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

[alsa-devel] fix memory leak in snd_pcm_set_chmap
by GitHub pull_request - opened 15 Sep '19

15 Sep '19

alsa-project/alsa-lib pull request #10 was opened from conradjones: asan detects this as a memory leak. snd_pcm_get_chmap clearly states that the caller is required to free the returned value, i cannot see it stored anywhere so cannot see how it could be freed. Request URL : https://github.com/alsa-project/alsa-lib/pull/10 Patch URL : https://github.com/alsa-project/alsa-lib/pull/10.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH] ALSA: usb-audio: Add Hiby device family to quirks for native DSD support
by ilya.pshonkin＠netforce.ua 15 Sep '19

15 Sep '19

From: Sudokamikaze <sudokamikaze(a)protonmail.com> This patch adds quirk VID ID for Hiby portable players family with native DSD playback support Signed-off-by: Sudokamikaze <sudokamikaze(a)protonmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 78858918cbc1..64a8d73972e3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1655,6 +1655,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ case 0x2ab6: /* T+A devices */ + case 0xc502: /* HiBy devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.23.0

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[alsa-devel] Jack detection in Lenovo OneLink+ and other docks
by Benjamin Berg 15 Sep '19

15 Sep '19

Hi, I have a Lenovo OneLink+ and TB3 docks here. The USB audio adapter on these docks work fine. However, Jack detection is not functional unfortunately which is somewhat annoying. There are HID devices available on the docking stations which seem to report the jack sensing information. Assuming we can decode this information correctly, does anyone have an idea how this can be hooked up so that the information will be available on the audio device itself? Benjamin The OneLink+ dock exposes the following interfaces: |__ Port 1: Dev 90, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 3: Dev 92, If 1, Class=CDC Data, Driver=cdc_ether, 480M |__ Port 3: Dev 92, If 0, Class=Communications, Driver=cdc_ether, 480M |__ Port 4: Dev 94, If 2, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 4: Dev 94, If 0, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 4: Dev 94, If 3, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 94, If 1, Class=Audio, Driver=snd-usb-audio, 12M Similar on the TB 3 Dock from Lenovo: /: Bus 05.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/4p, 480M |__ Port 1: Dev 2, If 0, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 2, If 1, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 2, If 2, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 2, If 3, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 3, If 0, Class=Human Interface Device, Driver=usbhid, 12M On each of these devices, plugging/unplugging will fire various HID events, seemingly with some sensing information. On the OneLink Plus: D: 0 R: 69 05 0c 09 01 a1 01 85 01 15 00 25 01 09 e9 09 ea 75 01 95 02 81 42 09 e2 95 01 81 06 06 01 ff 09 01 95 09 81 06 05 0c 09 cd 09 b7 09 b5 09 b6 95 04 81 06 06 07 ff 09 01 75 08 95 0d 81 06 09 00 95 0f 91 02 c0 N: C-Media Electronics Inc. ThinkPad OneLink Plus Dock Audio P: usb-0000:00:14.0-4.1.4/input3 I: 3 17ef 3055 # Plug Headphone D: 0 E: 0.000000 16 01 00 06 00 00 20 83 ab 08 00 08 c0 10 00 00 00 E: 3.103810 16 01 00 0a 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Unplug headphone E: 9.951613 16 01 00 08 00 00 20 83 ab 08 00 03 80 00 00 00 00 E: 10.016073 16 01 00 04 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Plug Microphone E: 17.824186 16 01 00 06 00 00 20 83 ab 08 00 03 80 00 00 00 00 E: 20.928234 16 01 00 0a 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Unplug Microphone E: 26.734048 16 01 00 08 00 00 20 83 ab 08 00 08 c0 10 00 00 00 E: 26.784007 16 01 00 04 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Plug headset E: 44.160216 16 01 00 06 00 00 20 83 ab 08 00 03 80 00 00 00 00 # Unplug headset E: 56.687367 16 01 00 04 00 00 20 83 ab 08 00 08 c0 10 00 00 00 On the TB 3 dock: D: 0 R: 120 05 0c 09 01 a1 01 85 01 15 00 25 01 09 e9 09 ea 75 01 95 02 81 02 09 e2 95 01 81 06 09 00 95 02 81 03 09 36 a1 02 05 09 19 01 29 02 75 02 95 01 15 01 25 02 81 40 c0 05 0c 09 00 15 00 25 01 75 01 95 01 81 03 85 02 05 0c 09 00 95 10 81 02 85 03 09 00 91 02 85 04 09 00 75 08 95 26 91 02 85 05 09 00 95 22 81 02 85 06 09 00 95 24 91 02 85 07 09 00 95 20 81 02 c0 N: Lenovo ThinkPad Thunderbolt 3 Dock USB Audio P: usb-0000:0a:00.0-1/input3 I: 3 17ef 306a # Plug in headphone D: 0 E: 0.000000 3 02 00 80 # Unplug headphone E: 16.480011 3 02 00 00 # Plug in microphone E: 34.224331 3 02 00 80 # Unplug microphone E: 46.960128 3 02 00 00 # Plug in headset E: 62.592182 3 02 00 80 E: 63.120449 3 02 00 c0 # Unplug headset E: 72.608250 3 02 00 80 E: 72.624253 3 02 00 00

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[alsa-devel] Applied "ASoC: sdm845: remove unneeded semicolon" to the asoc tree
by Mark Brown 15 Sep '19

15 Sep '19

The patch ASoC: sdm845: remove unneeded semicolon has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 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 fca11622d600228bec405456f41590155b3a3eca Mon Sep 17 00:00:00 2001 From: Saiyam Doshi <saiyamdoshi.in(a)gmail.com> Date: Sat, 14 Sep 2019 08:41:33 +0530 Subject: [PATCH] ASoC: sdm845: remove unneeded semicolon Remove excess semicolon after closing parenthesis. Signed-off-by: Saiyam Doshi <saiyamdoshi.in(a)gmail.com> Link: https://lore.kernel.org/r/20190914031133.GA28447@SD Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/sdm845.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 882f52ed8231..28f3cef696e6 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -319,7 +319,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, 0, SNDRV_PCM_STREAM_PLAYBACK); - }; + } break; case SECONDARY_MI2S_TX: -- 2.20.1

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[alsa-devel] [PATCH] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add dmic format constraint
by Yu-Hsuan Hsu 14 Sep '19

14 Sep '19

24 bits recording from DMIC is not supported for KBL platform because the TDM slot between PCH and codec is 16 bits only. We should add a constraint to remove that unsupported format. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a01..67b276a65a8d2d 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -400,6 +400,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dmic_constraints); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -- 2.23.0.162.g0b9fbb3734-goog

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[alsa-devel] [PATCH] ASoC: fsl_sai: Implement set_bclk_ratio
by Daniel Baluta 13 Sep '19

13 Sep '19

From: Viorel Suman <viorel.suman(a)nxp.com> This is to allow machine drivers to set a certain bitclk rate which might not be exactly rate * frame size. Cc: NXP Linux Team <linux-imx(a)nxp.com> Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 21 +++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fe126029f4e3..e896b577b1f7 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -137,6 +137,16 @@ static int fsl_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, return 0; } +static int fsl_sai_set_dai_bclk_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai); + + sai->bclk_ratio = ratio; + + return 0; +} + static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -423,8 +433,14 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, slot_width = sai->slot_width; if (!sai->is_slave_mode) { - ret = fsl_sai_set_bclk(cpu_dai, tx, - slots * slot_width * params_rate(params)); + if (sai->bclk_ratio) + ret = fsl_sai_set_bclk(cpu_dai, tx, + sai->bclk_ratio * + params_rate(params)); + else + ret = fsl_sai_set_bclk(cpu_dai, tx, + slots * slot_width * + params_rate(params)); if (ret) return ret; @@ -640,6 +656,7 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { + .set_bclk_ratio = fsl_sai_set_dai_bclk_ratio, .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, .set_tdm_slot = fsl_sai_set_dai_tdm_slot, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 3a3f6f8e5595..f96f8d97489d 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -177,6 +177,7 @@ struct fsl_sai { unsigned int mclk_streams; unsigned int slots; unsigned int slot_width; + unsigned int bclk_ratio; const struct fsl_sai_soc_data *soc_data; struct snd_dmaengine_dai_dma_data dma_params_rx; -- 2.17.1

