIO consumer device + * @gpiod_dev: GPIO consumer device * @name: gpio name. Also as connection ID for the GPIO consumer * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state - * @debouce_time: debouce time in ms + * @debounce_time: debouce time in ms * @wake: enable as wake source + * @jack: assigned jack + * @desc: GPIO descriptor * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an * ADC). + * @data: arbitrary data passed to jack_status_check callback */ struct snd_soc_jack_gpio { unsigned int gpio; @@ -640,11 +644,13 @@ struct snd_soc_jack_gpio { bool wake; struct snd_soc_jack *jack; - struct delayed_work work; struct gpio_desc *desc; void *data; int (*jack_status_check)(void *data); + + /* private: */ + struct delayed_work work; }; struct snd_soc_jack { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a23a0e9..74a8a80e2d9a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2113,6 +2113,8 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); /** * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions + * @ops: AC97 opts to set + * @pdev: platform to deal with * * This function sets the reset and warm_reset properties of ops and parses * the device node of pdev to get pinctrl states and gpio numbers to use. @@ -2685,7 +2687,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * @@ -3458,7 +3460,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ @@ -4082,6 +4084,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core + * @dev: device to unregister * */ void snd_soc_unregister_component(struct device *dev) @@ -4119,7 +4122,7 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform * @platform: The platform to add - * @platform_driver: The driver for the platform + * @platform_drv: The driver for the platform */ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, const struct snd_soc_platform_driver *platform_drv) @@ -4158,7 +4161,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: platform to register + * @platform_drv: the driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -4215,7 +4219,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -4306,7 +4310,10 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: codec to register + * @codec_drv: the codec driver to register + * @dai_drv: the array of DAI drivers to register + * @num_dai: the number of DAIs to register */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -4406,7 +4413,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9cedbcd..7050ec1d5b72 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2680,7 +2680,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card object containing widgets * * Checks the codec for any new dapm widgets and creates them if found. * diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 057e5ef7dcce..a57921eeee81 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -60,7 +60,7 @@ static void devm_platform_release(struct device *dev, void *res) /** * devm_snd_soc_register_platform - resource managed platform registration * @dev: Device used to manage platform - * @platform: platform to register + * @platform_drv: platform to register * * Register a platform driver with automatic unregistration when the device is * unregistered. diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab47fea997a3..ef1d42d7c6f6 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -116,7 +116,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_report); * * @jack: ASoC jack * @count: Number of zones - * @zone: Array of zones + * @zones: Array of zones * * After this function has been called the zones specified in the * array will be associated with the jack. -- 2.1.3

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[alsa-devel] [PATCH] usb-audio: Add ctrl message delay quirk for Marantz/Denon devices
by Jurgen Kramer 16 Nov '14

16 Nov '14

This patch adds a USB control message delay quirk for a few specific Marantz/Denon devices. Without the delay the DACs will not work properly and produces the following type of messages: Nov 15 10:09:21 orwell kernel: [ 91.342880] usb 3-13: clock source 41 is not valid, cannot use Nov 15 10:09:21 orwell kernel: [ 91.343775] usb 3-13: clock source 41 is not valid, cannot use There are likely other Marantz/Denon devices using the same USB module which exhibit the same problems. But as this cannot be verified I limited the patch to the devices I could test. The following two devices are covered by this path: - Marantz SA-14S1 - Marantz HD-DAC1 Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d2aa45a..a5941f8 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1146,6 +1146,20 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); + + /* Marantz/Denon devices with USB DAC functionality need a delay + * after each class compliant request + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x154e) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { + + switch (le16_to_cpu(dev->descriptor.idProduct)) { + case 0x3005: /* Marantz HD-DAC1 */ + case 0x3006: /* Marantz SA-14S1 */ + mdelay(20); + break; + } + } } /* -- 1.9.3

