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

[alsa-devel] [GIT PULL] ASoC fixes for v5.1
by Mark Brown 11 Apr '19

11 Apr '19

The following changes since commit 9e98c678c2d6ae3a17cb2de55d17f69dddaa231b: Linux 5.1-rc1 (2019-03-17 14:22:26 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.1-rc4 for you to fetch changes up to ac71317e6be01812cc0c54d8be6d3c1139c8380b: ASoC: wcd9335: Fix missing regmap requirement (2019-04-10 17:07:59 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.1 A few core fixes along with the driver specific ones, mainly fixing small issues that only affect x86 platforms for various reasons (their unusual machine enumeration mechanisms mainly, plus a fix for error handling in topology). There's some of the driver fixes that look larger than they are, like the hdmi-codec changes which resulted in an indentation change, and most of the other large changes are for new drivers like the STM32 changes. ---------------------------------------------------------------- Annaliese McDermond (2): ASoC: tlv320aic32x4: Fix Common Pins ASoC: tlv320aic32x4: Change author's name Arnaud Pouliquen (1): ASoC: stm32: fix sai driver name initialisation Charles Keepax (6): ASoC: wm_adsp: Correct handling of compressed streams that restart ASoC: wm_adsp: Correct error messages in wm_adsp_buffer_get_error ASoC: wm_adsp: Add locking to wm_adsp2_bus_error ASoC: wm_adsp: Shutdown any compressed streams on DSP watchdog timeout ASoC: wm_adsp: Check for buffer in trigger stop ASoC: cs35l35: Disable regulators on driver removal Daniel Mack (1): ASoC: cs4270: Set auto-increment bit for register writes Guenter Roeck (1): ASoC: intel: Fix crash at suspend/resume after failed codec registration Gustavo A. R. Silva (1): ASoC: ab8500: Mark expected switch fall-through Hans de Goede (1): ASoC: Intel: cht_bsw_max98090_ti: Enable codec clock once and keep it enabled Jenny TC (1): ASoC: Intel: Skylake: enable S24_LE format support Jerome Brunet (1): ASoC: dpcm: skip missing substream while applying symmetry Jiada Wang (2): ASoC: rsnd: src: Avoid a potential deadlock ASoC: rsnd: src: fix compiler warnings John Hsu (2): ASoC: nau8824: fix the issue of the widget with prefix name ASoC: nau8810: fix the issue of widget with prefixed name Jonathan Hunter (1): ASoC: soc-core: Fix probe deferral following prelink failure KaiChieh Chuang (2): ASoC: mediatek: btcvsd add loopback ASoC: dpcm: prevent snd_soc_dpcm use after free Kuninori Morimoto (2): ASoC: audio-graph-card: don't select DPCM via audio-graph-card ASoC: simple-card: don't select DPCM via simple-audio-card Marc Gonzalez (1): ASoC: wcd9335: Fix missing regmap requirement Mark Brown (2): Merge branch 'for-5.0' of https://git.kernel.org/.../broonie/sound into asoc-5.1 Merge tag 'v5.1-rc1' into asoc-5.1 Olivier Moysan (9): ASoC: stm32: sai: fix iec958 controls indexation ASoC: stm32: sai: fix exposed capabilities in spdif mode ASoC: stm32: sai: fix race condition in irq handler ASoC: stm32: sai: fix oversampling mode ASoC: stm32: sai: fix set_sync service ASoC: stm32: i2s: fix registers declaration in regmap ASoC: stm32: dfsdm: manage multiple prepare ASoC: stm32: dfsdm: fix debugfs warnings on entry creation ASoC: stm32: sai: fix master clock management