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[alsa-devel] [PATCH v3] drm: bridge/dw_hdmi: add audio sample channel status setting
by Cheng-Yi Chiang 13 Sep '19

13 Sep '19

From: Yakir Yang <ykk(a)rock-chips.com> When transmitting IEC60985 linear PCM audio, we configure the Aduio Sample Channel Status information in the IEC60958 frame. The status bit is already available in iec.status of hdmi_codec_params. This fix the issue that audio does not come out on some monitors (e.g. LG 22CV241) Note that these registers are only for interfaces: I2S audio interface, General Purpose Audio (GPA), or AHB audio DMA (AHBAUDDMA). For S/PDIF interface this information comes from the stream. Currently this function dw_hdmi_set_channel_status is only called from dw-hdmi-i2s-audio in I2S setup. Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Change from v2 to v3: 1. Reuse what is already set in iec.status in hw_param. 2. Remove all useless definition of registers and values. 3. Note that the original sampling frequency is not written to the channel status as we reuse create_iec958_consumer in pcm_iec958.c. Without that it can still play audio fine. .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 1 + drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 20 +++++++++++++++++++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.h | 2 ++ include/drm/bridge/dw_hdmi.h | 1 + 4 files changed, 24 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index 34d8e837555f..20f4f92dd866 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -102,6 +102,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, } dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); + dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); dw_hdmi_set_channel_allocation(hdmi, hparms->cea.channel_allocation); diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index bd65d0479683..aa7efd4da1c8 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -582,6 +582,26 @@ static unsigned int hdmi_compute_n(unsigned int freq, unsigned long pixel_clk) return n; } +/* + * When transmitting IEC60958 linear PCM audio, these registers allow to + * configure the channel status information of all the channel status + * bits in the IEC60958 frame. For the moment this configuration is only + * used when the I2S audio interface, General Purpose Audio (GPA), + * or AHB audio DMA (AHBAUDDMA) interface is active + * (for S/PDIF interface this information comes from the stream). + */ +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, + u8 *channel_status) +{ + /* + * Set channel status register for frequency and word length. + * Use default values for other registers. + */ + hdmi_writeb(hdmi, channel_status[3], HDMI_FC_AUDSCHNLS7); + hdmi_writeb(hdmi, channel_status[4], HDMI_FC_AUDSCHNLS8); +} +EXPORT_SYMBOL_GPL(dw_hdmi_set_channel_status); + static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, unsigned long pixel_clk, unsigned int sample_rate) { diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h index 6988f12d89d9..fcff5059db24 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -158,6 +158,8 @@ #define HDMI_FC_SPDDEVICEINF 0x1062 #define HDMI_FC_AUDSCONF 0x1063 #define HDMI_FC_AUDSSTAT 0x1064 +#define HDMI_FC_AUDSCHNLS7 0x106e +#define HDMI_FC_AUDSCHNLS8 0x106f #define HDMI_FC_DATACH0FILL 0x1070 #define HDMI_FC_DATACH1FILL 0x1071 #define HDMI_FC_DATACH2FILL 0x1072 diff --git a/include/drm/bridge/dw_hdmi.h b/include/drm/bridge/dw_hdmi.h index cf528c289857..4b3e863c4f8a 100644 --- a/include/drm/bridge/dw_hdmi.h +++ b/include/drm/bridge/dw_hdmi.h @@ -156,6 +156,7 @@ void dw_hdmi_setup_rx_sense(struct dw_hdmi *hdmi, bool hpd, bool rx_sense); void dw_hdmi_set_sample_rate(struct dw_hdmi *hdmi, unsigned int rate); void dw_hdmi_set_channel_count(struct dw_hdmi *hdmi, unsigned int cnt); +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, u8 *channel_status); void dw_hdmi_set_channel_allocation(struct dw_hdmi *hdmi, unsigned int ca); void dw_hdmi_audio_enable(struct dw_hdmi *hdmi); void dw_hdmi_audio_disable(struct dw_hdmi *hdmi); -- 2.23.0.162.g0b9fbb3734-goog

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[alsa-devel] [PATCH 0/3] update supported sample format
by Shengjiu Wang 13 Sep '19

13 Sep '19

update supported sample format Shengjiu Wang (3): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++++--------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 93 +++++++++++++++++++++++++++++++++--- 3 files changed, 135 insertions(+), 30 deletions(-) -- 2.21.0

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[alsa-devel] [PATCH 0/2] ASoC: mt8183: fix audio playback slowly after playback during bootup
by Jiaxin Yu 12 Sep '19

12 Sep '19

These two patches fix audio playack slowly after playbakc during bootup. The patches are based on broonie tree "for-next" branch. Jiaxin Yu (2): ASoC: dt-bingdings: mediatek: mt8183: add a property "medaitek,toprgu" ASoC: mt8183: fix audio playback slowly after playback during bootup .../bindings/sound/mt8183-afe-pcm.txt | 2 ++ sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 21 +++++++++++++++++++ sound/soc/mediatek/mt8183/mt8183-reg.h | 6 ++++++ 4 files changed, 30 insertions(+) -- 2.18.0

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[alsa-devel] [PATCH v3 00/10] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 12 Sep '19

12 Sep '19

Hi all, here's the third round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. This is now first series without the RFC tag. The series has all machine drivers converted that are directly impacted. v3 changes: - Merged two mst_no_extra_pcms patches (Takashi's feedback). - Use mach_params instead of module name lookups to determine which codec driver is used at runtime (Pierre's feedback). - Extended machine driver support to four more drivers. This includes now all machine drivers which are used with SOF and which support HDMI audio. With this patchset, the machine drivers continue to work with both SOF and SST DSP drivers, and can be configured to use hdac-hdmi also with SOF (via kernel module parameter -> this is for distros to provide backwards compatibility for UCM files). - Various smaller cleanups across the patchset. Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (10): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ALSA: hda/hdmi - allow control creation without a linked pcm ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 23 +++-- sound/soc/codecs/hdac_hda.c | 95 ++++++++++++++++--- sound/soc/codecs/hdac_hda.h | 12 ++- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 +++ sound/soc/intel/boards/bxt_rt298.c | 11 +++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 +++ sound/soc/intel/boards/hda_dsp_common.h | 87 +++++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 10 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 27 +++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 +++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 19 +++- sound/soc/sof/intel/hda.c | 6 ++ sound/soc/sof/intel/hda.h | 6 +- 17 files changed, 315 insertions(+), 28 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

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[alsa-devel] [PATCH v3] ASoC: Intel: Skylake: Recover BXT FW on DSP boot timeout error
by Pawel Harlozinski 12 Sep '19

12 Sep '19

When DSP boots with timeout error try to reload firmware to keep best user expeirience. Change-Id: Ic7377da687e7441a487c443f9de1ebae5967afa2 Signed-off-by: Szymon Mielczarek <szymonx.mielczarek(a)intel.com> Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 92a82e6b5fe6..fdf4c198445e 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -446,14 +446,19 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) /* If core 1 was turned on for booting core 0, turn it off */ skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); + /* FIXME this FW recovery applies only for BXT */ if (ret == 0) { - dev_err(ctx->dev, "%s: DSP boot timeout\n", __func__); - dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + dev_warn(ctx->dev, + "DSP boot timeout: Error code=0x%x: FW status=0x%x. Reloading FW\n", sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); - ret = -EIO; - goto err; + + ret = bxt_sst_init_fw(skl->dev, skl); + if (ret < 0) { + dev_err(ctx->dev, "Reload fw failed: %d. Failed to set core0 to D0 state\n", + ret); + goto err; + } } } @@ -478,7 +483,6 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) if (core_id == SKL_DSP_CORE0_ID) core_mask |= SKL_DSP_CORE_MASK(1); skl_dsp_disable_core(ctx, core_mask); - return ret; } -- 2.17.1