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[alsa-devel] [PATCH] snd_dlsym: only do versioning checks if built --with-versioning
by John Spencer 16 Nov '14

16 Nov '14

From 291553a0132936611d48f41e924601dff8941537 Mon Sep 17 00:00:00 2001 From: John Spencer <maillist-alsa(a)barfooze.de> Date: Sat, 15 Nov 2014 13:41:00 +0100 Subject: [PATCH] snd_dlsym: only do versioning checks if built --with-versioning a combination of some of the following garbage collecting LD/CFLAGS -Os -g0 -fdata-sections -ffunction-sections -s -Wl,--gc-sections -fno-unwind-tables -fno-asynchronous-unwind-tables causes the symbol versioning marker symbols to be removed from BSS since they're otherwise unreferenced. this causes dlsym failing to find them which results in runtime breakage: $ alsamixer ALSA lib dlmisc.c:142:(snd_dlsym_verify) unable to verify version for symbol snd_config_hook_load ALSA lib conf.c:3328:(snd_config_hooks_call) symbol snd_config_hook_load is not defined inside (null) ALSA lib conf.c:3788:(snd_config_update_r) hooks failed, removing configuration cannot open mixer: No such device or address correct DSO: $ objdump -T libasound.so.2.strip | grep _snd_config_hook_load_dlsym_config_hook_001 001196bc g DO .bss 00000001 _snd_config_hook_load_dlsym_config_hook_001 incorrect DSO: $ objdump -T libs/libasound.so.2 | grep _snd_config_hook_load_dlsym_config_hook_001 00000000 g DO *ABS* 00000001 _snd_config_hook_load_dlsym_config_hook_001 since alsa was built with --without-versioning, doing those versioning checks at all is unnecessary and harmful and thus now disabled in this case. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- src/dlmisc.c | 2 ++ 1 file changed, 2 insertions(+) From 291553a0132936611d48f41e924601dff8941537 Mon Sep 17 00:00:00 2001 From: John Spencer <maillist-alsa(a)barfooze.de> Date: Sat, 15 Nov 2014 13:41:00 +0100 Subject: [PATCH] snd_dlsym: only do versioning checks if built --with-versioning a combination of some of the following garbage collecting LD/CFLAGS -Os -g0 -fdata-sections -ffunction-sections -s -Wl,--gc-sections -fno-unwind-tables -fno-asynchronous-unwind-tables causes the symbol versioning marker symbols to be removed from BSS since they're otherwise unreferenced. this causes dlsym failing to find them which results in runtime breakage: $ alsamixer ALSA lib dlmisc.c:142:(snd_dlsym_verify) unable to verify version for symbol snd_config_hook_load ALSA lib conf.c:3328:(snd_config_hooks_call) symbol snd_config_hook_load is not defined inside (null) ALSA lib conf.c:3788:(snd_config_update_r) hooks failed, removing configuration cannot open mixer: No such device or address correct DSO: $ objdump -T libasound.so.2.strip | grep _snd_config_hook_load_dlsym_config_hook_001 001196bc g DO .bss 00000001 _snd_config_hook_load_dlsym_config_hook_001 incorrect DSO: $ objdump -T libs/libasound.so.2 | grep _snd_config_hook_load_dlsym_config_hook_001 00000000 g DO *ABS* 00000001 _snd_config_hook_load_dlsym_config_hook_001 since alsa was built with --without-versioning, doing those versioning checks at all is unnecessary and harmful and thus now disabled in this case. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- src/dlmisc.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/dlmisc.c b/src/dlmisc.c index 4b8a02c..62f7047 100644 --- a/src/dlmisc.c +++ b/src/dlmisc.c @@ -177,11 +177,13 @@ void *snd_dlsym(void *handle, const char *name, const char *version) } #endif #ifdef HAVE_LIBDL +#ifdef VERSIONED_SYMBOLS if (version) { err = snd_dlsym_verify(handle, name, version); if (err < 0) return NULL; } +#endif return dlsym(handle, name); #else return NULL; -- 1.8.4

