August 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Fwd: [PATCH 1/2] ALSA: hda - Turn CX8200 into D3 as well upon reboot
by Ju Hyung Park 18 Aug '18

18 Aug '18

Oh, and I forgot to add: this would better be applied to every stable trees, if applicable (probably by checking if the commit f6b28e4ded45bb91bc4cd115d55e35badedfce5f is already merged). Thanks again. On Sat, Aug 18, 2018 at 11:38 PM Greg KH <gregkh(a)linuxfoundation.org> wrote: > > On Sat, Aug 18, 2018 at 11:29:11PM +0900, Ju Hyung Park wrote: > > Hi, Greg. > > > > I'd like to know if there's a reason on why these ALSA patches(2 in > > total) aren't brought to linux-stable yet. > > What specific patches are you referring to here? I do not see anything > in this email :( > > What is the git commit id in Linus's tree? What stable tree(s) do you > want them applied to? > > > I see a lot of patches merged to linux-stable and stable-queue that > > are much newer than those patches. > > > > I'd appreciate it if you can merge those ASAP. > > Have you read: > https://www.kernel.org/doc/html/latest/process/stable-kernel-rules.html > for how to do this properly? > > If not, please do so. > > thanks, > > greg k-h

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[alsa-devel] [PATCH v2] ASoC: tegra: probe deferral error reporting
by Marcel Ziswiler 18 Aug '18

18 Aug '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Actually report the error codes from of_get_named_gpio() resp. devm_gpio_request_one() upon trying to get the codec reset resp. sync GPIOs unless it is just a probe deferral. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- Changes in v2: - Silence probe deferral as suggested by Stefan. - Fix line over 80 characters as reported by checkpatch. sound/soc/tegra/tegra20_ac97.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 682ef33afb5f..2ac6b2d73980 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -351,18 +351,31 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = devm_gpio_request_one(&pdev->dev, ac97->reset_gpio, GPIOF_OUT_INIT_HIGH, "codec-reset"); if (ret) { - dev_err(&pdev->dev, "could not get codec-reset GPIO\n"); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "could not get codec-reset GPIO: %d\n", + ret); + goto err_clk_put; } } else { - dev_err(&pdev->dev, "no codec-reset GPIO supplied\n"); + ret = ac97->reset_gpio; + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "no codec-reset GPIO supplied: %d\n", + ret); + goto err_clk_put; } ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, "nvidia,codec-sync-gpio", 0); if (!gpio_is_valid(ac97->sync_gpio)) { - dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); + ret = ac97->sync_gpio; + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "no codec-sync GPIO supplied: %d\n", + ret); + goto err_clk_put; } -- 2.14.4

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[alsa-devel] hinawa-utils v0.1.0 and libhinawa v1.0.0 release
by Takashi Sakamoto 18 Aug '18