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[alsa-devel] [PATCH v3] tas2770: add tas2770 smart PA kernel driver
by shifu0704＠thundersoft.com 12 Sep '19

12 Sep '19

From: Frank Shi <shifu0704(a)thundersoft.com> add tas2770 smart PA kernel driver Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2770.c | 952 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2770.h | 166 ++++++++ 4 files changed, 1125 insertions(+) create mode 100644 sound/soc/codecs/tas2770.c create mode 100644 sound/soc/codecs/tas2770.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8f3e787..cc92da3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -111,6 +111,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_STI_SAS select SND_SOC_TAS2552 if I2C + select SND_SOC_TAS2770 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C select SND_SOC_TFA9879 if I2C @@ -652,6 +653,10 @@ config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" depends on I2C +config SND_SOC_TAS2770 + tristate "Texas Instruments TAS2770 speaker amplifier" + depends on I2C + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5305cc6..63b8488 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -116,6 +116,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-sti-sas-objs := sti-sas.o snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o +snd-soc-tas2770-objs := tas2770.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o @@ -332,6 +333,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o +obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c new file mode 100644 index 0000000..65aeace --- /dev/null +++ b/sound/soc/codecs/tas2770.c @@ -0,0 +1,952 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2770 20-W Digital Input Mono Class-D +// Audio Amplifier with Speaker I/V Sense +// +// Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ +// Author: Tracy Yi <tracy-yi(a)ti.com> +// Frank Shi <shifu0704(a)thundersoft.com> + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/err.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/firmware.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "tas2770.h" + +#define TAS2770_MDELAY 0xFFFFFFFE +#define TAS2770_CHECK_PERIOD 5000 /* 5 second */ + +static void tas2770_reset(struct tas2770_priv *tas2770) +{ + if (tas2770->reset_gpio) { + gpiod_set_value_cansleep(tas2770->reset_gpio, 0); + msleep(20); + gpiod_set_value_cansleep(tas2770->reset_gpio, 1); + } + snd_soc_component_write(tas2770->component, TAS2770_SW_RST, + TAS2770_RST); +} + +static int tas2770_regmap_read(struct tas2770_priv *tas2770, + unsigned int reg, unsigned int *value) +{ + int result = 0; + int retry_count = TAS2770_I2C_RETRY_COUNT; + + while (retry_count--) { + result = snd_soc_component_read(tas2770->component, reg, + value); + if (!result) + break; + + msleep(20); + } + if (!retry_count) + return -ETIMEDOUT; + + return 0; +} + +static int tas2770_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_ACTIVE); + tas2770->power_state = SND_SOC_BIAS_STANDBY; + break; + + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_MUTE); + tas2770->power_state = SND_SOC_BIAS_PREPARE; + break; + + case SND_SOC_BIAS_OFF: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_SHUTDOWN); + tas2770->power_state = SND_SOC_BIAS_OFF; + break; + + default: + dev_err(tas2770->dev, + "wrong power level setting %d\n", level); + return -EINVAL; + } + + return 0; +} + +#ifdef CONFIG_PM +static int tas2770_codec_suspend(struct snd_soc_component *component) +{ + int ret; + + ret = tas2770_set_bias_level(component, SND_SOC_BIAS_OFF); + if (ret) + return -EINVAL; + + return 0; +} + +static int tas2770_codec_resume(struct snd_soc_component *component) +{ + int ret; + + ret = tas2770_set_bias_level(component, SND_SOC_BIAS_STANDBY); + if (ret) + return -EINVAL; + + return 0; +} +#else +#define tas2770_codec_suspend NULL +#define tas2770_codec_resume NULL +#endif + +static const char * const tas2770_ASI1_src[] = { + "I2C offset", "Left", "Right", "LeftRightDiv2", +}; + +static SOC_ENUM_SINGLE_DECL( + tas2770_ASI1_src_enum, TAS2770_TDM_CFG_REG2, + 4, tas2770_ASI1_src); + +static const struct snd_kcontrol_new tas2770_asi1_mux = + SOC_DAPM_ENUM("ASI1 Source", tas2770_ASI1_src_enum); + +static int tas2770_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + ret = tas2770_set_bias_level(component, + SND_SOC_BIAS_PREPARE); + if (ret) + goto end; + break; + case SND_SOC_DAPM_PRE_PMD: + ret = tas2770_set_bias_level(component, + SND_SOC_BIAS_OFF); + if (ret) + goto end; + break; + default: + dev_err(tas2770->dev, "Not supported evevt\n"); + return -EINVAL; + } + +end: + return ret; +} + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", TAS2770_PWR_CTRL, 3, 1, 1); +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", TAS2770_PWR_CTRL, 2, 1, 1); + +static const struct snd_soc_dapm_widget tas2770_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, + &tas2770_asi1_mux), + SND_SOC_DAPM_SWITCH("ISENSE", TAS2770_PWR_CTRL, 3, 1, + &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", TAS2770_PWR_CTRL, 2, 1, + &vsense_switch), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2770_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON") +}; + +static const struct snd_soc_dapm_route tas2770_audio_map[] = { + {"ASI1 Sel", "I2C offset", "ASI1"}, + {"ASI1 Sel", "Left", "ASI1"}, + {"ASI1 Sel", "Right", "ASI1"}, + {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, + {"DAC", NULL, "ASI1 Sel"}, + {"OUT", NULL, "DAC"}, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + +static int tas2770_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + int ret; + + if (mute) + ret = tas2770_set_bias_level(component, SND_SOC_BIAS_PREPARE); + else + ret = tas2770_set_bias_level(component, SND_SOC_BIAS_STANDBY); + + return ret; +} + +static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) +{ + int ret; + struct snd_soc_component *component = tas2770->component; + + switch (bitwidth) { + case SNDRV_PCM_FORMAT_S16_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_16BITS); + tas2770->v_sense_slot = tas2770->i_sense_slot + 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_24BITS); + tas2770->v_sense_slot = tas2770->i_sense_slot + 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_32BITS); + tas2770->v_sense_slot = tas2770->i_sense_slot + 4; + break; + + default: + return -EINVAL; + } + + tas2770->channel_size = bitwidth; + + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG5, + TAS2770_TDM_CFG_REG5_VSNS_MASK | + TAS2770_TDM_CFG_REG5_50_MASK, + TAS2770_TDM_CFG_REG5_VSNS_ENABLE | + tas2770->v_sense_slot); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG6, + TAS2770_TDM_CFG_REG6_ISNS_MASK | + TAS2770_TDM_CFG_REG6_50_MASK, + TAS2770_TDM_CFG_REG6_ISNS_ENABLE | + tas2770->i_sense_slot); + +end: + return ret; +} + +static int tas2770_set_samplerate(struct tas2770_priv *tas2770, + int samplerate) +{ + int ret; + struct snd_soc_component *component = tas2770->component; + + switch (samplerate) { + case 48000: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); + if (ret) + goto end; + break; + case 44100: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); + if (ret) + goto end; + break; + case 96000: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); + break; + case 88200: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); + break; + case 19200: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); + if (ret) + goto end; + break; + case 17640: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); + break; + default: + ret = -EINVAL; + } + +end: + if (!ret) + tas2770->sampling_rate = samplerate; + return ret; +} + +static int tas2770_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + ret = tas2770_set_bitwidth(tas2770, params_format(params)); + if (ret < 0) + goto end; + + + ret = tas2770_set_samplerate(tas2770, params_rate(params)); + +end: + return ret; +} + +static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + int ret; + int value = 0; + struct snd_soc_component *component = dai->component; + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(tas2770->dev, "ASI format master is not found\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_RSING; + break; + case SND_SOC_DAIFMT_IB_NF: + asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_FALING; + break; + default: + dev_err(tas2770->dev, "ASI format Inverse is not found\n"); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, + TAS2770_TDM_CFG_REG1_RX_MASK, + asi_cfg_1); + if (ret) + return ret; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case (SND_SOC_DAIFMT_I2S): + tdm_rx_start_slot = 1; + break; + case (SND_SOC_DAIFMT_DSP_A): + tdm_rx_start_slot = 0; + break; + case (SND_SOC_DAIFMT_DSP_B): + tdm_rx_start_slot = 1; + break; + case (SND_SOC_DAIFMT_LEFT_J): + tdm_rx_start_slot = 0; + break; + default: + dev_err(tas2770->dev, + "DAI Format is not found, fmt=0x%x\n", fmt); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, + TAS2770_TDM_CFG_REG1_MASK, + (tdm_rx_start_slot << TAS2770_TDM_CFG_REG1_51_SHIFT)); + if (ret) + return ret; + + value = snd_soc_component_read32(component, TAS2770_TDM_CFG_REG3); + + tas2770->asi_format = fmt; + + return 0; +} + +static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + int left_slot, right_slot; + int ret; + + if (tx_mask == 0 || rx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, + TAS2770_TDM_CFG_REG3_30_MASK, + (left_slot << TAS2770_TDM_CFG_REG3_30_SHIFT)); + if (ret) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, + TAS2770_TDM_CFG_REG3_RXS_MASK, + (right_slot << TAS2770_TDM_CFG_REG3_RXS_SHIFT)); + if (ret) + return ret; + + switch (slot_width) { + case 16: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_16BITS); + break; + + case 24: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_24BITS); + break; + + case 32: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_32BITS); + break; + + case 0: + /* Do not change slot width */ + ret = 0; + break; + + default: + ret = -EINVAL; + } + + if (!ret) + tas2770->slot_width = slot_width; + + return ret; +} + +static struct snd_soc_dai_ops tas2770_dai_ops = { + .digital_mute = tas2770_mute, + .hw_params = tas2770_hw_params, + .set_fmt = tas2770_set_fmt, + .set_tdm_slot = tas2770_set_dai_tdm_slot, +}; + +#define TAS2770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +#define TAS2770_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_192000\ + ) + +static struct snd_soc_dai_driver tas2770_dai_driver[] = { + { + .name = "tas2770 ASI1", + .id = 0, + .playback = { + .stream_name = "ASI1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TAS2770_RATES, + .formats = TAS2770_FORMATS, + }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 0, + .channels_max = 2, + .rates = TAS2770_RATES, + .formats = TAS2770_FORMATS, + }, + .ops = &tas2770_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int tas2770_codec_probe(struct snd_soc_component *component) +{ + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + + tas2770->component = component; + + return 0; +} + +static DECLARE_TLV_DB_SCALE(tas2770_digital_tlv, 1100, 50, 0); +static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -12750, 50, 0); + +static const struct snd_kcontrol_new tas2770_snd_controls[] = { + SOC_SINGLE_TLV("Amp Output Gain", TAS2770_PLAY_CFG_REG0, + 0, 0x14, 0, + tas2770_digital_tlv), + SOC_SINGLE_TLV("Playback Volume", TAS2770_PLAY_CFG_REG2, + 0, TAS2770_PLAY_CFG_REG2_VMAX, 1, + tas2770_playback_volume), +}; + +static const struct snd_soc_component_driver soc_component_driver_tas2770 = { + .probe = tas2770_codec_probe, + .suspend = tas2770_codec_suspend, + .resume = tas2770_codec_resume, + .set_bias_level = tas2770_set_bias_level, + .controls = tas2770_snd_controls, + .num_controls = ARRAY_SIZE(tas2770_snd_controls), + .dapm_widgets = tas2770_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2770_dapm_widgets), + .dapm_routes = tas2770_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2770_audio_map), + .idle_bias_on = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int tas2770_register_codec(struct tas2770_priv *tas2770) +{ + return devm_snd_soc_register_component(tas2770->dev, + &soc_component_driver_tas2770, + tas2770_dai_driver, ARRAY_SIZE(tas2770_dai_driver)); +} + +static void tas2770_load_config(struct tas2770_priv *tas2770) +{ + int ret; + struct snd_soc_component *component = tas2770->component; + + tas2770_reset(tas2770); + snd_soc_component_write(component, TAS2770_SW_RST, + TAS2770_RST); + + ret = tas2770_set_bitwidth(tas2770, tas2770->channel_size); + if (ret) + goto error_out; + + ret = tas2770_set_samplerate(tas2770, tas2770->sampling_rate); + if (ret) + goto error_out; + + ret = tas2770_set_bias_level(component, tas2770->power_state); + if (ret) + goto error_out; + +error_out: + return; +} + +static const struct reg_default tas2770_reg_defaults[] = { + { TAS2770_PAGE, 0x00 }, + { TAS2770_SW_RST, 0x00 }, + { TAS2770_PWR_CTRL, 0x0e }, + { TAS2770_PLAY_CFG_REG0, 0x10 }, + { TAS2770_PLAY_CFG_REG1, 0x01 }, + { TAS2770_PLAY_CFG_REG2, 0x00 }, + { TAS2770_MSC_CFG_REG0, 0x07 }, + { TAS2770_TDM_CFG_REG1, 0x02 }, + { TAS2770_TDM_CFG_REG2, 