July 2020 - Alsa-devel - mailman.alsa-project.org

Re: [PATCH v2 1/7] Documentation: device-tree: sound: Update lpass-cpu driver binding
by Rohit Kumar 06 Jul '20

06 Jul '20

On 5/16/2020 11:07 PM, Ajit Pandey wrote: > On 5/14/2020 10:14 PM, Mark Brown wrote: >> On Thu, May 14, 2020 at 10:08:12PM +0530, Ajit Pandey wrote: >>> Done the required cleanups to update lpass-cpu binding with newer >>> yaml formats. >> Please do YAML conversions as the last thing in any patch series, there >> is quite a big backlog of reviews of YAML conversions so they can slow >> down the code development. It's good to do them, just try to make sure >> other patches don't depend on them. >> >> Please submit patches using subject lines reflecting the style for the >> subsystem, this makes it easier for people to identify relevant patches. >> Look at what existing commits in the area you're changing are doing and >> make sure your subject lines visually resemble what they're doing. >> There's no need to resubmit to fix this alone. > > Ok.. we'll do the yaml conversion as a separate patch chain so that it > won't block > > merging of this feature in ASoC core. @Mark, I was planning to post v3 patchset series for this and keep yaml change as last patch in the series as suggested. However, we have one update in the driver to support another compatible string. If we keep Documentation change at the last, the driver change will have checkpatch errors. Can you please suggest if should make first Documentation change in text file itself before driver change and finally have a patch to convert it to yaml at the end? Thanks, Rohit -- Qualcomm INDIA, on behalf of Qualcomm Innovation Center, Inc.is a member of the Code Aurora Forum, hosted by the Linux Foundation.

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[PATCH v2] ASoC: cros_ec_codec: Log results when EC commands fail
by Yu-Hsuan Hsu 06 Jul '20

06 Jul '20

Log results of failed EC commands to identify a problem more easily. Replace cros_ec_cmd_xfer_status with cros_ec_cmd_xfer because the result has already been checked in this function. The wrapper is not needed. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/cros_ec_codec.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8d45c628e988e..a4ab62f59efa6 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -90,10 +90,17 @@ static int send_ec_host_command(struct cros_ec_device *ec_dev, uint32_t cmd, if (outsize) memcpy(msg->data, out, outsize); - ret = cros_ec_cmd_xfer_status(ec_dev, msg); + ret = cros_ec_cmd_xfer(ec_dev, msg); if (ret < 0) goto error; + if (msg->result != EC_RES_SUCCESS) { + dev_err(ec_dev->dev, "Command %d failed: %d\n", cmd, + msg->result); + ret = -EPROTO; + goto error; + } + if (insize) memcpy(in, msg->data, insize); -- 2.27.0.212.ge8ba1cc988-goog

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Move ownership of snd-firewire-ctl-services repository
by Takashi Sakamoto 05 Jul '20

05 Jul '20

Hi Jaroslav, I'm currently working for the idea of 'ALSA control service in userspace'[1] for audio and music units on IEEE 1394 bus under my personal repository[2], and the work mostly done for Digidesign Digi 002/003 family and Tascam FireWire series[3]. As well as libhinawa and alsa-gobject repositories[4], I'd like to move ownership of the repository to 'GObject Introspection' team of alsa-project account, then continue the work. I'd like you to arrange access rights and so on, again. Besides, as I tols you in previous message[5], the nested team is not good in reasons. I strongly request you to leave 'GObject Introspection' team and drop 'Hinawa' team, again. Would I expect your arrangement for it? [1] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169141.html [2] https://github.com/takaswie/snd-firewire-ctl-services/ [3] https://github.com/takaswie/snd-firewire-ctl-services/tree/topic/services-f… [4] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169141.html [5] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169191.html Thanks Takashi Sakamoto

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Some questions about asoc codec mixer elements naming
by Станислав Александров 05 Jul '20

