November 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/4] treewide: sys/poll to poll
by Rosen Penev 20 Nov '19

20 Nov '19

Fixes warning on musl: warning redirecting incorrect #include <sys/poll.h> to <poll.h> Signed-off-by: Rosen Penev <rosenp(a)gmail.com> --- amidi/amidi.c | 2 +- amixer/amixer.c | 2 +- aplay/aplay.c | 2 +- seq/aplaymidi/arecordmidi.c | 2 +- seq/aseqdump/aseqdump.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/amidi/amidi.c b/amidi/amidi.c index b6e05e1..c6268e4 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -31,7 +31,7 @@ #include <signal.h> #include <sys/timerfd.h> #include <sys/types.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/stat.h> #include <unistd.h> #include <fcntl.h> diff --git a/amixer/amixer.c b/amixer/amixer.c index ad9c482..f7f31f0 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -28,7 +28,7 @@ #include <errno.h> #include <assert.h> #include <alsa/asoundlib.h> -#include <sys/poll.h> +#include <poll.h> #include <stdint.h> #include "amixer.h" #include "../alsamixer/volume_mapping.h" diff --git a/aplay/aplay.c b/aplay/aplay.c index 274bbce..5241068 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -44,7 +44,7 @@ #include <assert.h> #include <termios.h> #include <signal.h> -#include <sys/poll.h> +#include <poll.h> #include <sys/uio.h> #include <sys/time.h> #include <sys/stat.h> diff --git a/seq/aplaymidi/arecordmidi.c b/seq/aplaymidi/arecordmidi.c index f3db65e..604cd0d 100644 --- a/seq/aplaymidi/arecordmidi.c +++ b/seq/aplaymidi/arecordmidi.c @@ -27,7 +27,7 @@ #include <string.h> #include <signal.h> #include <getopt.h> -#include <sys/poll.h> +#include <poll.h> #include <alsa/asoundlib.h> #include "aconfig.h" #include "version.h" diff --git a/seq/aseqdump/aseqdump.c b/seq/aseqdump/aseqdump.c index 7904540..578e06f 100644 --- a/seq/aseqdump/aseqdump.c +++ b/seq/aseqdump/aseqdump.c @@ -25,7 +25,7 @@ #include <string.h> #include <signal.h> #include <getopt.h> -#include <sys/poll.h> +#include <poll.h> #include <alsa/asoundlib.h> #include "aconfig.h" #include "version.h" -- 2.23.0

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[alsa-devel] [PATCH v2 1/3] ASoC: add control components management
by Jaroslav Kysela 20 Nov '19

20 Nov '19

This ASCII string can carry additional information about soundcard components or configuration. Add the possibility to set this string via the ASoC card. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 13 +++++++++++++ 2 files changed, 14 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index e0855dc08d30..bd943b5d7d45 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -982,6 +982,7 @@ struct snd_soc_card { const char *name; const char *long_name; const char *driver_name; + const char *components; char dmi_longname[80]; char topology_shortname[32]; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 55014e7ae0d8..b4683d4588ee 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2121,6 +2121,19 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) soc_setup_card_name(card->snd_card->driver, card->driver_name, card->name, 1); + if (card->components) { + /* the current implementation of snd_component_add() accepts */ + /* multiple components in the string separated by space, */ + /* but the string collision (identical string) check might */ + /* not work correctly */ + ret = snd_component_add(card->snd_card, card->components); + if (ret < 0) { + dev_err(card->dev, "ASoC: %s snd_component_add() failed: %d\n", + card->name, ret); + goto probe_end; + } + } + if (card->late_probe) { ret = card->late_probe(card); if (ret < 0) { -- 2.20.1

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[alsa-devel] [PATCH v5 0/7] ALSA: aloop: Support sound timer as clock source instead of jiffies
by Andrew Gabbasov 20 Nov '19