0x0a }, + { TAS2770_TDM_CFG_REG3, 0x10 }, + { TAS2770_INT_MASK_REG0, 0xfc }, + { TAS2770_INT_MASK_REG1, 0xb1 }, + { TAS2770_INT_CFG, 0x05 }, + { TAS2770_MISC_IRQ, 0x81 }, + { TAS2770_CLK_CGF, 0x0c }, + +}; + +static bool tas2770_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2770_PAGE: /* regmap implementation requires this */ + case TAS2770_SW_RST: /* always clears after write */ + case TAS2770_BO_PRV_REG0:/* has a self clearing bit */ + case TAS2770_LVE_INT_REG0: + case TAS2770_LVE_INT_REG1: + case TAS2770_LAT_INT_REG0:/* Sticky interrupt flags */ + case TAS2770_LAT_INT_REG1:/* Sticky interrupt flags */ + case TAS2770_VBAT_MSB: + case TAS2770_VBAT_LSB: + case TAS2770_TEMP_MSB: + case TAS2770_TEMP_LSB: + return true; + } + return false; +} + +static bool tas2770_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2770_LVE_INT_REG0: + case TAS2770_LVE_INT_REG1: + case TAS2770_LAT_INT_REG0: + case TAS2770_LAT_INT_REG1: + case TAS2770_VBAT_MSB: + case TAS2770_VBAT_LSB: + case TAS2770_TEMP_MSB: + case TAS2770_TEMP_LSB: + case TAS2770_TDM_CLK_DETC: + case TAS2770_REV_AND_GPID: + return false; + } + return true; +} + +static const struct regmap_range_cfg tas2770_regmap_ranges[] = { + { + .range_min = 0, + .range_max = 1 * 128, + .selector_reg = TAS2770_PAGE, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tas2770_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = tas2770_writeable, + .volatile_reg = tas2770_volatile, + .reg_defaults = tas2770_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2770_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = tas2770_regmap_ranges, + .num_ranges = ARRAY_SIZE(tas2770_regmap_ranges), + .max_register = 1 * 128, +}; + +static irqreturn_t tas2770_irq_handler(int irq, void *dev_id) +{ + struct tas2770_priv *tas2770 = (struct tas2770_priv *)dev_id; + struct snd_soc_component *component = tas2770->component; + unsigned int device_status_1 = 0, device_status_2 = 0; + int result; + + if (tas2770->runtime_suspend) + goto end; + + if (tas2770->power_state == TAS2770_POWER_SHUTDOWN) + goto end; + + result = snd_soc_component_write(component, TAS2770_INT_MASK_REG0, + TAS2770_INT_MASK_REG0_DISABLE); + if (result) + goto reload; + result = snd_soc_component_write(component, TAS2770_INT_MASK_REG1, + TAS2770_INT_MASK_REG1_DISABLE); + if (result) + goto reload; + + result = tas2770_regmap_read(tas2770, + TAS2770_LAT_INT_REG0, &device_status_1); + if (!result && (device_status_1 & 0x3)) { + if (device_status_1 & + TAS2770_LAT_INT_REG0_OCE_FLG) { + tas2770->err_code |= ERROR_OVER_CURRENT; + dev_err(tas2770->dev, "SPK over current!\n"); + } else { + tas2770->err_code &= ~ERROR_OVER_CURRENT; + } + + if (device_status_1 & + TAS2770_LAT_INT_REG0_OTE_FLG) { + tas2770->err_code |= ERROR_DIE_OVERTEMP; + dev_err(tas2770->dev, "die over temperature!\n"); + } else { + tas2770->err_code &= ~ERROR_DIE_OVERTEMP; + } + } + + result = tas2770_regmap_read(tas2770, + TAS2770_LAT_INT_REG1, &device_status_2); + if (!result && (device_status_2 & 0x0f)) { + if (device_status_2 & + TAS2770_LAT_INT_REG1_VBA_TOV) { + tas2770->err_code |= ERROR_OVER_VOLTAGE; + dev_err(tas2770->dev, "SPK over voltage!\n"); + } else { + tas2770->err_code &= ~ERROR_OVER_VOLTAGE; + } + + if (device_status_2 & + TAS2770_LAT_INT_REG1_VBA_TUV) { + tas2770->err_code |= ERROR_UNDER_VOLTAGE; + dev_err(tas2770->dev, "SPK under voltage!\n"); + } else { + tas2770->err_code &= ~ERROR_UNDER_VOLTAGE; + } + + if (device_status_2 & + TAS2770_LAT_INT_REG1_BOUT_FLG) { + tas2770->err_code |= ERROR_BROWNOUT; + dev_err(tas2770->dev, "brownout!\n"); + } else { + tas2770->err_code &= ~ERROR_BROWNOUT; + } + } + + result = snd_soc_component_write(component, + TAS2770_INT_MASK_REG0, TAS2770_INT_REG0_DEFAULT); + if (result) + goto reload; + + result = snd_soc_component_write(component, + TAS2770_INT_MASK_REG1, TAS2770_INT_REG1_DEFAULT); + if (result) + goto reload; + + goto end; + +reload: + /* hardware reset and reload */ + tas2770_load_config(tas2770); + +end: + return IRQ_HANDLED; +} + +static int tas2770_parse_dt(struct device *dev, struct tas2770_priv *tas2770) +{ + int rc = 0; + + rc = fwnode_property_read_u32(dev->fwnode, "ti,asi-format", + &tas2770->asi_format); + if (rc) { + dev_err(tas2770->dev, "Looking up %s property failed %d\n", + "ti,asi-format", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,imon-slot-no", + &tas2770->i_sense_slot); + if (rc) { + dev_err(tas2770->dev, "Looking up %s property failed %d\n", + "ti,imon-slot-no", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,vmon-slot-no", + &tas2770->v_sense_slot); + if (rc) { + dev_err(tas2770->dev, "Looking up %s property failed %d\n", + "ti,vmon-slot-no", rc); + goto end; + } + +end: + return rc; +} + +static int tas2770_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tas2770_priv *tas2770; + int result; + + tas2770 = devm_kzalloc(&client->dev, + sizeof(struct tas2770_priv), GFP_KERNEL); + if (!tas2770) + return -ENOMEM; + tas2770->dev = &client->dev; + tas2770->irq_gpio = client->irq; + + i2c_set_clientdata(client, tas2770); + dev_set_drvdata(&client->dev, tas2770); + tas2770->power_state = TAS2770_POWER_SHUTDOWN; + + tas2770->regmap = devm_regmap_init_i2c(client, &tas2770_i2c_regmap); + if (IS_ERR(tas2770->regmap)) { + result = PTR_ERR(tas2770->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + result); + goto end; + } + + if (client->dev.of_node) { + result = tas2770_parse_dt(&client->dev, tas2770); + if (result) { + dev_err(tas2770->dev, "%s: Failed to parse devicetree\n", + __func__); + goto end; + } + } + + tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev, + "reset-gpio", + GPIOD_OUT_HIGH); + if (IS_ERR(tas2770->reset_gpio)) { + if (PTR_ERR(tas2770->reset_gpio) == -EPROBE_DEFER) { + tas2770->reset_gpio = NULL; + return -EPROBE_DEFER; + } + } + + if (tas2770->irq_gpio > 0) { + result = devm_request_threaded_irq(tas2770->dev, + tas2770->irq_gpio, + NULL, tas2770_irq_handler, + IRQF_TRIGGER_FALLING | IRQF_ONESHOT, client->name, + tas2770); + if (result) { + dev_err(tas2770->dev, "Unable to request IRQ\n"); + return result; + } + } + + tas2770->channel_size = 0; + tas2770->slot_width = 0; + + tas2770_reset(tas2770); + + result = tas2770_register_codec(tas2770); + if (result) + dev_err(tas2770->dev, "Register codec failed.