Pankaj Bharadiya (1): ASoC: dapm: Fix NULL pointer dereference in snd_soc_dapm_free_kcontrol Philipp Puschmann (1): ASoC: tlv320aic3x: fix reset gpio reference counting Rander Wang (3): ASoC:soc-pcm:fix a codec fixup issue in TDM case ASoC:hdac_hda:use correct format to setup hda codec ASoC:intel:skl:fix a simultaneous playback & capture issue on hda platform Ranjani Sridharan (6): ASoC: dapm: set power_check callback for widgets that shouldnt be always on ASoC: intel: skylake: add remove() callback for component driver ASoC: topology: Use the correct dobj to free enum control values and texts ASoC: core: conditionally increase module refcount on component open ASoC: pcm: update module refcount if module_get_upon_open is set ASoC: pcm: fix error handling when try_module_get() fails. Russell King (2): ASoC: hdmi-codec: fix S/PDIF DAI ASoC: hdmi-codec: avoid limiting params->msbits in hw_params() S.j. Wang (2): ASoC: fsl_asrc: add constraint for the asrc of older version ASoC: fsl_esai: fix channel swap issue when stream starts Shuming Fan (3): ASoC: rt5682: Check JD status when system resume ASoC: rt5682: fix jack type detection issue ASoC: rt5682: recording has no sound after booting Sugar Zhang (2): ASoC: rockchip: pdm: fix regmap_ops hang issue ASoC: rockchip: pdm: change dma burst to 8 Sylwester Nawrocki (2): ASoC: samsung: i2s: Fix DAPM routes for capture stream ASoC: samsung: odroid: Fix clock configuration for 44100 sample rate Tzung-Bi Shih (2): ASoC: mediatek: mt8183: skip for i2s5 in mck_disable ASoC: Intel: kbl: fix wrong number of channels include/sound/soc.h | 11 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/ab8500-codec.c | 1 + sound/soc/codecs/cs35l35.c | 11 ++ sound/soc/codecs/cs4270.c | 1 + sound/soc/codecs/hdac_hda.c | 53 ++++++--- sound/soc/codecs/hdac_hda.h | 1 + sound/soc/codecs/hdmi-codec.c | 121 ++++++++++----------- sound/soc/codecs/nau8810.c | 4 +- sound/soc/codecs/nau8824.c | 46 ++++++-- sound/soc/codecs/rt5682.c | 56 +++++----- sound/soc/codecs/tlv320aic32x4-i2c.c | 4 +- sound/soc/codecs/tlv320aic32x4-spi.c | 4 +- sound/soc/codecs/tlv320aic32x4.c | 2 + sound/soc/codecs/tlv320aic3x.c | 4 +- sound/soc/codecs/wm_adsp.c | 40 +++++-- sound/soc/codecs/wm_adsp.h | 1 + sound/soc/fsl/fsl_asrc.c | 14 +++ sound/soc/fsl/fsl_esai.c | 47 ++++++-- sound/soc/generic/audio-graph-card.c | 11 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 8 ++ sound/soc/intel/boards/cht_bsw_max98090_ti.c | 47 +++++++- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/skylake/skl-messages.c | 1 + sound/soc/intel/skylake/skl-pcm.c | 28 ++++- sound/soc/mediatek/common/mtk-btcvsd.c | 69 +++++++++++- sound/soc/mediatek/mt8183/mt8183-afe-clk.c | 4 + sound/soc/rockchip/rockchip_pdm.c | 4 +- sound/soc/samsung/i2s.c | 10 +- sound/soc/samsung/odroid.c | 4 +- sound/soc/sh/rcar/core.c | 2 + sound/soc/sh/rcar/rsnd.h | 5 + sound/soc/sh/rcar/src.c | 21 ++-- sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 11 ++ sound/soc/soc-pcm.c | 66 +++++++++-- sound/soc/soc-topology.c | 7 +- sound/soc/stm/stm32_adfsdm.c | 38 ++++++- sound/soc/stm/stm32_i2s.c | 3 +- sound/soc/stm/stm32_sai.c | 8 +- sound/soc/stm/stm32_sai_sub.c | 114 ++++++++++++++----- 42 files changed, 668 insertions(+), 235 deletions(-)