20 Nov '19

This patch set is an updated version of patches by Timo Wischer: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-March/146871.html This patch set is required for forwarding audio data between a HW sound card and an aloop device without the usage of an asynchronous sample rate converter. Most of sound and timers related code is kept the same as in previous set. The code, related to snd_pcm_link() functionality and its using for timer source setting, is removed (as rejected earlier). The changes in this update are mainly related to the parameters handling and some cleanup. The timer source can be initially selected by "timer_source" kernel module parameter. It is supposed to have the following format: [<pref>:](<card name>|<card idx>)[{.,}<dev idx>[{.,}<subdev idx>]] For example: "hw:I82801AAICH.1.0", or "1.1", or just "I82801AAICH". (Prefix is ignored, just allowed here to be able to use the strings, the user got used to). Although the parsing function recognizes both '.' and ',' as a separator, module parameters handling routines use ',' to separate parameters for different module instances (array elements), so we have to use '.' to separate device and subdevice numbers from the card name or number in module parameters. Empty string indicates using jiffies as a timer source. Besides "static" selection of timer source at module load time, it is possible to dynamically change it via sound "info" interface (using "/proc/asound/<card>/timer_source" file in read-write mode. The contents of this file is used as a timer source string for a particular loopback card, e.g. "hw:0,0,0" (and here ',' can be used as a separator). The timer source string value can be changed at any time, but it is latched by PCM substream open callback "loopback_open()" (the first one for a particular cable). At this point it is actually used, that is the string is parsed, and the timer is looked up and opened. This seems to be a good trade-off between flexibility of updates and synchronizations or racing complexity. The timer source is set for a loopback card (the same as initial setting by module parameter), but every cable uses the value, current at the moment of opening. Theoretically, it's possible to set the timer source for each cable independently (via separate files), but it would be inconsistent with the initial setting via module parameters on a per-card basis. v2: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-November/157961.… v3: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-November/158312.… - Change sound card lookup to use snd_card_ref() and avoid direct access to sound cards array - Squash commits on returning error codes for timer start and stop - Some locking related fixes - Some code cleanup v4: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-November/158896.… - Change to use updated API for snd_timer_open() (separate timer instance) - Change to use snd_timer_close() returning void - Some code cleanup v5: - Change to initialize timer event tasklet before calling snd_timer_open() Andrew Gabbasov (1): ALSA: aloop: Support runtime change of snd_timer via info interface Timo Wischer (6): ALSA: aloop: Describe units of variables ALSA: aloop: Support return of error code for timer start and stop ALSA: aloop: Use callback functions for timer specific implementations ALSA: aloop: Rename all jiffies timer specific functions ALSA: aloop: Move CABLE_VALID_BOTH to the top of file ALSA: aloop: Support selection of snd_timer instead of jiffies sound/drivers/aloop.c | 663 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 628 insertions(+), 35 deletions(-) -- 2.21.0

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[alsa-devel] [PATCH 1/2] ASoC: improve the DMI long card code in asoc-core
by Jaroslav Kysela 20 Nov '19