\n"); + +end: + return result; +} + +static int tas2770_i2c_remove(struct i2c_client *client) +{ + pm_runtime_disable(&client->dev); + return 0; +} + + +static const struct i2c_device_id tas2770_i2c_id[] = { + { "tas2770", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2770_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id tas2770_of_match[] = { + { .compatible = "ti,tas2770" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2770_of_match); +#endif + +static struct i2c_driver tas2770_i2c_driver = { + .driver = { + .name = "tas2770", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2770_of_match), + }, + .probe = tas2770_i2c_probe, + .remove = tas2770_i2c_remove, + .id_table = tas2770_i2c_id, +}; + +module_i2c_driver(tas2770_i2c_driver); + +MODULE_AUTHOR("Shi Fu <shifu0704(a)thundersoft.com>"); +MODULE_DESCRIPTION("TAS2770 I2C Smart Amplifier driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/tas2770.h b/sound/soc/codecs/tas2770.h new file mode 100644 index 0000000..bf59dc2 --- /dev/null +++ b/sound/soc/codecs/tas2770.h @@ -0,0 +1,166 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC TAS2770 codec driver + * + * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ + */ +#ifndef __TAS2770__ +#define __TAS2770__ + +/* Book Control Register (available in page0 of each book) */ +#define TAS2770_BOOKCTL_PAGE 0 +#define TAS2770_BOOKCTL_REG 127 +#define TAS2770_REG(page, reg) ((page * 128) + reg) + /* Page */ +#define TAS2770_PAGE TAS2770_REG(0X0, 0x00) +#define TAS2770_PAGE_PAGE_MASK 255 + /* Software Reset */ +#define TAS2770_SW_RST TAS2770_REG(0X0, 0x01) +#define TAS2770_RST BIT(0) + /* Power Control */ +#define TAS2770_PWR_CTRL TAS2770_REG(0X0, 0x02) +#define TAS2770_PWR_CTRL_MASK 0x3 +#define TAS2770_PWR_CTRL_ACTIVE 0x0 +#define TAS2770_PWR_CTRL_MUTE BIT(0) +#define TAS2770_PWR_CTRL_SHUTDOWN 0x2 + /* Playback Configuration Reg0 */ +#define TAS2770_PLAY_CFG_REG0 TAS2770_REG(0X0, 0x03) + /* Playback Configuration Reg1 */ +#define TAS2770_PLAY_CFG_REG1 TAS2770_REG(0X0, 0x04) + /* Playback Configuration Reg2 */ +#define TAS2770_PLAY_CFG_REG2 TAS2770_REG(0X0, 0x05) +#define TAS2770_PLAY_CFG_REG2_VMAX 0xc9 + /* Misc Configuration Reg0 */ +#define TAS2770_MSC_CFG_REG0 TAS2770_REG(0X0, 0x07) + /* TDM Configuration Reg0 */ +#define TAS2770_TDM_CFG_REG0 TAS2770_REG(0X0, 0x0A) +#define TAS2770_TDM_CFG_REG0_SMP_MASK BIT(5) +#define TAS2770_TDM_CFG_REG0_SMP_48KHZ 0x0 +#define TAS2770_TDM_CFG_REG0_SMP_44_1KHZ BIT(5) +#define TAS2770_TDM_CFG_REG0_31_MASK 0xe +#define TAS2770_TDM_CFG_REG0_31_44_1_48KHZ 0x6 +#define TAS2770_TDM_CFG_REG0_31_88_2_96KHZ 0x8 +#define TAS2770_TDM_CFG_REG0_31_176_4_192KHZ 0xa + /* TDM Configuration Reg1 */ +#define TAS2770_TDM_CFG_REG1 TAS2770_REG(0X0, 0x0B) +#define TAS2770_TDM_CFG_REG1_MASK 0x3e +#define TAS2770_TDM_CFG_REG1_51_SHIFT 1 +#define TAS2770_TDM_CFG_REG1_RX_MASK BIT(0) +#define TAS2770_TDM_CFG_REG1_RX_RSING 0x0 +#define TAS2770_TDM_CFG_REG1_RX_FALING BIT(0) + /* TDM Configuration Reg2 */ +#define TAS2770_TDM_CFG_REG2 TAS2770_REG(0X0, 0x0C) +#define TAS2770_TDM_CFG_REG2_RXW_MASK 0xc +#define TAS2770_TDM_CFG_REG2_RXW_16BITS 0x0 +#define TAS2770_TDM_CFG_REG2_RXW_24BITS 0x8 +#define TAS2770_TDM_CFG_REG2_RXW_32BITS 0xc +#define TAS2770_TDM_CFG_REG2_RXS_MASK 0x3 +#define TAS2770_TDM_CFG_REG2_RXS_16BITS 0x0 +#define TAS2770_TDM_CFG_REG2_RXS_24BITS BIT(0) +#define TAS2770_TDM_CFG_REG2_RXS_32BITS 0x2 + /* TDM Configuration Reg3 */ +#define TAS2770_TDM_CFG_REG3 TAS2770_REG(0X0, 0x0D) +#define TAS2770_TDM_CFG_REG3_RXS_MASK 0xf0 +#define TAS2770_TDM_CFG_REG3_RXS_SHIFT 0x4 +#define TAS2770_TDM_CFG_REG3_30_MASK 0xf +#define TAS2770_TDM_CFG_REG3_30_SHIFT 0 + /* TDM Configuration Reg5 */ +#define TAS2770_TDM_CFG_REG5 TAS2770_REG(0X0, 0x0F) +#define TAS2770_TDM_CFG_REG5_VSNS_MASK BIT(6) +#define TAS2770_TDM_CFG_REG5_VSNS_ENABLE BIT(6) +#define TAS2770_TDM_CFG_REG5_50_MASK 0x3f + /* TDM Configuration Reg6 */ +#define TAS2770_TDM_CFG_REG6 TAS2770_REG(0X0, 0x10) +#define TAS2770_TDM_CFG_REG6_ISNS_MASK BIT(6) +#define TAS2770_TDM_CFG_REG6_ISNS_ENABLE BIT(6) +#define TAS2770_TDM_CFG_REG6_50_MASK 0x3f + /* Brown Out Prevention Reg0 */ +#define TAS2770_BO_PRV_REG0 TAS2770_REG(0X0, 0x1B) + /* Interrupt MASK Reg0 */ +#define TAS2770_INT_MASK_REG0 TAS2770_REG(0X0, 0x20) +#define TAS2770_INT_REG0_DEFAULT 0xfc +#define TAS2770_INT_MASK_REG0_DISABLE 0xff + /* Interrupt MASK Reg1 */ +#define TAS2770_INT_MASK_REG1 TAS2770_REG(0X0, 0x21) +#define TAS2770_INT_REG1_DEFAULT 0xb1 +#define TAS2770_INT_MASK_REG1_DISABLE 0xff + /* Live-Interrupt Reg0 */ +#define TAS2770_LVE_INT_REG0 TAS2770_REG(0X0, 0x22) + /* Live-Interrupt Reg1 */ +#define TAS2770_LVE_INT_REG1 TAS2770_REG(0X0, 0x23) + /* Latched-Interrupt Reg0 */ +#define TAS2770_LAT_INT_REG0 TAS2770_REG(0X0, 0x24) +#define TAS2770_LAT_INT_REG0_OCE_FLG BIT(1) +#define TAS2770_LAT_INT_REG0_OTE_FLG BIT(0) + /* Latched-Interrupt Reg1 */ +#define TAS2770_LAT_INT_REG1 TAS2770_REG(0X0, 0x25) +#define TAS2770_LAT_INT_REG1_VBA_TOV BIT(3) +#define TAS2770_LAT_INT_REG1_VBA_TUV BIT(2) +#define TAS2770_LAT_INT_REG1_BOUT_FLG BIT(1) + /* VBAT MSB */ +#define TAS2770_VBAT_MSB TAS2770_REG(0X0, 0x27) + /* VBAT LSB */ +#define TAS2770_VBAT_LSB TAS2770_REG(0X0, 0x28) + /* TEMP MSB */ +#define TAS2770_TEMP_MSB TAS2770_REG(0X0, 0x29) + /* TEMP LSB */ +#define TAS2770_TEMP_LSB TAS2770_REG(0X0, 0x2A) + /* Interrupt Configuration */ +#define TAS2770_INT_CFG TAS2770_REG(0X0, 0x30) + /* Misc IRQ */ +#define TAS2770_MISC_IRQ TAS2770_REG(0X0, 0x32) + /* Clock Configuration */ +#define TAS2770_CLK_CGF TAS2770_REG(0X0, 0x3C) + /* TDM Clock detection monitor */ +#define TAS2770_TDM_CLK_DETC TAS2770_REG(0X0, 0x77) + /* Revision and PG ID */ +#define TAS2770_REV_AND_GPID TAS2770_REG(0X0, 0x7D) + +#define TAS2770_POWER_ACTIVE 0 +#define TAS2770_POWER_MUTE 1 +#define TAS2770_POWER_SHUTDOWN 2 +#define ERROR_OVER_CURRENT 0x0000001 +#define ERROR_DIE_OVERTEMP 0x0000002 +#define ERROR_OVER_VOLTAGE 0x0000004 +#define ERROR_UNDER_VOLTAGE 0x0000008 +#define ERROR_BROWNOUT 0x0000010 +#define ERROR_CLASSD_PWR 0x0000020 +#define TAS2770_SLOT_16BIT 16 +#define TAS2770_SLOT_32BIT 32 +#define TAS2770_I2C_RETRY_COUNT 3 + +struct tas2770_register { + int book; + int page; + int reg; +}; + +struct tas2770_dai_cfg { + unsigned int dai_fmt; + unsigned int tdm_delay; +}; + +struct tas2770_priv { + struct device *dev; + struct regmap *regmap; + struct snd_soc_codec *codec; + struct snd_soc_component *component; + struct mutex dev_lock; + struct hrtimer mtimer; + int power_state; + int asi_format; + struct gpio_desc *reset_gpio; + int irq_gpio; + bool irq_enable; + int sampling_rate; + int frame_size; + int channel_size; + int slot_width; + int v_sense_slot; + int i_sense_slot; + bool runtime_suspend; + unsigned int err_code; + struct mutex codec_lock; +}; + +#endif /* __TAS2770__ */ -- 2.7.4