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[alsa-devel] [PATCH 00/12] ASoC: intel: add device_link to HDMI audio
by libin.yang＠intel.com 11 Apr '19

11 Apr '19

From: Libin Yang <libin.yang(a)intel.com> This patchset add the device_link between the machine devices of intel boards and HDMI audio codec. This can make sure that display audio power domain is always turned on before operating on the HDMI audio codecs. patch 2 adds the helper functions in a new created header file. However skl_hda_dsp_generic doesn't use these helper functions because skl_hda_dsp_generic is a special driver, the add link and delete link operations are in different source code files. If it includes the header file, there is compiling warning. Libin Yang (12): ASoC: intel: skl_hda_dsp_generic: add device_link to HDMI audio ASoC: intel: boards: define some general functions for hdac_hdmi ASoC: intel: bxt_da7219_max98357a: add device_link to HDMI audio ASoC: intel: bxt_rt298: add device_link to HDMI audio ASoC: intel: glk_rt5682_max98357a: add device_link to HDMI audio ASoC: intel: kbl_da7219_max98357a: add device_link to HDMI audio ASoC: intel: kbl_da7219_max98927: add device_link to HDMI audio ASoC: intel: kbl_rt5660: add device_link to HDMI audio ASoC: intel: kbl_rt5663_max98927 add device_link to HDMI audio ASoC: intel: kbl_rt5663_rt5514_max98927 add device_link to HDMI audio ASoC: intel: skl_nau88l25_max98357a add device_link to HDMI audio ASoC: intel: skl_nau88l25_ssm4567 add device_link to HDMI audio sound/soc/intel/boards/bxt_da7219_max98357a.c | 14 +++++++ sound/soc/intel/boards/bxt_rt298.c | 14 ++++++- sound/soc/intel/boards/glk_rt5682_max98357a.c | 14 +++++++ sound/soc/intel/boards/hdac_hdmi_common.h | 46 ++++++++++++++++++++++ sound/soc/intel/boards/kbl_da7219_max98357a.c | 14 +++++++ sound/soc/intel/boards/kbl_da7219_max98927.c | 14 +++++++ sound/soc/intel/boards/kbl_rt5660.c | 14 ++++++- sound/soc/intel/boards/kbl_rt5663_max98927.c | 14 ++++++- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 14 ++++++- sound/soc/intel/boards/skl_hda_dsp_common.c | 22 +++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 12 ++++++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 36 ++++++++++++++++- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 36 ++++++++++++++++- 14 files changed, 259 insertions(+), 6 deletions(-) create mode 100644 sound/soc/intel/boards/hdac_hdmi_common.h -- 2.7.4

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Re: [alsa-devel] power state in power save mode
by Takashi Iwai 11 Apr '19