20 Nov '19

Add append_dmi_string() function and make the code more readable. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 66 +++++++++++++++++--------------------------- 1 file changed, 25 insertions(+), 41 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b4683d4588ee..1d1baf78818c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1755,6 +1755,23 @@ static int is_dmi_valid(const char *field) return 1; } +/* + * + */ +static void append_dmi_string(struct snd_soc_card *card, const char *str) +{ + char *dst = card->dmi_longname; + size_t dst_len = sizeof(card->dmi_longname); + size_t len; + + len = strlen(dst); + snprintf(dst + len, dst_len - len, "-%s", str); + + len++; /* skip the separator "-" */ + if (len < dst_len) + cleanup_dmi_name(dst + len); +} + /** * snd_soc_set_dmi_name() - Register DMI names to card * @card: The card to register DMI names @@ -1789,61 +1806,36 @@ static int is_dmi_valid(const char *field) int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) { const char *vendor, *product, *product_version, *board; - size_t longname_buf_size = sizeof(card->snd_card->longname); - size_t len; if (card->long_name) return 0; /* long name already set by driver or from DMI */ - /* make up dmi long name as: vendor.product.version.board */ + /* make up dmi long name as: vendor-product-version-board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor || !is_dmi_valid(vendor)) { dev_warn(card->dev, "ASoC: no DMI vendor name!\n"); return 0; } - snprintf(card->dmi_longname, sizeof(card->snd_card->longname), - "%s", vendor); + snprintf(card->dmi_longname, sizeof(card->dmi_longname), "%s", vendor); cleanup_dmi_name(card->dmi_longname); product = dmi_get_system_info(DMI_PRODUCT_NAME); if (product && is_dmi_valid(product)) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", product); - - len++; /* skip the separator "-" */ - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); + append_dmi_string(card, product); /* * some vendors like Lenovo may only put a self-explanatory * name in the product version field */ product_version = dmi_get_system_info(DMI_PRODUCT_VERSION); - if (product_version && is_dmi_valid(product_version)) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", product_version); - - len++; - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); - } + if (product_version && is_dmi_valid(product_version)) + append_dmi_string(card, product_version); } board = dmi_get_system_info(DMI_BOARD_NAME); if (board && is_dmi_valid(board)) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", board); - - len++; - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); + append_dmi_string(card, board); } else if (!product) { /* fall back to using legacy name */ dev_warn(card->dev, "ASoC: no DMI board/product name!\n"); @@ -1851,16 +1843,8 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) } /* Add flavour to dmi long name */ - if (flavour) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", flavour); - - len++; - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); - } + if (flavour) + append_dmi_string(card, flavour); /* set the card long name */ card->long_name = card->dmi_longname; -- 2.20.1

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[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5640: Update quirk for Acer Switch 10 SW5-012 2-in-1
by Hans de Goede 20 Nov '19

20 Nov '19

When the Acer Switch 10 SW5-012 quirk was added we did not have jack-detection support yet; and the builtin microphone selection of the original quirk is wrong too. Fix the microphone-input quirk and add jack-detection info so that the internal-microphone and headphone/set jack on the Switch 10 work properly. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 9c1aa4ec9cba..dd2b5ad08659 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -405,10 +405,12 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Acer"), DMI_MATCH(DMI_PRODUCT_NAME, "Aspire SW5-012"), }, - .driver_data = (void *)(BYT_RT5640_IN1_MAP | - BYT_RT5640_MCLK_EN | - BYT_RT5640_SSP0_AIF1), - + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), }, { .matches = { -- 2.23.0

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[alsa-devel] [PATCH 1/2] ASoC: add control components management
by Jaroslav Kysela 20 Nov '19

20 Nov '19

This ASCII string can carry additional information about soundcard components or configuration. Add the possibility to set this string via the ASoC card. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 13 +++++++++++++ 2 files changed, 14 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index e0855dc08d30..bd943b5d7d45 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -982,6 +982,7 @@ struct snd_soc_card { const char *name; const char *long_name; const char *driver_name; + const char *components; char dmi_longname[80]; char topology_shortname[32]; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 55014e7ae0d8..b4683d4588ee 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2121,6 +2121,19 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) soc_setup_card_name(card->snd_card->driver, card->driver_name, card->name, 1); + if (card->components) { + /* the current implementation of snd_component_add() accepts */ + /* multiple components in the string separated by space, */ + /* but the string collision (identical string) check might */ + /* not work correctly */ + ret = snd_component_add(card->snd_card, card->components); + if (ret < 0) { + dev_err(card->dev, "ASoC: %s snd_component_add() failed: %d\n", + card->name, ret); + goto probe_end; + } + } + if (card->late_probe) { ret = card->late_probe(card); if (ret < 0) { -- 2.20.1

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[alsa-devel] [PATCH 0/2] ASoC: pcm3168a: Update the reset GPIO handling
by Peter Ujfalusi 20 Nov '19