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[alsa-devel] [PATCH] ASoC: wcd9335: remove redundant use of ret variable
by Saiyam Doshi 11 Sep '19

11 Sep '19

All these functions declares and initializes variable ret with '0' and without modifying 'ret' variable, it is returned. This patch removes this redundancy and returns '0' directly. Signed-off-by: Saiyam Doshi <saiyamdoshi.in(a)gmail.com> --- sound/soc/codecs/wcd9335.c | 24 ++++++++---------------- 1 file changed, 8 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 03f8a94bba2f..f318403133e9 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -3022,7 +3022,6 @@ static int wcd9335_codec_enable_slim(struct snd_soc_dapm_widget *w, struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = snd_soc_component_get_drvdata(comp); struct wcd_slim_codec_dai_data *dai = &wcd->dai[w->shift]; - int ret = 0; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -3034,7 +3033,7 @@ static int wcd9335_codec_enable_slim(struct snd_soc_dapm_widget *w, break; } - return ret; + return 0; } static int wcd9335_codec_enable_mix_path(struct snd_soc_dapm_widget *w, @@ -3539,7 +3538,6 @@ static int wcd9335_codec_hphl_dac_event(struct snd_soc_dapm_widget *w, struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); int hph_mode = wcd->hph_mode; u8 dem_inp; - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3579,7 +3577,7 @@ static int wcd9335_codec_hphl_dac_event(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static int wcd9335_codec_lineout_dac_event(struct snd_soc_dapm_widget *w, @@ -3607,7 +3605,6 @@ static int wcd9335_codec_ear_dac_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3621,7 +3618,7 @@ static int wcd9335_codec_ear_dac_event(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static void wcd9335_codec_hph_post_pa_config(struct wcd9335_codec *wcd, @@ -3692,7 +3689,6 @@ static int wcd9335_codec_hphr_dac_event(struct snd_soc_dapm_widget *w, struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); int hph_mode = wcd->hph_mode; u8 dem_inp; - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3731,7 +3727,7 @@ static int wcd9335_codec_hphr_dac_event(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, @@ -3741,7 +3737,6 @@ static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); int hph_mode = wcd->hph_mode; - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3780,7 +3775,7 @@ static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w, @@ -3789,7 +3784,6 @@ static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w, { struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); int vol_reg = 0, mix_vol_reg = 0; - int ret = 0; if (w->reg == WCD9335_ANA_LO_1_2) { if (w->shift == 7) { @@ -3837,7 +3831,7 @@ static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static void wcd9335_codec_init_flyback(struct snd_soc_component *component) @@ -3892,7 +3886,6 @@ static int wcd9335_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); int hph_mode = wcd->hph_mode; - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3930,14 +3923,13 @@ static int wcd9335_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static int wcd9335_codec_enable_ear_pa(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); - int ret = 0; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -3967,7 +3959,7 @@ static int wcd9335_codec_enable_ear_pa(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static irqreturn_t wcd9335_slimbus_irq(int irq, void *data) -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: dmaengine: Replace strncpy() with strscpy_pad() for pcm->name
by Peter Ujfalusi 11 Sep '19

11 Sep '19

While it is safe to use strncpy in this case, the advice is to move to strscpy or strscpy_pad. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index d93db2c2b527..5552c66ca642 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -308,9 +308,9 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE; if (rtd->pcm->streams[i].pcm->name[0] == '\0') { - strncpy(rtd->pcm->streams[i].pcm->name, - rtd->pcm->streams[i].pcm->id, - sizeof(rtd->pcm->streams[i].pcm->name)); + strscpy_pad(rtd->pcm->streams[i].pcm->name, + rtd->pcm->streams[i].pcm->id, + sizeof(rtd->pcm->streams[i].pcm->name)); } } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH v2] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 11 Sep '19

11 Sep '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: fsl_mqs: add DT binding documentation
by Shengjiu Wang 11 Sep '19

11 Sep '19

Add the DT binding documentation for NXP MQS driver Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- .../devicetree/bindings/sound/fsl,mqs.txt | 20 +++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt new file mode 100644 index 000000000000..a1dbe181204a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt @@ -0,0 +1,20 @@ +fsl,mqs audio CODEC + +Required properties: + + - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" + "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs". + - clocks : A list of phandles + clock-specifiers, one for each entry in + clock-names + - clock-names : Must contain "mclk" + - gpr : The gpr node. + +Example: + +mqs: mqs { + compatible = "fsl,imx6sx-mqs"; + gpr = <&gpr>; + clocks = <&clks IMX6SX_CLK_SAI1>; + clock-names = "mclk"; + status = "disabled"; +}; -- 2.21.0

3 4

[alsa-devel] [PATCH] tas2770: add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 11 Sep '19

11 Sep '19

From: Frank Shi <shifu0704(a)thundersoft.com> add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- Documentation/devicetree/bindings/tas2770.txt | 38 +++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) create mode 100644 Documentation/devicetree/bindings/tas2770.txt diff --git a/Documentation/devicetree/bindings/tas2770.txt b/Documentation/devicetree/bindings/tas2770.txt new file mode 100644 index 0000000..f70b310 --- /dev/null +++ b/Documentation/devicetree/bindings/tas2770.txt @@ -0,0 +1,38 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,left-slot: - Sets TDM RX left time slots. + - ti,right-slot: - Sets TDM RX right time slots. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + + - reset-gpio: Reset GPIO number of left device. + - irq-gpio: IRQ GPIO number of left device. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + reset-gpio = <&gpio15 1 GPIO_ACTIVE_LOW>; + irq-gpio = <&gpio16 1 GPIO_ACTIVE_LOW>; + ti,asi-format = <0>; + ti,left-slot = <0>; + ti,right-slot = <1>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

3 7

[alsa-devel] [PATCH] ASoC: dmaengine: Make the pcm->name equal to pcm->id if the name is not set
by Peter Ujfalusi 11 Sep '19

11 Sep '19

Some tools use the snd_pcm_info_get_name() to try to identify PCMs or for other purposes. Currently it is left empty with the dmaengine-pcm, in this case copy the pcm->id string as pcm->name. For example IGT is using this to find the HDMI PCM for testing audio on it. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reported-by: Arthur She <arthur.she(a)linaro.org> --- Hi, this was actually reported for 4.14 kernel with omap-pcm (replaced by sdma-pcm in v4.18), since then we only use the generic dmaengine PCM but the same issue applies today. Regards, Peter sound/soc/soc-generic-dmaengine-pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 748f5f641002..d93db2c2b527 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -306,6 +306,12 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i])) pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE; + + if (rtd->pcm->streams[i].pcm->name[0] == '\0') { + strncpy(rtd->pcm->streams[i].pcm->name, + rtd->pcm->streams[i].pcm->id, + sizeof(rtd->pcm->streams[i].pcm->name)); + } } return 0; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 7

[alsa-devel] [PATCH 00/13] ASoC: soc-core cleanup - step 3
by Kuninori Morimoto 11 Sep '19

