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

[alsa-devel] [PATCH][next] ASoC: amd: fix spelling mistake: "failer" -> "failure"
by Colin King 01 May '18

01 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err error message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/acp-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 9c026c4d26d4..2b9a020e0b3e 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1014,7 +1014,7 @@ static int acp_dma_new(struct snd_soc_pcm_runtime *rtd) } if (ret < 0) dev_err(component->dev, - "buffer preallocation failer error:%d\n", ret); + "buffer preallocation failure error:%d\n", ret); return ret; } -- 2.17.0

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[alsa-devel] [PATCH RFC] ASoC: dai playback_active and capture_active may be greater than 1
by Jerome Brunet 01 May '18

01 May '18

At the moment playback_active and capture_active are using only 1 bit so the maximum active count is 1. However, snd_soc_runtime_activate() may be called several time on the same dai. This happens when a dai is part of several dai_links. It is often the case for "snd-soc-dummy-dai". This is a problem if snd_soc_runtime_activate() is called an even number of times on a dai. In this case the active count overflow back to 0. As consequence, ASoC functions, such as soc_dpcm_runtime_update(), won't run correctly. Storing these usage counts on plain 'unsigned int' solves the problem. Fixes: f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Hi, I found this problem while working on ASoC support for a new platform. During a playback, using DPCM, if I changed the backend dai, nothing happened, old backend path was not pruned, new backend path was not added. Here is how it goes: 1 FE and 2 BE, all 3 using snd-soc-dummy-dai for the codec_dai (ATM). On playback start: - BE #1 activates and increments snd-soc-dummy-dai's playback_active - FE activates and increments snd-soc-dummy-dai's playback_active The playback works but snd-soc-dummy-dai's playback_active value is now 0. Through a mux kcontrol, change BE client of FE, from BE #1 to BE #2. -> trigger soc_dpcm_runtime_update(). -> checks fe's codec_dai playback_active which is zero so playback path processing is skipped, keeping the old invalid path. include/sound/soc-dai.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 8ad11669e4d8..35ebb0be5114 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -294,8 +294,8 @@ struct snd_soc_dai { struct snd_soc_dai_driver *driver; /* DAI runtime info */ - unsigned int capture_active:1; /* stream is in use */ - unsigned int playback_active:1; /* stream is in use */ + unsigned int capture_active; /* stream usage count */ + unsigned int playback_active; /* stream usage count */ unsigned int probed:1; unsigned int active; -- 2.14.3

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[alsa-devel] [PATCH v3 0/3] ASoc: TAS6424: Add support for mute, standby, and faster power on
by Jean-Jacques Hiblot 01 May '18

01 May '18

mute and standby pins are available on the codec. If they are connected, they should be managed by the driver, instead of relying on gpio hogs or on the initial state of the GPIOs. This series also includes a patch to improve the start-up time of the channels by disabling built-in DC diagnostics. Those diagnosdtics basically serve to detect : - wires shorted together - wire shorted to ground or vbat - wire disconnected This is not useful for all platforms and the addition to the startup time is quite noticeable (230ms). The diagnostics can be enabled/disabled using a dedicated ALSA control. Changes in v3: - use SOC_SINGLE_STROBE() to register the "auto diagniostics switch" control Changes in v2: - use a ALSA control to enable/disable the auto diagnostics instead of relying on a DTS property. - fixed typos - use gpios (plural) instead of gpio in the documentation of the bindings Jean-Jacques Hiblot (3): ASoC: tas6424: Add support for the standby pin ASoC: tas6424: Add support for the mute pin ASoC: tas6424: Allow disabling auto diagnostics for faster power-on .../devicetree/bindings/sound/ti,tas6424.txt | 2 + sound/soc/codecs/tas6424.c | 72 +++++++++++++++++++++- sound/soc/codecs/tas6424.h | 4 ++ 3 files changed, 75 insertions(+), 3 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2 0/6] ARM/ASoC: omap: n810 audio support
by Peter Ujfalusi 01 May '18

01 May '18

Hi, Changes since v1: - Fixed up the aic33 dts patch with missing entries needed for capture - Added comment and new widget for the non working HS Mic The verification of the sdma-pcm on OMAP2 got delayed by non working audio on n810. With the following patches audio works on n810 (both with the old omap-pcm and the new sdma-pcm). I believe the DTS and ASoC patches are fine to go via different tree, but I wanted to keep the patches together for easier testing. The series does not have any dependencies, I will rebase the sdma-pcm series on top of this and resend. Playback to Speaker, Headset is working. Capture from DMIC is working. Capture from HS mic is not working due to missing mic bias (from Retu/Vilma). Regards, Peter --- Jarkko Nikula (1): ASoC: omap: Remove OMAP_MUX dependency from Nokia N810 audio support Peter Ujfalusi (5): ARM: dts: omap2420-n810: Enable McBSP2 for audio ARM: dts: omap2420-n810: Correct the audio codec (tlv320aic33) node ASoC: omap: n810: Correct the card level dapm_route ASoC: omap: n810: Correct the cpu_dai, platform and codec name ASoC: omap: n810: HS mic is not working, add a widget for it with comment arch/arm/boot/dts/omap2420-n810.dts | 63 ++++++++++++++++++++++++++++- sound/soc/omap/Kconfig | 1 - sound/soc/omap/n810.c | 21 ++++++---- 3 files changed, 75 insertions(+), 10 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 0/2] ASoC: add mt6351 codec driver
by KaiChieh Chuang 01 May '18