11 Apr '19

On Thu, 11 Apr 2019 13:39:21 +0200, Kailang wrote: > > Hi Takashi, > > I also checked the value by simply printk. > I put printk in alc256_init() and alc256_shutup(). > Without power save, i could see the expected value at suspend and resume. > > With power save, alc256_init() will run after music play, it will print out value 0x0 at this time. > alc256_shutup() already run at count down time value of power_save = ?. > So, alc256_shutup will not run again when system suspend. power state value will also not print out during suspend. > Is this the expectation behavior? Yes, the same suspend/resume path is used for the codec driver for both runtime and system suspend/resume. For the runtime PM, the callbacks are called with power_state=0. > I test add your patch. > Then test with power save option. Wait time up, codec go to power save mode(run alc256_shutup()). > Let system to suspend.(it will not run alc256_shutup() again) > Then resume it. Play a short wave file. alc256_init() will run after music play, it will value 0x10 at this time. > > I think alc256_shutup() will not update power state when it with power save option. > If I want to use alc256_shutup() to check power state then do some thing, I think it's impossible. Right? When the system was already in the runtime suspended state, it won't trigger the codec suspend again. With the recent code (after the commit b5a236c175b0), we enforce the resume even if it wasn't used at the suspend time. So for the resume, it's always woken up with the proper power_state when the system is resumed via S3/S4 (not via runtime PM). Takashi > > BR, > Kailang > > ________________________________________ > 從: Takashi Iwai [tiwai(a)suse.de] > 寄件日期: 2019年4月10日下午 11:52 > 至: Kailang > 副本: (alsa-devel(a)alsa-project.org) > 主旨: Re: power state in power save mode > > On Wed, 10 Apr 2019 05:00:31 +0200, > Kailang wrote: > > > > > -----Original Message----- > > > From: Takashi Iwai <tiwai(a)suse.de> > > > Sent: Tuesday, April 9, 2019 6:15 PM > > > To: Kailang <kailang(a)realtek.com> > > > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > > > Subject: Re: power state in power save mode > > > > > > On Tue, 09 Apr 2019 11:54:27 +0200, > > > Kailang wrote: > > > > > > > > Hi Takashi, > > > > > > > > https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit > > > > /?h=for-next&id=98081ca62cbac31fb0f7efaf90b2e7384ce22257 > > > > > > > > Use upper patch to print power state in function alc256_init() and > > > alc256_shutup(). > > > > But when kernel turn on power_save = 1. (echo 1 > > > > > /sys/module/snd_hda_intel/parameters/power_save) > > > > > > > > The print power state value always be 0x0. > > > > > > That's expected, it means PM_EVENT_ON. > > > The power_state value is changed only during the suspend and resume. > > > Once after returning from resume, it's set to PM_EVENT_ON. > > > > I also test suspend and resume. It also show 0x0. > > If it turn off power save, power state value will normally. > > What exactly did you see? I checked the value by simply printk and it > behaved as expected, both with and without power save. > > > > > > > > ( If it turn off power save, power state value will normally) > > > > > > What does it mean exactly...? > > > > > > Actually one missing piece is to set PMSG_ON at the device initialization. A > > > patch like below. > > > > Add this patch, the power state value will show on suspend and resume. Right? > > I will test it later. > > This won't change anything about the suspend/resume behavior, but it's > just for some consistency in case the driver refers to the state > before entering the suspend/resume callback. > > > Takashi > > > > > > > > > > > thanks, > > > > > > Takashi > > > > > > --- a/sound/pci/hda/hda_codec.c > > > +++ b/sound/pci/hda/hda_codec.c > > > @@ -969,6 +969,7 @@ int snd_hda_codec_device_new(struct hda_bus *bus, > > > struct snd_card *card, > > > > > > /* power-up all before initialization */ > > > hda_set_power_state(codec, AC_PWRST_D0); > > > + codec->core.dev.power.power_state = PMSG_ON; > > > > > > snd_hda_codec_proc_new(codec); > > > > > > > > > ------Please consider the environment before printing this e-mail. > >

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[alsa-devel] [PATCH V4] ASoC: fsl_esai: Fix missing break in switch statement
by S.j. Wang 11 Apr '19

11 Apr '19

case ESAI_HCKT_EXTAL and case ESAI_HCKR_EXTAL should be independent of each other, so replace fall-through with break. Fixes: 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: <stable(a)vger.kernel.org> --- Change in v4 - Add Acked-by and cc stable - change the subject sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 3623aa9a6f2e..15202a637197 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -251,7 +251,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, break; case ESAI_HCKT_EXTAL: ecr |= ESAI_ECR_ETI; - /* fall through */ + break; case ESAI_HCKR_EXTAL: ecr |= ESAI_ECR_ERI; break; -- 1.9.1

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[alsa-devel] [PATCH V2] ASoC: fsl_esai: replace fall-through with break
by S.j. Wang 11 Apr '19