18 Aug '18

Hi, Today I released hinawa-utils v0.1.0[1]. This package consists of Python 3 modules and scripts to control elements such as output volume in Audio and Music units on IEEE 1394 bus, handled by ALSA firewire stack. At present, all of the scripts are designed to work as CUI, without any server applications. They're listed below: * hinawa-alesis-io-cui * CUI tool for Alesis iO|14 and iO|26 * hinawa-bebob-parser * Plug structure parser for BeBoB firmware * hinawa-config-rom-printer * A lexer/parser of configuration ROM on IEEE 1394 bus * hinawa-dg00x-cui * CUI tool for Digidesign Digi 00x family * hinawa-dice-common-cui * CUI tool for Dice common functionalities * hinawa-dice-extension-cui * CUI tool for Dice extended functionalities a.k.a EAP * hinawa-fireworks-cui * CUI tool for Echo Audio Fireworks module * hinawa-focusrite-saffirepro-io-cui * CUI tool for Focusrite SaffirePro IO series * hinawa-griffin-firewave-cui * CUI tool for Griffin Firewave * hinawa-lacie-speakers-cui * CUI tool for Lacie FireWire speakers * hinawa-motu-common-cui * CUI tool for common functionalities of MOTU FireWire series * hinawa-maudio-bebob-cui * CUI tool for M-Audio FireWire series based on BeBoB platform * hinawa-oxfw-generic-cui * CUI tool for OXFW generic functionalities * hinawa-tascam-fireone-cui * CUI tool for Tascam FireOne * hinawa-tascam-fw-rack-cui * CUI tool for rack models of Tascam FireWire series (FW1804) * hinawa-tascam-fw-console-cui * CUI tool for console models of Tascam FireWire series (FW1082/1884) * hinawa-yamaha-terratec-cui * CUI tool for Yamaha GO series and Terratec PHASE series They're written by Python 3 language, with a help of PyGObject[2] for gobject-introspection[3] to libhinawa API. At the same time, I released libhinawa v1.0.0[4]. This I/O library is designed for asynchronous transactions on IEEE 1394 bus via ioctl commands defined by Linux FireWire subsystem. Additionally, this library is also an application of ALSA HwDep interface supported by ALSA firewire stack. An overview of its design is illustrated[5]. I firstly introduced this library three years ago[6]. This v1.0.0 release loses backward compatibility to a part of public APIs against v0 releases in these three years, and now becomes stable with updated major version. Python 3 modules and scripts in hinawa-utils work with libhinawa v1 APIs. If you'd like to use them, please install libhinawa v1 release instead of v0 releases. I note that modules in hinawa-utils are not stable yet, as its version means. This library has already been packaged to Debian distribution[7] and available in it and its derivatives such as Ubuntu. The v1.0.0 release is already available in their repositories[8][9]. For hinawa-utils, I already start packaging to Debian distribution and get a help of Debian contributer, Kentaro Hayashi, again. We already registered ITP[10] and RFS[11] and wait for sponsors of debian developers to publish. If you find bugs, or have wishes, I suggest to file them as issues in each project on github.com. Of course, I welcome your Pull Requests. I note that these projects are heavily based on vendor-dependent functionalities, thus all of filed issues can not be solved because what I can do is for devices I own. # Off the subject, Kentaro is currently on a process to be Debian # Maintainer(DM)[12]. I greatly appreciate his care and contribution to # libhinawa/hinawa-utils. I hope he will get good experiences from his # next great journey as DM. [1] Release version 0.1.0 release · takaswie/hinawa-utils https://github.com/takaswie/hinawa-utils/releases/tag/v0.1.0 [2] Overview — PyGObject https://pygobject.readthedocs.io/en/latest/ [3] Overview — GObject Introspection https://gi.readthedocs.io/en/latest/ [4] Release version 1.0.0 release · takaswie/libhinawa https://github.com/takaswie/libhinawa/releases/tag/1.0.0 [5] libhinawa/doc/overview.png https://github.com/takaswie/libhinawa/blob/master/doc/overview.png [6] [alsa-devel] [RFC][PATCH 00/13] alsa-tools: libhinawa for control applications of FireWire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2015-January/086969.ht… [7] [alsa-devel] About planning of libhinawa ITP to debian project http://mailman.alsa-project.org/pipermail/alsa-devel/2016-January/103693.ht… [8] Debian Package Tracker - libhinawa https://tracker.debian.org/pkg/libhinawa [9] libhinawa package : Ubuntu https://launchpad.net/ubuntu/+source/libhinawa [10] #905840 - ITP: hinawa-utils - Utility tools for libhinawa - Debian Bug report logs https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=905840 [11] #905906 - RFS: hinawa-utils/0.0.99-1 [ITP] - Debian Bug report logs https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=905906 [12] Debian New Member Process https://nm.debian.org/process/531 Regards Takashi Sakamoto

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[alsa-devel] [PATCH] sound/core.c: fix wrong return code of rsnd_probe() when devm_kzalloc() fails
by Jiecheng Wu 18 Aug '18

18 Aug '18

Function rsnd_probe() defined in sound/soc/sh/rcar/core.c calls devm_kzalloc(). As devm_kzalloc() may return NULL when OOM happens, the caller should return -ENOMEM when devm_kzalloc() fails. --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f8425d8..fd92906 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1502,7 +1502,7 @@ static int rsnd_probe(struct platform_device *pdev) */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) - return -ENODEV; + return -ENOMEM; priv->pdev = pdev; priv->flags = (unsigned long)of_device_get_match_data(dev); -- 2.6.4

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[alsa-devel] [PATCH v6 0/7] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 17 Aug '18