20 Nov '19

Hi, Switch the rst-gpios to generic reset-gpios as suggested by Rob. Based on other threads around GPIOs, revise the GPIO active vs assert/deassert configuration: The RST pin is a reset pin and it is active low. Regards, Peter --- Peter Ujfalusi (2): ASoC: dt-bindings: pcm3168a: Update the optional RST gpio for clarity ASoC: pcm3168a: Update the RST gpio handling to align with documentation .../devicetree/bindings/sound/ti,pcm3168a.txt | 9 +++++---- sound/soc/codecs/pcm3168a.c | 20 ++++++++++++++----- 2 files changed, 20 insertions(+), 9 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] Revert "ASoC: soc-pcm: check symmetry after hw_params"
by Shengjiu Wang 20 Nov '19

20 Nov '19

This reverts commit 957ce0c6b8a1f26559864507ae0bfcba29d924ad. That commit cause soc_pcm_params_symmetry can't take effect. cpu_dai->rate, cpu_dai->channels and cpu_dai->sample_bits are updated in the middle of soc_pcm_hw_params, so move soc_pcm_params_symmetry to the end of soc_pcm_hw_params is not a good solution, for judgement of symmetry in the function is always true. FIXME: According to the comments of that commit, I think the case described in the commit should disable symmetric_rates in Back-End, rather than changing the position of soc_pcm_params_symmetry. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/soc-pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8655df6a6089..b7800c95327a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -861,6 +861,11 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, int i, ret = 0; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + + ret = soc_pcm_params_symmetry(substream, params); + if (ret) + goto out; + if (rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { @@ -940,9 +945,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } component = NULL; - ret = soc_pcm_params_symmetry(substream, params); - if (ret) - goto component_err; out: mutex_unlock(&rtd->card->pcm_mutex); return ret; -- 2.21.0

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[alsa-devel] [PATCH] ASoC: Fix Kconfig indentation
by Krzysztof Kozlowski 20 Nov '19

20 Nov '19

Adjust indentation from spaces to tab (+optional two spaces) as in coding style with command like: $ sed -e 's/^ /\t/' -i */Kconfig Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- sound/soc/intel/boards/Kconfig | 24 ++++++++++++------------ sound/soc/sof/intel/Kconfig | 10 +++++----- 2 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 2702aefee775..ef20316e83d1 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -441,18 +441,18 @@ config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH If unsure select "N". config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH - tristate "CML with RT1011 and RT5682 in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_RT1011 - select SND_SOC_RT5682 - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC machine driver for SOF platform with - RT1011 + RT5682 I2S codec. - Say Y if you have such a device. - If unsure select "N". + tristate "CML with RT1011 and RT5682 in I2S Mode" + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT1011 + select SND_SOC_RT5682 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC machine driver for SOF platform with + RT1011 + RT5682 I2S codec. + Say Y if you have such a device. + If unsure select "N". endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index b27fd3fdf335..cc09bb606f7d 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -264,16 +264,16 @@ config SND_SOC_SOF_ELKHARTLAKE config SND_SOC_SOF_JASPERLAKE_SUPPORT bool "SOF support for JasperLake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the JasperLake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the JasperLake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_JASPERLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_HDA_COMMON -- 2.17.1

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[alsa-devel] cp: upmix: code formatting and cleanup, generalize format
by GitHub pull_request - opened 20 Nov '19

20 Nov '19

alsa-project/alsa-plugins pull request #2 was opened from bku-sue: The main purpose of this change is to be able to change the format to e.g. S32 instead of S16 by changing only two lines on the top of the source. So a new symbol `UPMIX_PCM_FORMAT` is added to hide the actual format, and a special type `upmix_sample_t` marks the sample type, instead of fixed `short`. I fixed the `average_copy` function as well, which was actually expecting the `nchns` parameter to be 2, otherwise its local arrays would be overindexed. The rest is just better naming for `i` as it denotes the channel index in every case, and applying consistent whitespace. The behavior of the plugin remains identical though. Request URL : https://github.com/alsa-project/alsa-plugins/pull/2 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/2.patch Repository URL: https://github.com/alsa-project/alsa-plugins

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