11 Apr '19

case ESAI_HCKT_EXTAL and case ESAI_HCKR_EXTAL should be independent of each other, so replace fall-through with break. Fixes: 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v2 - fix the fixes tag. sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7410bbfd2af..bad0dfed6b68 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -251,7 +251,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, break; case ESAI_HCKT_EXTAL: ecr |= ESAI_ECR_ETI; - /* fall through */ + break; case ESAI_HCKR_EXTAL: ecr |= esai_priv->synchronous ? ESAI_ECR_ETI : ESAI_ECR_ERI; break; -- 1.9.1

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[alsa-devel] [PATCH V3] ASoC: fsl_esai: Fix missing break in switch statement
by S.j. Wang 11 Apr '19

11 Apr '19

case ESAI_HCKT_EXTAL and case ESAI_HCKR_EXTAL should be independent of each other, so replace fall-through with break. Fixes: 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Cc: <stable(a)vger.kernel.org> --- changes in v3 - add cc stable - change the subject sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 3623aa9a6f2e..15202a637197 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -251,7 +251,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, break; case ESAI_HCKT_EXTAL: ecr |= ESAI_ECR_ETI; - /* fall through */ + break; case ESAI_HCKR_EXTAL: ecr |= ESAI_ECR_ERI; break; -- 1.9.1

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[alsa-devel] ASoC fsl_sai? Output stuck high causes loud pop
by Cameron Kellough 10 Apr '19

10 Apr '19

Hi ASoC developers, I'm working with kernel 4.9.88 on an NXP IMX6ULL based PicoSOM from Technexion. We have developed our own integration of this CPU with an audio codec and a DSP. I'm hearing a loud pop at the end of playback for sounds and after a lot of tracing, I think there may be a problem in or near the fsl_sai driver. I suspect that the SAI transmitter is not being disabled at the end of playback as the bus appears to be driven with all ones after the end of playback. Briefly, we have the IMX6ULL configured as an I2S slave to a DSP chip. In this case, the DSP is continuously providing BCLK and LRCLK to the IMX6ULL and the connected codec. When no sound is playing and I examine the output of the Serial Data Out line from the IMX6ULL with an o-scope, the values on it encode all high bits instead of all low bits. The popping noise we hear seems to be the bus going from a relatively low value at the end of a played sound to an infinite length set of samples of maximum volume. Based on having set the pad control registers to no pullup, one would expect the I2S bus to be low when no sound is being transmitted. While the exact problem above involves a custom setup, a popping can be observed using the Technexion IMX6ULL on the PI baseboard and playing a sound with aplay. Some evidence on the NXP forum suggests the problem also exists using the IMX6. I'm sorry this isn't a lot to go on. I'm happy to provide more information if anyone is interested in this issue. --Cameron Kellough

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[alsa-devel] [PATCH] ASoC: fsl_asrc: replace the process_option table with function
by S.j. Wang 10 Apr '19

