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

[alsa-devel] Question about soc_bind_aux_dev()
by Kuninori Morimoto 23 May '19

23 May '19

Hi ALSA ML Do you know why snd_soc_aux_dev has both name and codec_name ?? struct snd_soc_aux_dev { => const char *name; /* Codec name */ ... => const char *codec_name; ... }; For example, samsung's speyside.c has both name and codec_name static struct snd_soc_aux_dev speyside_aux_dev[] = { { => .name = "wm9081", => .codec_name = "wm9081.1-006c", ... }, }; In my research, there is no driver which is using .name only. All drivers are having codec_of_node, or .codec_name Now, in the soc_bind_aux_dev(), it checks .codec_name or codec_of_node first. This means, there is zero chance to use .name ? static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { ... => if (aux_dev->codec_of_node || aux_dev->codec_name) { ... component = soc_find_component(...); ... => } else if (aux_dev->name) { ... component = soc_find_component(...); ... } ... } Why .name is needed ?? Thank you for your help !! Best regards --- Kuninori Morimoto

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[alsa-devel] Add polling mode support on snd_hdac_bus_get_response(),
by Liao, Bard 23 May '19

23 May '19

Hi Takashi, We would like to add polling mode support on snd_hdac_bus_get_response(), and use it as a workaround for some specific platforms. Just like azx_rirb_get_response() and azx_create() if (chip->polling_mode || do_poll) snd_hdac_bus_update_rirb(bus); and if (IS_CFL(pci)) chip->polling_mode = 1; However struct azx is a local variable and other driver may have their own chip structure. Like struct skl and struct sof_intel_hda_dev. So I am thinking does it make sense to add the polling_mode in struct hdac_bus? So we can get the polling_mode directly from bus->polling_mode. Or it makes more sense to create a global chip struct similar to struct azx for snd_hdac_bus_get_response()? Thanks, Bard

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[alsa-devel] [PATCH - alsa-lib 0/4] Add pcm_file input file support for MMAP
by Adam Miartus 22 May '19

22 May '19

implement overwriting read buffer with input file in case of RW_NONINTERLEAVED and MMAP access modes for alsa file plugin first two patches introduce a helper function for copying data from input file to readi and readn buffers for user. to facilitate this snd_pcm_areas_from_buf and snd_pcm_areas_from_bufs with snd_pcm_areas_copy is used. Previous readi behaviour in corner case of input file EOF is changed (more in patch comment). Patch 3,4 introduce new api callback mmap_begin to ease overwritting mmaped buffer in case of MMAP access. Adam Miartus (4): pcm_file: add support for infile reading in non interleaved mode pcm_file: use snd_pcm_file_areas_read_infile for readi pcm: add mmap_begin callback to snd_pcm_fast_ops_t api pcm_file: add infile read support for mmap mode src/pcm/pcm.c | 6 +++ src/pcm/pcm_dmix.c | 1 + src/pcm/pcm_dshare.c | 1 + src/pcm/pcm_dsnoop.c | 1 + src/pcm/pcm_file.c | 125 ++++++++++++++++++++++++++++++++++++++---------- src/pcm/pcm_generic.h | 1 - src/pcm/pcm_hooks.c | 1 + src/pcm/pcm_hw.c | 2 + src/pcm/pcm_ioplug.c | 1 + src/pcm/pcm_local.h | 1 + src/pcm/pcm_meter.c | 1 + src/pcm/pcm_mmap_emul.c | 1 + src/pcm/pcm_multi.c | 1 + src/pcm/pcm_null.c | 1 + src/pcm/pcm_plugin.c | 1 + src/pcm/pcm_rate.c | 1 + src/pcm/pcm_share.c | 1 + src/pcm/pcm_shm.c | 1 + 18 files changed, 122 insertions(+), 26 deletions(-) -- 2.7.4

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[alsa-devel] Change Dell headset mode rule
by Kailang 22 May '19

22 May '19

Hi Takashi, In Dell headset mode, it will recheck during runtime resume when headset was plugged. This patch will move check headset type on unplug and system resume. Many Thanks. BR, Kailang

