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

[alsa-devel] [RFC PATCH 0/9] soundwire: add Master device support, GreyBus style
by Pierre-Louis Bossart 18 Sep '19

18 Sep '19

The current use of platform devices leads to limitations in terms of API and problematic reference counts flagged during the review of sysfs patches. As suggested by Greg KH, this patch series introduces a 'Master device' and gets rid of platform devices. The code is inspired by GreyBus, e.g. 'gb_hd_driver' is translated for the SoundWire context as 'sdw_md_driver' and likewise 'gb_host_device' is translated as 'sdw_master_device'. There are differences in the way devices are created, using device_register instead of the multiple steps used in GreyBus. All I know about GreyBus is that it's based on Unipro, so the code updates are based on a first-order analysis, it could very well be I missed concepts while trying to reverse-engineer the code in staging/greybus/. I focused on the concepts mainly and it's also possible that the dual-license for new files or plan-vanilla EXPORT_SYMBOL are not appropriate for all the changes in this series. Please don't take errors as deliberate intent to work-around GPL but rather as an indicator of maturity of the code only developed in the last few days. These changes make it possible to provide new callbacks, e.g splitting the bus startup from the initializations in probe, which is very much desired on Intel platforms to detect the machine driver as early as possible before the hardware is fully powered/enable. These patches will be provided as a separate RFC later today to illustrate the benefit of these changes. To make the code more consistent the first series of patches are renames. I did not rename 'sdw_slave' or 'module_sdw_driver' to avoid compatibility issues since there are codec drivers almost ready for integration. I don't have any specific opinion on if/when additional renames should be done, as long as there is a means to clearly identify what is specific to a SoundWire Master I am fine. Feedback and reviews welcome. Bard Liao (1): soundwire: add device driver to sdw_md_driver Pierre-Louis Bossart (8): soundwire: renames to prepare support for master drivers/devices soundwire: rename dev_to_sdw_dev macro soundwire: rename drv_to_sdw_slave_driver macro soundwire: bus_type: rename sdw_drv_ to sdw_slave_drv soundwire: intel: rename res field as link_res soundwire: add support for sdw_slave_type soundwire: add initial definitions for sdw_master_device soundwire: intel: remove platform devices and provide new interface drivers/base/regmap/regmap-sdw.c | 4 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 2 +- drivers/soundwire/bus_type.c | 60 +++--- drivers/soundwire/intel.c | 117 ++++++----- drivers/soundwire/intel.h | 22 +-- drivers/soundwire/intel_init.c | 293 +++++++++++++++++++--------- drivers/soundwire/master.c | 80 ++++++++ drivers/soundwire/slave.c | 9 +- include/linux/soundwire/sdw.h | 39 +++- include/linux/soundwire/sdw_intel.h | 86 +++++++- include/linux/soundwire/sdw_type.h | 34 +++- 12 files changed, 551 insertions(+), 197 deletions(-) create mode 100644 drivers/soundwire/master.c -- 2.20.1

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[alsa-devel] [PATCH] ASoC: rsnd: do error check after rsnd_channel_normalization()
by Kuninori Morimoto 18 Sep '19

18 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SSI need to use rsnd_channel_normalization() for TDM-split mode, thus, channel check need to do after that. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark This is for v5.4 sound/soc/sh/rcar/ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f6a7466..fc5d089 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -286,6 +286,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; + if (rsnd_runtime_is_tdm_split(io)) + chan = rsnd_io_converted_chan(io); + + chan = rsnd_channel_normalization(chan); + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); @@ -300,11 +305,6 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, return 0; } - if (rsnd_runtime_is_tdm_split(io)) - chan = rsnd_io_converted_chan(io); - - chan = rsnd_channel_normalization(chan); - main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); -- 2.7.4

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[alsa-devel] [PATCH V2 0/4] update supported sample format
by Shengjiu Wang 18 Sep '19

18 Sep '19

This patch serial is to update the supported format for fsl_asrc and fix some format issue. Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_set_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 48 +++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 62 ++------------------ 6 files changed, 181 insertions(+), 89 deletions(-) -- 2.21.0

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[alsa-devel] RFC: TDM mode support on the tegra30
by Ben Dooks 18 Sep '19

18 Sep '19

This is an initial set that adds support for both the TDM and for formats other than 8/16bit audio through the audio system. It also fixes a minor issue with the FIFO stopping order. I have further patches following on which deal with selection of clocks to allow the use of the tegra's audio unit to run as clock slave (all current examples have the tegra as clock master).