01 May '18

This patch adds the MediaTek MT6351 codec driver. KaiChieh Chuang (2): ASoC: add documents for mt6351 ASoC: add mt6351 codec driver Documentation/devicetree/bindings/sound/mt6351.txt | 16 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6351.c | 1506 ++++++++++++++++++++ sound/soc/codecs/mt6351.h | 105 ++ 5 files changed, 1633 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6351.txt create mode 100644 sound/soc/codecs/mt6351.c create mode 100644 sound/soc/codecs/mt6351.h -- 2.12.5

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[alsa-devel] [PATCH v2 00/13] ALSA: dice: use cache of stream formats to generate PCM rules/constraints
by Takashi Sakamoto 01 May '18

01 May '18

Hi, This patchset updates my previous one: [alsa-devel] [PATCH 00/11] ALSA: dice: use cache of stream formats to generate PCM rules/constraints http://mailman.alsa-project.org/pipermail/alsa-devel/2018-April/134983.html Once, ALSA dice driver cached stream formats to generate PCM constraints/rules. The cache is created by changing current sampling transmission frequency step by step directly at unit probing process. This feature was dropped because it changes state of clock on target device thus it's against users' expectation[1]. However this is not necessarily convenient because reconfiguration of sampling rate is not available. TC Applied Technologies (TCAT) has extended protocol for DICE interface, a.k.a EAP (Extended Application Protocol). In this extension, drivers can retrieve information of stream formats without changing state of target unit. Fortunately, this protocol is supported widely by many models. This patchset regain the cache to generate PCM constraints/rules by using this protocol. Userspace applications can reconfigure sampling rate via ALSA PCM interface. For units which doesn't support the extension, available set of sampling rate is still restricted at current sampling transmission frequency. This is also inconvenient and this patchset introduces a solution. Structure for unit entry (struct ieee1394_device_id) has a member for unit-specific data. This patchset uses it for callback function to handle hard-coded stream format. In this time, I adopt this way for some known models produced by TC Electronic and Alesis. If users find their models still have restrictions in a point of available sampling transmission frequency, it's preferable to work for this way. Changes from v1: - rename members for the channel cache. - add minor refactoring. - fix a bug that sampling transmission frequency is not changed via ALSA PCM interface. - fix a bug that current stream formats are reused after changing sampling transmission frequency. [1] [alsa-devel] [PATCH 0/6 v2] ALSA: dice: constrain PCM substreams to current sampling transfer frequency http://mailman.alsa-project.org/pipermail/alsa-devel/2016-February/104065.h… [2] [alsa-devel] [PATCH] ALSA: dice: fix OUI for TC group http://mailman.alsa-project.org/pipermail/alsa-devel/2018-April/134804.html Takashi Sakamoto (13): ALSA: dice: add cache of stream formats ALSA: dice: add 'firewire' directory for proc nodes ALSA: dice: add proc node for stream formation ALSA: dice: cache stream formats at current mode of sampling transmission frequency ALSA: dice: add parameters of stream formats for models produced by TC Electronic ALSA: dice: add parameters of stream formats for models produced by Alesis ALSA: dice: use extended protocol to detect available stream formats ALSA: dice: use cache of stream format to check running stream ALSA: dice: add a helper function to restart all of available streams ALSA: dice: enable to change current sampling transmission frequency ALSA: dice: use stream formats to add MIDI substreams ALSA: dice: use cache for PCM constraints and rules ALSA: dice: remove local frag of force_two_pcms sound/firewire/dice/Makefile | 3 +- sound/firewire/dice/dice-alesis.c | 48 ++++++ sound/firewire/dice/dice-extension.c | 172 ++++++++++++++++++++ sound/firewire/dice/dice-midi.c | 23 +-- sound/firewire/dice/dice-pcm.c | 230 ++++++++++++++++++--------- sound/firewire/dice/dice-proc.c | 70 +++++++- sound/firewire/dice/dice-stream.c | 273 ++++++++++++++++++++++++-------- sound/firewire/dice/dice-tcelectronic.c | 100 ++++++++++++ sound/firewire/dice/dice.c | 139 ++++++++++------ sound/firewire/dice/dice.h | 17 +- 10 files changed, 867 insertions(+), 208 deletions(-) create mode 100644 sound/firewire/dice/dice-alesis.c create mode 100644 sound/firewire/dice/dice-extension.c create mode 100644 sound/firewire/dice/dice-tcelectronic.c -- 2.14.1

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[alsa-devel] [PATCH] ASoC: mediatek: preallocate pages use platform device
by KaiChieh Chuang 01 May '18

01 May '18

preallocate pages should use platform device, since we set dma mask for platform device. Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- sound/soc/mediatek/common/mtk-afe-platform-driver.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 53215b52e4f2..f8a06709f76d 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -64,14 +64,14 @@ static const struct snd_pcm_ops mtk_afe_pcm_ops = { static int mtk_afe_pcm_new(struct snd_soc_pcm_runtime *rtd) { size_t size; - struct snd_card *card = rtd->card->snd_card; struct snd_pcm *pcm = rtd->pcm; struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); size = afe->mtk_afe_hardware->buffer_bytes_max; return snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - card->dev, size, size); + rtd->platform->dev, + size, size); } static void mtk_afe_pcm_free(struct snd_pcm *pcm) -- 2.12.5

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