05 Jul '20

Hello. I am writing an asoc codec driver and stuck with mixer and output naming. Could someone clarify a few things? This codec has two playback PGAs and 4 outputs: PGA1 -> [Speaker,Headphone,Line Out] PGA2 -> [Mono Output] [PGA1] controls volume and mute for all three outputs. [Speaker,Headphone,Line Out] have only mute. [PGA2] controls volume and mute. [Mono Output] also has mute. Codec is bound to platform driver via simple-audio-card: simple-audio-card,widgets = "Headphone", "Headphones", "Speaker", "Speakers", "Line", "Line Out Jack"; simple-audio-card,routing = "Headphones", "HPOUT", "Speakers", "SPKOUT", "Line Out Jack", "LINEOUT"; Is it possible to name mixer controls so that no further configuration is required when the driver is loaded? i.e. pulseaudio automatically detects all these outputs and controls, correctly maps them and allows switching between them? Currently i have this configuration(simplified): SOC_DOUBLE_S8_TLV("Master Playback Volume", PGA1, ...) SOC_DOUBLE("Master Playback Switch", PGA1, ...) SOC_SINGLE("Speaker Playback Switch", ...) SOC_SINGLE("Headphone Playback Switch", ...) SOC_SINGLE("Line Out Playback Switch", ...) SOC_SINGLE_S8_TLV("Mono Output Playback Volume", PGA2...) SOC_SINGLE("Mono Output Playback Switch", PGA2....) With these settings pulseaudio only detects Speaker and Headphone (but not Line Out because it wants to find its Jack) But as I understand "Master Playback" assumes control for all sound streams, so this is probably not correct =\ If it is impossible to name those elements that way so everything is autodetected, what should i use then? Write a machine audio driver or use alsaucm?

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Asus Xonar SE
by Jefferson Smith Pizarro Gutierrez 05 Jul '20

05 Jul '20

hi, i have issues with my Asus Xonar SE souncard, i need to get a solution, please helpme to get sound through the spdif of the soundcard. i dont have sound

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Looking for guidance for porting home made audio driver to ASoC/FSL
by Christophe Leroy 04 Jul '20

04 Jul '20

Hello I'm looking for guidance for porting a home made audio driver to ASoC/FSL on a custom board. The board has an MPC8321 freescale/NXP cpu. The board has a TDM bus connected to one of the cpu TDM port. On that TDM bus, there are three IDT 821034 quad codecs (PCM G.711) and an E1 chip that carries IP over E1. Today, there is a home made audio driver providing raw read/write devices plus an home made WAN HDLC driver for the E1. The CPU Quicc Engine (QE) is configured to connect the TDM port to one UCC. QUICC Multi-Channel Controller (QMC) is used to distribute E1 timeslots to the HDLC driver and each of the 12 codec timeslots to the audio driver. (The QMC is able to manage two types of channels: transparent and HDLC). We would like to make all this more standard and port it to use ALSA ASoC for the audio part, and the Freescale Quicc Engine HDLC. As of today, the Freescale QE HDLC wan driver uses the UCC directly, it doesn't interface through the QMC. And the FSL ASoC drivers seem to neither handle QE TDM via QE QMC nor QE UCC. We are thinking about adding a QE QMC layer in driver/soc/fsl/ which would more or less present to the consumer the same kind of API as the FSL UCC layer, and enhance the WAN FSL HDLC driver to be able to use that new layer. For the ALSA ASoC part, could some parts of existing FSL ASoC drivers be re-used of would we have to start implementing an ASoC from scratch ? Any recommendation and/or guidance would be welcomed and appreciated. Thanks in advance Christophe

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[PATCH] ASoC: amd: fix Kconfig warning for Renoir
by Randy Dunlap 04 Jul '20

04 Jul '20

From: Randy Dunlap <rdunlap(a)infradead.org> SND_SOC_DMIC depends on GPIOLIB so this driver should also depend on GPIOLIB to make kconfig happy. WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_AMD_RENOIR_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_RENOIR [=y] Fixes: 1d3776669323 ("ASoC: amd: enable build for RN machine driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Cc: Mark Brown <broonie(a)kernel.org> --- Found in mmotm. There are no object file build errors. I suppose that gpiolib stubs take care of that. Maybe some other patch would be more appropriate. sound/soc/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) --- mmotm-2020-0703-1503.orig/sound/soc/amd/Kconfig +++ mmotm-2020-0703-1503/sound/soc/amd/Kconfig @@ -47,5 +47,6 @@ config SND_SOC_AMD_RENOIR_MACH tristate "AMD Renoir support for DMIC" select SND_SOC_DMIC depends on SND_SOC_AMD_RENOIR + depends on GPIOLIB help This option enables machine driver for DMIC

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[PATCH v2] ASoC: fsl_asrc: Add an option to select internal ratio mode
by Shengjiu Wang 04 Jul '20