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[alsa-devel] Licence change for your alsa-lib UCM commits (Marcin Juszkiewicz: 5 total)
by Jaroslav Kysela 18 Sep '19

18 Sep '19

As discussed, we would like to move the UCM configuration files from the alsa-lib repository to new alsa-ucm-conf repository with the licence change from LGPL-2.1 to BSD-3-Clause. I would like to check, if you agree with this licence change. Please, answer to this e-mail and write your agreement / disagreement (keep CC to the alsa-devel mailing list for the archiving purposes). Thank you for your time and co-operation. Reference: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-July/153257.html List of your commit(s): f07c14b08280fb461b7f070d55a625e1f5e88a1e conf/ucm: DAISY-I2S: add Samsung ARM Chromebook UCM config from ChromeOS 5b019ea310e6a5d6ff8933bcbca1f3f3471be33b conf/ucm: PandaBoard: add UCM config from Ubuntu 9d0f93af0b35bfdcbd3d758d98715508b69d6008 conf/ucm: PandaBoardES: add UCM config from Ubuntu 7db247be11381bef6e38a41f1d51fa78a99350b5 conf/ucm: SDP4430: add UCM config from Ubuntu 52de910868f56ee7e6b07a1a2ad6a54913f745d0 conf/ucm: tegraalc5632: add UCM config from Ubuntu --- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[alsa-devel] 5.1 to stereo using libasound
by Eric Desjardins 18 Sep '19

18 Sep '19

Hi, I am playing a mkv with 5.1 channels from a mkv. I configure libasound for 6 channels and I feed them interleave audio using snd_pcm_writei. I use the default pulse audio device. So far, nothing is wrong. However, I do not ear things correctly. I listen this on my stereo speaker. There is no mixdown to stereo and I was wondering how to do that. I tried to use chmap api but anything I try on the default device returns an error. I was wondering what was the correct way to do that? Using the mixing api? How do we detect that my device output stereo if chmap doesn't work? I am on a ubuntu 18.04 linux box. Thanks, Eric

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[alsa-devel] [PATCH v6 0/4] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 18 Sep '19

18 Sep '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. The dependent change on hdmi-codec.c https://patchwork.kernel.org/patch/11047447 has been picked up by Mark Brown in ASoC tree for-5.4 branch. Changes from v5 to v6: 1. Remove the patch for sound/soc/codecs/hdmi-codec.c because it is accepted. 2. Rebase the rest of patches based on drm-misc-next tree. Cheng-Yi Chiang (4): drm: bridge: dw-hdmi: Report connector status using callback drm: dw-hdmi-i2s: Use fixed id for codec device ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 13 +- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++++++- include/drm/bridge/dw_hdmi.h | 4 + sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 +- sound/soc/rockchip/rockchip_max98090.c | 116 ++++++++++++++---- 6 files changed, 153 insertions(+), 27 deletions(-) -- 2.23.0.237.gc6a4ce50a0-goog

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[alsa-devel] [PATCH v2] ASoC: atmel_ssc_dai: Remove wrong spinlock usage
by Gregory CLEMENT 18 Sep '19