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[alsa-devel] [PATCH 0/4] ALSA: firewire-lib: use IT context header to compute cycle count
by Takashi Sakamoto 22 May '19

22 May '19

Hi, In Linux firewire subsystem, the series of timestamp for each packets is passed as argument to callback of IT context. The timestamp comes from descriptor of 1394 OHCI IT context and represents the cycle to process queued packet. In current implementation of ALSA IEC 61883-1/6 packet streaming engine, the timestamp comes from calculation with the last packet to generate the callback, thus it's not per-packet information. For simplicity, this patchset uses the header to compute the cycle for packet to be going to be queued. Additionally, this patchset adds some minor refactoring. Takashi Sakamoto (4): ALSA: firewire-lib: use union for directional parameters ALSA: firewire-lib: add helper function to cancel context inner callback handler ALSA: firewire-lib: obsolete macro for header of IT context ALSA: firewire-lib: use IT context header to compute cycle count for scheduling packet sound/firewire/amdtp-am824.c | 2 +- sound/firewire/amdtp-stream.c | 181 +++++++++++--------- sound/firewire/amdtp-stream.h | 35 ++-- sound/firewire/digi00x/amdtp-dot.c | 2 +- sound/firewire/fireworks/fireworks_stream.c | 4 +- sound/firewire/motu/amdtp-motu.c | 2 +- sound/firewire/tascam/amdtp-tascam.c | 2 +- 7 files changed, 135 insertions(+), 93 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH v2] ASoC: soc-pcm: fixup try_module_get()/module_put() timing
by Kuninori Morimoto 22 May '19

22 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc_pcm_components_open/close() try to call try_module_get()/module_put() based on component->driver->module_get_upon_open. Here, the purpose why we need to call these functions are to checking module reference. Thus, we need to call try_module_open() even though it doesn't have .open callback. The same reason, we need to call module_put() even though it doesn't have .close This patch calls try_module_get()/module_put() regardless of .open/.close Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - merge both try_module_get()/module_put() patch into 1 sound/soc/soc-pcm.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7737e00..e24eab3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -440,10 +440,6 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream, component = rtdcom->component; *last = component; - if (!component->driver->ops || - !component->driver->ops->open) - continue; - if (component->driver->module_get_upon_open && !try_module_get(component->dev->driver->owner)) { dev_err(component->dev, @@ -452,6 +448,10 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream, return -ENODEV; } + if (!component->driver->ops || + !component->driver->ops->open) + continue; + ret = component->driver->ops->open(substream); if (ret < 0) { dev_err(component->dev, @@ -477,11 +477,9 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream, if (component == last) break; - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); + if (component->driver->ops && + component->driver->ops->close) + component->driver->ops->close(substream); if (component->driver->module_get_upon_open) module_put(component->dev->driver->owner); -- 2.7.4

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[alsa-devel] [PATCH RESEND V6 0/3] Support more sample rate in asrc
by S.j. Wang 22 May '19

22 May '19

Support more sample rate in asrc Shengjiu Wang (3): ASoC: fsl_asrc: Fix the issue about unsupported rate ASoC: fsl_asrc: replace the process_option table with function ASoC: fsl_asrc: Unify the supported input and output rate Changes in RESEND V6 - change the Content-Transfer-Encoding to "quoted-printable", for - "base64" can't be applied Changes in v6 - add acked-by - fixed minor issue according to comments in v5 Changes in v5 - fix the [1/24, 8] - move fsl_asrc_sel_proc before setting Changes in v4 - add patch to Fix the [8kHz, 30kHz] open set issue. Changes in v3 - remove FSL_ASRC_RATES - refine fsl_asrc_sel_proc according to comments Changes in v2 - add more comments in code - add commit "Unify the supported input and output rate" sound/soc/fsl/fsl_asrc.c | 105 ++++++++++++++++++++++++++------------- 1 file changed, 71 insertions(+), 34 deletions(-) -- 2.21.0

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[alsa-devel] Applied "ASoC: rename functions that pollute the simple_xxx namespace" to the asoc tree
by Mark Brown 21 May '19