10 Apr '19

The table is not flexible if supported sample rate is not in the table, so use a function to replace it. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 73 +++++++++++++++++++++++++++++++++++------------- 1 file changed, 53 insertions(+), 20 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0b937924d2e4..a57c6c829060 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -26,24 +26,6 @@ #define pair_dbg(fmt, ...) \ dev_dbg(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) -/* Sample rates are aligned with that defined in pcm.h file */ -static const u8 process_option[][12][2] = { - /* 8kHz 11.025kHz 16kHz 22.05kHz 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */ - {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */ - {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */ - {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */ - {{1, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */ - {{1, 2}, {1, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */ - {{1, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */ - {{2, 2}, {2, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */ - {{2, 2}, {2, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */ - {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */ - {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */ - {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */ - {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */ - {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */ -}; - /* Corresponding to process_option */ static int supported_input_rate[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, @@ -79,6 +61,49 @@ static unsigned char *clk_map[2]; +static int proc_autosel(int Fsin, int Fsout, int *pre_proc, int *post_proc) +{ + bool det_out_op2_cond; + bool det_out_op0_cond; + + det_out_op2_cond = (((Fsin * 15 > Fsout * 16) & (Fsout < 56000)) | + ((Fsin > 56000) & (Fsout < 56000))); + det_out_op0_cond = (Fsin * 23 < Fsout * 8); + + /* + * Not supported case: Tsout>16.125*Tsin, and Tsout>8.125*Tsin. + */ + if (Fsin * 8 > 129 * Fsout) + *pre_proc = 5; + else if (Fsin * 8 > 65 * Fsout) + *pre_proc = 4; + else if (Fsin * 8 > 33 * Fsout) + *pre_proc = 2; + else if (Fsin * 8 > 15 * Fsout) { + if (Fsin > 152000) + *pre_proc = 2; + else + *pre_proc = 1; + } else if (Fsin < 76000) + *pre_proc = 0; + else if (Fsin > 152000) + *pre_proc = 2; + else + *pre_proc = 1; + + if (det_out_op2_cond) + *post_proc = 2; + else if (det_out_op0_cond) + *post_proc = 0; + else + *post_proc = 1; + + if (*pre_proc == 4 || *pre_proc == 5) + return -EINVAL; + + return 0; +} + /** * Request ASRC pair * @@ -239,8 +264,10 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) u32 inrate, outrate, indiv, outdiv; u32 clk_index[2], div[2]; int in, out, channels; + int pre_proc, post_proc; struct clk *clk; bool ideal; + int ret; if (!config) { pair_err("invalid pair config\n"); @@ -377,11 +404,17 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) ASRCTR_IDRi_MASK(index) | ASRCTR_USRi_MASK(index), ASRCTR_IDR(index) | ASRCTR_USR(index)); + ret = proc_autosel(inrate, outrate, &pre_proc, &post_proc); + if (ret) { + pair_err("No supported pre-processing options\n"); + return ret; + } + /* Apply configurations for pre- and post-processing */ regmap_update_bits(asrc_priv->regmap, REG_ASRCFG, ASRCFG_PREMODi_MASK(index) | ASRCFG_POSTMODi_MASK(index), - ASRCFG_PREMOD(index, process_option[in][out][0]) | - ASRCFG_POSTMOD(index, process_option[in][out][1])); + ASRCFG_PREMOD(index, pre_proc) | + ASRCFG_POSTMOD(index, post_proc)); return fsl_asrc_set_ideal_ratio(pair, inrate, outrate); } -- 1.9.1

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[alsa-devel] [PATCH v2 0/3] ASoC: fsl: audmix: remove "model" attribute and fix ref leaks
by Viorel Suman 10 Apr '19

10 Apr '19

The latest audmix patch-set (v5) had the "model" attribute removed as requested by Nicolin Chen, but looks like (v4) version of DAI driver reached "for-next" branch - fix this by removing "model" attribute. Asside of this fix object reference leaks in machine probe reported by Julia Lawall. Viorel Suman (3): ASoC: fsl_audmix: remove "model" attribute dt-bindings: fsl,audmix: remove "model" attribute ASoC: imx-audmix: fix object reference leaks in probe Changes since V1: a) Removed "model" attribute from dt-bindings documentation b) Adressed Daniel's comments .../devicetree/bindings/sound/fsl,audmix.txt | 4 -- sound/soc/fsl/fsl_audmix.c | 43 ++++++++++++---------- sound/soc/fsl/imx-audmix.c | 4 ++ 3 files changed, 27 insertions(+), 24 deletions(-) -- 2.7.4

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[alsa-devel] power state in power save mode
by Kailang 10 Apr '19

10 Apr '19

Hi Takashi, https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=f… Use upper patch to print power state in function alc256_init() and alc256_shutup(). But when kernel turn on power_save = 1. (echo 1 > /sys/module/snd_hda_intel/parameters/power_save) The print power state value always be 0x0. ( If it turn off power save, power state value will normally) BR, Kailang

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