17 Aug '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. The default HDaudio legacy driver does not allow for the use of the DSP, this patch set makes it possible while reusing existing code for HDAudio codecs and without significant changes to the legacy driver. This v6 is based on Marks' for-next branch - the merge of topic/drm_audio_component is not included here. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). Both the HDaudio and HDMI outputs were tested. Credits: all the initial code was written by Rakesh Ughreja, the rebase to broonie/for-next, cleanups and additional tests were done by Pierre Bossart. TODO in future update: 1. fix the HDMI jack detection which only works after the mixer values are set, which isn't practical for headless devices always connected (this is a problem in the hdac_hdmi codec that was present before this series was submitted) 2. when the screen goes blank, HDMI playback stops. This isn't related to this patchset since we rely on the existing hdac_hdmi codec, but will need additional investigations. Changes v6 (comments from Mark and Takashi) - simplify Kconfig options, remove FORCE for now and make code simpler - change SPDIX style to C++ for C files - add log message if invalid configuration reached - remove static int counter, added private counter in context instead - fixed a couple of indendation issues to make checkpatch.pl --strict happy Changes v5 (comments from Vinod) - fix SPDIF style across patches - patch 1: update commit message to remove reference to "fix" - patch 3: remove platform name assignment and Xmas tree style - patch 7: simplify error handling, add missing _put() on errors, remove tests on ops, add switch statement, fix alignment - patch 8: fix indentation issues, use bool instead of tristate Changes v4: - rebase/update on Takashi's topic/drm_audio_component branch - changes in the HDaudio detection to avoid adding a fake ACPI ID - new Kconfigs to control HDaudio detection Changes v3: - port to component model - additional tests on ApolloLake and KabyLake NUC devices - cleanups (alignment, typos, etc) Changes v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Pierre-Louis Bossart (2): ASoC: Intel: common: add table for HDA-based platforms ASoC: Intel: Skylake: add option to control HDAudio + DSP usage Rakesh Ughreja (5): ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers include/sound/soc-acpi-intel-match.h | 6 + sound/pci/hda/hda_bind.c | 12 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 485 ++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 + sound/soc/intel/Kconfig | 9 + sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 127 +++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 38 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 181 +++++++ sound/soc/intel/common/Makefile | 3 +- .../intel/common/soc-acpi-intel-hda-match.c | 40 ++ sound/soc/intel/skylake/skl-pcm.c | 70 ++- sound/soc/intel/skylake/skl.c | 104 +++- sound/soc/intel/skylake/skl.h | 12 +- 17 files changed, 1102 insertions(+), 27 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c -- 2.17.1

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[alsa-devel] [PATCH v2 1/2] ASoC: adau17x1: Unused exported functions changed to internal
by Robert Rosengren 17 Aug '18

17 Aug '18

adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.11.0

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[alsa-devel] [RFC PATCH] ASoC: core: Optimise suspend/resume of DAPM widgets
by Jon Hunter 16 Aug '18