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[alsa-devel] DTS-HD over HDMI
by Kip Warner 15 Nov '14

15 Nov '14

Hey list, I apologize if this ought to have gone to alsa-user. I wasn't sure as a developer using API, but not developing ALSA itself, if it have been better there than here. I'd like to be able to stream DTS-HD Master Audio bits out the HDMI port in their compressed form, and not decompressed as LPCM. From what I can gather, this may be possible with the ALSA API, provided the HDMI port is >= 1.3 and the audio card supports high bit-rate (HBR). Is my thinking correct? -- Kip Warner -- Senior Software Engineer OpenPGP encrypted/signed mail preferred http://www.thevertigo.com

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[alsa-devel] usb-audio: check_input_term (mixer.c) iterating to clock selectors?
by Johan Aires Rastén 15 Nov '14

15 Nov '14

Hi everybody! I've just started looking at ALSA because I want to get the mic working with my Asus Xonar U7. Some of you might have seen me on IRC as johanar. I'm wondering about check_input_term(...) in sound/usb/mixer.c, which for case UAC_INPUT_TERMINAL and UAC_VERSION_2 will recursively go into the clock source. Is this correct behaviour? For my card both mic and line will show up as "Unit 22" when I call "amixer contents" since the two inputs have the same unnamed clock source (they also share physical connector). However, if I comment out the recursive call they show up as "mic" and "line" as they're supposed to. Btw, if this is correct, couldn't we set id = d->bCSourceID and break; in the same way that case UAC_FEATURE_UNIT does? //Johan Rastén

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[alsa-devel] [PATCH 1/2] spi: Fix mapping from vmalloc-ed buffer to scatter list
by Charles Keepax 14 Nov '14

14 Nov '14

We can only use page_address on memory that has been mapped using kmap, when the buffer passed to the SPI has been allocated by vmalloc the page has not necessarily been mapped through kmap. This means sometimes page_address will return NULL causing the pointer we pass to sg_set_buf to be invalid. As we only call page_address such that we can pass a virtual address to sg_set_buf which will then immediately call virt_to_page on it, fix this by calling sg_set_page directly rather then relying on the sg_set_buf helper. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/spi/spi.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/spi/spi.c b/drivers/spi/spi.c index ebcb33d..50f20f2 100644 --- a/drivers/spi/spi.c +++ b/drivers/spi/spi.c @@ -615,13 +615,13 @@ static int spi_map_buf(struct spi_master *master, struct device *dev, sg_free_table(sgt); return -ENOMEM; } - sg_buf = page_address(vm_page) + - ((size_t)buf & ~PAGE_MASK); + sg_set_page(&sgt->sgl[i], vm_page, + min, offset_in_page(buf)); } else { sg_buf = buf; + sg_set_buf(&sgt->sgl[i], sg_buf, min); } - sg_set_buf(&sgt->sgl[i], sg_buf, min); buf += min; len -= min; -- 1.7.2.5

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[alsa-devel] Question on Compressed offload session
by gsantosh＠codeaurora.org 14 Nov '14

14 Nov '14

Hi All, The Question is for the compressed offload session. For a generic codec driver during the startup function it will set some of the hw_constraints rule similarly like this. snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_12_24); pcm_lib.c will try to add the rule to the runtime structure by accessing the pointers which will be initialized during opening of the session, as The Constraints added by the codec driver will be updated in the struct snd_pcm_hw_constraints of runtime structure which will be part of substream handle. But for the compressed offload I do not see the initialization done for HW constraints, as done in pcm session 2092int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, 2093 struct file *file, 2094 struct snd_pcm_substream **rsubstream) most of the existing drivers which has the hw_constraint_list code will not be applicable for compress offload session, how to solve this? Regards, Santosh M G. Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,a Linux Foundation Collaborative Project

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