04 Jul '20

The ASRC not only supports ideal ratio mode, but also supports internal ratio mode. For internal rato mode, the rate of clock source should be divided with no remainder by sample rate, otherwise there is sound distortion. Add function fsl_asrc_select_clk() to find proper clock source for internal ratio mode, if the clock source is available then internal ratio mode will be selected. With change, the ideal ratio mode is not the only option for user. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - update according to Nicolin's comments sound/soc/fsl/fsl_asrc.c | 54 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 52 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 95f6a9617b0b..4105ef2c4f99 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -582,11 +582,55 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints); } +/** + * Select proper clock source for internal ratio mode + */ +static int fsl_asrc_select_clk(struct fsl_asrc_priv *asrc_priv, + struct fsl_asrc_pair *pair, + int in_rate, + int out_rate) +{ + struct fsl_asrc_pair_priv *pair_priv = pair->private; + struct asrc_config *config = pair_priv->config; + int rate[2], select_clk[2]; /* Array size 2 means IN and OUT */ + int clk_rate, clk_index; + int i = 0, j = 0; + + rate[0] = in_rate; + rate[1] = out_rate; + + /* Select proper clock source for internal ratio mode */ + for (j = 0; j < 2; j++) { + for (i = 0; i < ASRC_CLK_MAP_LEN; i++) { + clk_index = asrc_priv->clk_map[j][i]; + clk_rate = clk_get_rate(asrc_priv->asrck_clk[clk_index]); + /* Only match a perfect clock source with no remainder */ + if (clk_rate != 0 && (clk_rate / rate[j]) <= 1024 && + (clk_rate % rate[j]) == 0) + break; + } + + select_clk[j] = i; + } + + /* Switch to ideal ratio mode if there is no proper clock source */ + if (select_clk[IN] == ASRC_CLK_MAP_LEN || select_clk[OUT] == ASRC_CLK_MAP_LEN) { + select_clk[IN] = INCLK_NONE; + select_clk[OUT] = OUTCLK_ASRCK1_CLK; + } + + config->inclk = select_clk[IN]; + config->outclk = select_clk[OUT]; + + return 0; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct fsl_asrc *asrc = snd_soc_dai_get_drvdata(dai); + struct fsl_asrc_priv *asrc_priv = asrc->private; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; struct fsl_asrc_pair_priv *pair_priv = pair->private; @@ -605,8 +649,6 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.pair = pair->index; config.channel_num = channels; - config.inclk = INCLK_NONE; - config.outclk = OUTCLK_ASRCK1_CLK; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config.input_format = params_format(params); @@ -620,6 +662,14 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } + ret = fsl_asrc_select_clk(asrc_priv, pair, + config.input_sample_rate, + config.output_sample_rate); + if (ret) { + dev_err(dai->dev, "fail to select clock\n"); + return ret; + } + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); -- 2.21.0

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[PATCH v2 0/2] ASoC: rt5670: 2 small cleanups
by Hans de Goede 03 Jul '20

03 Jul '20

Hi All, Here is in essence a resend of my 2 cleanup patches for the rt5670 ASoC codec code, rebased on top of broonie/sound/for-5.9 with broonie/sound/for-5.8 merged in. Regards, Hans p.s. I'll also send out the patch improving the comment in cht_bsw_rt5672.c cht_codec_fixup() which Pierre-Louis requested soon.

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[PATCH v2 0/6] ASoC: fsl: fix kernel-doc
by Pierre-Louis Bossart 03 Jul '20

03 Jul '20

This cleanup patchset is only about kernel-doc, mostly trivial edits and format correction. v2: added Nicolin Chen's Acked-by tags and included the three suggested edits. Pierre-Louis Bossart (6): ASoC: fsl: fsl_ssi_dbg: remove spurious kernel-doc comment start ASoC: fsl: fsl_ssi: fix kernel-doc ASoC: fsl: fsl-asoc-card: fix kernel-doc ASoC: fsl: fsl_spdif: fix kernel-doc ASoC: fsl: fsl_asrc: fix kernel-doc ASoC: fsl: fsl_esai: fix kernel-doc sound/soc/fsl/fsl-asoc-card.c | 21 +++++------ sound/soc/fsl/fsl_asrc.c | 57 +++++++++++++++++++--------- sound/soc/fsl/fsl_esai.c | 32 +++++++++------- sound/soc/fsl/fsl_spdif.c | 5 ++- sound/soc/fsl/fsl_ssi.c | 70 ++++++++++++++++++++++------------- sound/soc/fsl/fsl_ssi_dbg.c | 4 +- 6 files changed, 115 insertions(+), 74 deletions(-) -- 2.25.1

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