16 Aug '18

For soundcards that have several DAI links and many DAPM widgets the time taken for snd_soc_suspend to execute has been observed to be several milliseconds. The time is largely spent executing dapm_power_widgets() for each for the DAI links that need to be suspended. Given that dapm_power_widgets() is called again after suspending/resuming the DAI links, one way to optimise the suspend/resume time is to avoid calling dapm_power_widgets() for each DAI link and reduces the suspend time significantly. Please note that this has been observed on the Tegra210 Jetson TX1 platform which is not currently supported in the mainline for audio but has been tested with out-of-tree patches to enable I2S audio. For Tegra210 the audio drivers (for I2S, etc) are implemented as component drivers to offer greater flexibility with dynamic audio routing within the Tegra Audio Processing Engine (APE) from userspace. The downside of this flexibility is that now there are a lof more DAI links and so the time to suspend the soundcard can take several milliseconds. In the resume path, it is not clear if there could be any issues from not sync'ing the DAPM power state until after unmuting and resuming the CPU DAI drivers, because this will happen later with this change. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- It is not 100% clear to me if there are any downsides to this approach but I am looking to find away to avoid calling dapm_power_widgets() so many times in the suspend path when there are many DAI links. Unfortunately, I don't think that I can simply mark some of the DAI links with 'ignore_suspend' because it could be possible that they are active. include/sound/soc-dapm.h | 2 ++ sound/soc/soc-core.c | 18 ++++-------------- sound/soc/soc-dapm.c | 43 ++++++++++++++++++++++++++++++++++++++++--- 3 files changed, 46 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index af9ef16cc34d..5dbd601154c4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -427,6 +427,8 @@ void snd_soc_dapm_reset_cache(struct snd_soc_dapm_context *dapm); /* dapm events */ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event); +void snd_soc_dapm_stream_suspend(struct snd_soc_pcm_runtime *rtd, int stream); +void snd_soc_dapm_stream_resume(struct snd_soc_pcm_runtime *rtd, int stream); void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* external DAPM widget events */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 81b27923303d..03c5fa95afb8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -519,13 +519,8 @@ int snd_soc_suspend(struct device *dev) if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_SUSPEND); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_SUSPEND); + snd_soc_dapm_stream_suspend(rtd, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dapm_stream_suspend(rtd, SNDRV_PCM_STREAM_CAPTURE); } /* Recheck all endpoints too, their state is affected by suspend */ @@ -637,13 +632,8 @@ static void soc_resume_deferred(struct work_struct *work) if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_RESUME); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_RESUME); + snd_soc_dapm_stream_resume(rtd, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dapm_stream_resume(rtd, SNDRV_PCM_STREAM_CAPTURE); } /* unmute any active DACs */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e96793050c9..7d4f75d29d55 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4201,7 +4201,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) } static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - int event) + int event, bool power_widgets) { int i; @@ -4209,7 +4209,8 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, for (i = 0; i < rtd->num_codecs; i++) soc_dapm_dai_stream_event(rtd->codec_dais[i], stream, event); - dapm_power_widgets(rtd->card, event); + if (power_widgets) + dapm_power_widgets(rtd->card, event); } /** @@ -4229,7 +4230,43 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, struct snd_soc_card *card = rtd->card; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - soc_dapm_stream_event(rtd, stream, event); + soc_dapm_stream_event(rtd, stream, event, true); + mutex_unlock(&card->dapm_mutex); +} + +/** + * snd_soc_dapm_stream_suspend - send a suspend stream event to the dapm core + * @rtd: PCM runtime data + * @stream: stream name + * + * Sends a suspend stream event to the dapm core. + * + * Returns 0 for success else error. + */ +void snd_soc_dapm_stream_suspend(struct snd_soc_pcm_runtime *rtd, int stream) +{ + struct snd_soc_card *card = rtd->card; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + soc_dapm_stream_event(rtd, stream, SND_SOC_DAPM_STREAM_SUSPEND, false); + mutex_unlock(&card->dapm_mutex); +} + +/** + * snd_soc_dapm_stream_resume - send a resume stream event to the dapm core + * @rtd: PCM runtime data + * @stream: stream name + * + * Sends a resume stream event to the dapm core. + * + * Returns 0 for success else error. + */ +void snd_soc_dapm_stream_resume(struct snd_soc_pcm_runtime *rtd, int stream) +{ + struct snd_soc_card *card = rtd->card; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + soc_dapm_stream_event(rtd, stream, SND_SOC_DAPM_STREAM_RESUME, false); mutex_unlock(&card->dapm_mutex); } -- 2.7.4

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[alsa-devel] [ANN] Linux Media Summit 2018 together with ELC-EU
by Mauro Carvalho Chehab 16 Aug '18

16 Aug '18

Hi all, We're organizing this year's media summit to happen together with the Embedded Linux Conference - Europe, on Thursday, October 25 2018: https://events.linuxfoundation.org/events/elc-openiot-europe-2018/ Media Summit is the premier forum to discuss the Linux multimedia development for cameras, audio and video streaming devices, analog/digital TV support, remote controller and HDMI Consumer Electronics Control (CEC) at the Linux Kernel and its related userspace APIs. The ones interested on participating on it should be registered at ELC or OSS Europe, and indicate the participation at the media summit via Linux Foundation's register site: https://events.linuxfoundation.org/events/elc-openiot-europe-2018/register/ We're also asking for the themes to be discussed there. Please send your proposals to the linux-media(a)vger.kernel.org mailing list. In order to make easier for us to track, please add a [RFP] at the beginning of the email's subject. Hoping to see you there, Regards, Mauro

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[alsa-devel] [PATCH] ASoC: max98373: Added speaker FS gain cotnrol register to volatile.
by Ryan Lee 16 Aug '18

16 Aug '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes : Added Speaker FS gain maximum control(0x203E) register to volatile. sound/soc/codecs/max98373.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 92b7125..2764fae 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -520,6 +520,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case MAX98373_R2000_SW_RESET ... MAX98373_R2009_INT_FLAG3: + case MAX98373_R203E_AMP_PATH_GAIN: case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: rt5514: Fix the issue of the delay volume applied again
by Oder Chiou 16 Aug '18

16 Aug '18

After our evaluation, we need to modify the default values to make sure the volume applied immediately. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index dca82dd..32fe76c 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000342}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000342}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000342}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000342}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- 2.7.4

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