21 May '19

The patch ASoC: rename functions that pollute the simple_xxx namespace has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.3 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b0a821daf0d04e5a8ae99829e24f2fe538f25763 Mon Sep 17 00:00:00 2001 From: Amir Goldstein <amir73il(a)gmail.com> Date: Thu, 16 May 2019 13:26:28 +0300 Subject: [PATCH] ASoC: rename functions that pollute the simple_xxx namespace include/linux/fs.h defines a bunch of simple fs helpers, (e.g. simple_rename) and we intend to add an fs helper named simple_remove. Rename the ASoC driver static functions, so they will not collide with the upcoming fs helper function name. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Amir Goldstein <amir73il(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 9b568f578bcd..d16e894fce2b 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -607,7 +607,7 @@ static int simple_soc_probe(struct snd_soc_card *card) return 0; } -static int simple_probe(struct platform_device *pdev) +static int asoc_simple_probe(struct platform_device *pdev) { struct asoc_simple_priv *priv; struct device *dev = &pdev->dev; @@ -705,7 +705,7 @@ static int simple_probe(struct platform_device *pdev) return ret; } -static int simple_remove(struct platform_device *pdev) +static int asoc_simple_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); @@ -726,8 +726,8 @@ static struct platform_driver asoc_simple_card = { .pm = &snd_soc_pm_ops, .of_match_table = simple_of_match, }, - .probe = simple_probe, - .remove = simple_remove, + .probe = asoc_simple_probe, + .remove = asoc_simple_remove, }; module_platform_driver(asoc_simple_card); -- 2.20.1

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[alsa-devel] [PATCH] ASoC: rt5677-spi: Handle over reading when flipping bytes
by Curtis Malainey 21 May '19

21 May '19

There is a case when a we want to read a large number of bytes that require a burst but is not a multiple of the word size (8). When this happens rt5677_spi_reverse will run off the end of the buffer. The solution is to tell spi_reverse the actual size of the destination and stop if we reach it even if we have data left that we read. Cc: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 84b6bd8b50e1..a4dfa0345c6e 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -101,7 +101,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) u32 word_size = min_t(u32, dstlen, 8); for (w = 0; w < dstlen; w += word_size) { - for (i = 0; i < word_size; i++) { + for (i = 0; i < word_size && i + w < dstlen; i++) { si = w + word_size - i - 1; dst[w + i] = si < srclen ? src[si] : 0; } @@ -152,8 +152,9 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) status |= spi_sync(g_spi, &m); mutex_unlock(&spi_mutex); + /* Copy data back to caller buffer */ - rt5677_spi_reverse(cb + offset, t[1].len, body, t[1].len); + rt5677_spi_reverse(cb + offset, len - offset, body, t[1].len); } return status; } -- 2.21.0.1020.gf2820cf01a-goog

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[alsa-devel] [PATCH][RFC] ASoC: soc-dpm: fixup DAI active unbalance
by Kuninori Morimoto 21 May '19

21 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_dai_link_event() is updating snd_soc_dai :: active, but it is unbalance. It counts up if it has startup callback. case SND_SOC_DAPM_PRE_PMU: ... snd_soc_dapm_widget_for_each_source_path(w, path) { ... if (source->driver->ops->startup) { ... => source->active++; } ... } ... But, always counts down case SND_SOC_DAPM_PRE_PMD: ... snd_soc_dapm_widget_for_each_source_path(w, path) { ... => source->active--; ... } This patch always counts up when SND_SOC_DAPM_PRE_PMD. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark, Liam I think this is bug, but I can't confirm it, because my driver need to have .startup. Thus, I added [RFC] on this patch. I'm happy if someone can confirm it. sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c864502..147ad9d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3828,8 +3828,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret); goto out; } - source->active++; } + source->active++; ret = soc_dai_hw_params(&substream, params, source); if (ret < 0) goto out; @@ -3850,8 +3850,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret); goto out; } - sink->active++; } + sink->active++; ret = soc_dai_hw_params(&substream, params, sink); if (ret < 0) goto out; -- 2.7.4

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