18 Sep '19

A potential bug was reported in the email "[BUG] atmel_ssc_dai: a possible sleep-in-atomic bug in atmel_ssc_shutdown"[1] Indeed in the function atmel_ssc_shutdown() free_irq() was called in a critical section protected by spinlock. However this spinlock is only used in atmel_ssc_shutdown() and atmel_ssc_startup() functions. After further analysis, it occurred that the call to these function are already protected by mutex used on the calling functions. Then we can remove the spinlock which will fix this bug as a side effect. Thanks to this patch the following message disappears: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:909" [1]: https://www.spinics.net/lists/alsa-devel/msg71286.html Reviewed-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Signed-off-by: Gregory CLEMENT <gregory.clement(a)bootlin.com> --- Changelog: v1 -> v2: - Removed the spinlock from the atmel_ssc_info struct - Added the Reviewed-by flag form Alex sound/soc/atmel/atmel_ssc_dai.c | 12 ++---------- sound/soc/atmel/atmel_ssc_dai.h | 1 - 2 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6f89483ac88c..786b48ae4905 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -116,19 +116,16 @@ static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { { .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, @@ -317,13 +314,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, dma_params); - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) return -EBUSY; - } + ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); return 0; } @@ -355,7 +349,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, dir_mask = 1 << dir; - spin_lock_irq(&ssc_p->lock); ssc_p->dir_mask &= ~dir_mask; if (!ssc_p->dir_mask) { if (ssc_p->initialized) { @@ -369,7 +362,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; ssc_p->forced_divider = 0; } - spin_unlock_irq(&ssc_p->lock); /* Shutdown the SSC clock. */ pr_debug("atmel_ssc_dai: Stopping clock\n"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index ae764cb541c7..3470b966e449 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -93,7 +93,6 @@ struct atmel_ssc_state { struct atmel_ssc_info { char *name; struct ssc_device *ssc; - spinlock_t lock; /* lock for dir_mask */ unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ unsigned short initialized; /* true if SSC has been initialized */ unsigned short daifmt; -- 2.23.0

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[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: Remove wrong spinlock usage
by Gregory CLEMENT 18 Sep '19

18 Sep '19

A potential bug was reported in the email "[BUG] atmel_ssc_dai: a possible sleep-in-atomic bug in atmel_ssc_shutdown"[1] Indeed in the function atmel_ssc_shutdown() free_irq() was called in a critical section protected by spinlock. However this spinlock is only used in atmel_ssc_shutdown() and atmel_ssc_startup() functions. After further analysis, it occurred that the call to these function are already protected by mutex used on the calling functions. Then we can remove the spinlock which will fix this bug as a side effect. Thanks to this patch the following message disappears: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:909" [1]: https://www.spinics.net/lists/alsa-devel/msg71286.html Signed-off-by: Gregory CLEMENT <gregory.clement(a)bootlin.com> --- sound/soc/atmel/atmel_ssc_dai.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6f89483ac88c..365957e86419 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -116,19 +116,16 @@ static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { { .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, @@ -317,13 +314,11 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, dma_params); - spin_lock_irq(&ssc_p->lock); if (ssc_p->dir_mask & dir_mask) { spin_unlock_irq(&ssc_p->lock); return -EBUSY; } ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); return 0; } @@ -355,7 +350,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, dir_mask = 1 << dir; - spin_lock_irq(&ssc_p->lock); ssc_p->dir_mask &= ~dir_mask; if (!ssc_p->dir_mask) { if (ssc_p->initialized) { @@ -369,7 +363,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; ssc_p->forced_divider = 0; } - spin_unlock_irq(&ssc_p->lock); /* Shutdown the SSC clock. */ pr_debug("atmel_ssc_dai: Stopping clock\n"); -- 2.23.0

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[alsa-devel] alsa acting up
by liebrecht＠grossmann-venter.com 18 Sep '19

18 Sep '19

See images in links. 1) "http://grossmann-venter.com/visitors/ug/alsa/alsa-pulse-181vsl_unplug.png" 2) "http://grossmann-venter.com/visitors/ug/alsa/alsa-dont recognize1818VSLpng" 1) As soon as pulseaudio is started Alsa says 1818VSL is unplugged....? 2) Clearly Presonus 1818VSL is found as lsusb output says, but alsa has no clue about it. How do I get alsa to recognise 181VSL as it is found by lsusb so it is presnet on the system. It used to work fine but alsa just stopped working with 1818VSL recently for no reason. I know this is a developers group, but please at least answer where I can get help. Alsa help is absolutely non-existing. I also posted a bug with no response in another thread. Thanks.

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