11 Sep '19

Hi Mark These are soc-core.c cleanup (maybe) step 3. These try to clenaup around kmalloc/kfree related code. Basically these do nothing from technical point of view. Kuninori Morimoto (13): ASoC: soc-core: move soc_free_pcm_runtime() ASoC: soc-core: merge soc_add_pcm_runtime() into soc_new_pcm_runtime() ASoC: soc-core: call list_del(&rtd->list) at soc_free_pcm_runtime() ASoC: soc-core: create rtd->codec_dais first ASoC: soc-core: merge soc_new_pcm_runtime() and soc_rtd_init() ASoC: soc-core: merge soc_free_pcm_runtime() and soc_rtd_free() ASoC: soc-core: tidyup soc_new_pcm_runtime() alloc order ASoC: soc-core: remove snd_soc_rtdcom_del_all() ASoC: soc-core: use devm_kzalloc() for rtd ASoC: soc-core: remove soc_remove_dai_links() ASoC: soc-core: add soc_setup_card_name() ASoC: soc-core: use devm_xxx for component related resource ASoC: soc-core: setup card->rtd_num at snd_soc_instantiate_card() include/sound/soc.h | 1 - sound/soc/soc-core.c | 276 +++++++++++++++++++++++++-------------------------- 2 files changed, 138 insertions(+), 139 deletions(-) -- 2.7.4

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[alsa-devel] Measuring Device Throughput - MMAP vs. RW
by Pavel Hofman 10 Sep '19

10 Sep '19

Hi, Gathering data for USB async gadget feedback ( https://mailman.alsa-project.org/pipermail/alsa-devel/2019-August/154819.ht… ) requires measuring average data rate to/from a device. For testing I added measuring/averaging code to methods snd_pcm_hw_writei, snd_pcm_hw_readi, and snd_pcm_hw_mmap_commit in https://github.com/alsa-project/alsa-lib/blob/master/src/pcm/pcm_hw.c . In all cases I take the parameter "size" of all the three methods as the number of samples written/read. RW: The averaging works very well for R/W access (hw:X) for playback and capture, I get same values on the same device (playback/capture timed by one clock). MMAP: When mmap is used (e.g. plughw:X + sample format conversion), I get very stable results for MMAP playback on all soundcards I have tested. But for capture I get identical results to playback for one soundcard (PCI Envy24), while the averaged rate calculated from "size" in snd_pcm_hw_mmap_commit fluctuates for capture on Intel HDA. The averaging is simply summing "size" in every call of the method and dividing by passed nanosecs every approx 10 seconds: Working OK (Infrasonic Quartet, and all soundcards for read/write mode): Playback MMAP Time: 19440.900391, averaged samplerate: 48000.511829 MMAP Time: 19451.140625, averaged samplerate: 48000.360218 MMAP Time: 19461.380859, averaged samplerate: 48000.176846 MMAP Time: 19471.619141, averaged samplerate: 48000.664570 MMAP Time: 19481.861328, averaged samplerate: 48000.199970 MMAP Time: 19492.101562, averaged samplerate: 48000.589854 MMAP Time: 19502.341797, averaged samplerate: 48000.288977 MMAP Time: 19512.580078, averaged samplerate: 48000.516104 MMAP Time: 19522.820312, averaged samplerate: 48000.225282 Capture: MMAP Time: 19454.578125, averaged samplerate: 48000.430766 MMAP Time: 19464.818359, averaged samplerate: 48000.397611 MMAP Time: 19475.056641, averaged samplerate: 48000.400049 MMAP Time: 19485.296875, averaged samplerate: 48000.293275 MMAP Time: 19495.539062, averaged samplerate: 48000.549048 MMAP Time: 19505.779297, averaged samplerate: 48000.445214 MMAP Time: 19516.017578, averaged samplerate: 48000.417299 MMAP Time: 19526.257812, averaged samplerate: 48000.376264 MMAP Time: 19536.498047, averaged samplerate: 48000.422957 MMAP Time: 19546.738281, averaged samplerate: 48000.538619 MMAP Time: 19556.976562, averaged samplerate: 48000.241647 MMAP Time: 19567.216797, averaged samplerate: 48000.447243 Fluctuating Capture (Intel HDA): Playback - OK: MMAP Time: 19574.498047, averaged samplerate: 47998.971874 MMAP Time: 19584.568359, averaged samplerate: 47998.871734 MMAP Time: 19594.636719, averaged samplerate: 47998.929850 MMAP Time: 19604.707031, averaged samplerate: 47998.972555 MMAP Time: 19614.775391, averaged samplerate: 47998.925531 MMAP Time: 19624.845703, averaged samplerate: 47998.893351 MMAP Time: 19634.916016, averaged samplerate: 47998.949489 MMAP Time: 19644.986328, averaged samplerate: 47998.593678 MMAP Time: 19655.056641, averaged samplerate: 47999.230857 MMAP Time: 19665.125000, averaged samplerate: 47998.777034 MMAP Time: 19675.195312, averaged samplerate: 47999.177460 MMAP Time: 19685.263672, averaged samplerate: 47998.798503 Capture - fluctuating, clearly some samples are not counted: MMAP Time: 19539.853516, averaged samplerate: 47191.039837 MMAP Time: 19549.923828, averaged samplerate: 46369.921890 MMAP Time: 19559.994141, averaged samplerate: 47998.831508 MMAP Time: 19570.148438, averaged samplerate: 47190.998307 MMAP Time: 19580.302734, averaged samplerate: 47191.275432 MMAP Time: 19590.373047, averaged samplerate: 47998.894343 MMAP Time: 19600.443359, averaged samplerate: 47998.906779 MMAP Time: 19610.511719, averaged samplerate: 47999.174771 MMAP Time: 19620.582031, averaged samplerate: 47998.747971 MMAP Time: 19630.650391, averaged samplerate: 47998.881964 MMAP Time: 19640.722656, averaged samplerate: 47998.999974 MMAP Time: 19650.791016, averaged samplerate: 47998.769240 I suspect the size parameter of snd_pcm_hw_mmap_commit is not the actual amount of samples read during the MMAP capture. Please how to calculate the number of samples read from the variables available in that method? Thanks a lot. Pavel.

2 7

[alsa-devel] [PATCH 0/2] ALSA: firewire-tascam: fix to handle error code to read status of sampling clock
by Takashi Sakamoto 10 Sep '19

10 Sep '19

Hi, ALSA firewire-tascam driver has a bug not to check return value from a call of snd_tscm_stream_get_clock() since its first commit. This patchset fixes the bug. In models of TASCAM FIreWire series, the status of sampling clock is represented in 2 bytes in MSB of a certain register. These bits are zero in intermediate state after configuring the sampling clock. This behaviour sometimes takes ALSA PCM applications to receive error when starting both playback/capture PCM substreams because the operation to start one PCM substream can change the sampling clock. In this case, next operation for another PCM substream possible fails. This patchset adds check and retry for the above case. Although the change is a bit intrusive for stable releases, it's required to avoid superficial regression due to the additional error check. Takashi Sakamoto (2): ALSA: firewire-tascam: handle error code when getting current source of clock ALSA: firewire-tascam: check intermediate state of clock status and retry sound/firewire/tascam/tascam-pcm.c | 3 ++ sound/firewire/tascam/tascam-stream.c | 42 ++++++++++++++++++--------- 2 files changed, 31 insertions(+), 14 deletions(-) -- 2.20.1

2 4

[alsa-devel] [GIT PULL] ASoC updates v5.4
by Mark Brown 10 Sep '19

10 Sep '19

The following changes since commit f74c2bb98776e2de508f4d607cd519873065118e: Linux 5.3-rc8 (2019-09-08 13:33:15 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-v5.4 for you to fetch changes up to bb831786117519fc16dfd3eaa7b84e4f6bbb8d99: Merge branch 'asoc-5.4' into asoc-next (2019-09-09 14:55:20 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.4 Quite a big update this time around, particularly in the core where we've had a lot of cleanups from Morimoto-san - there's not much functional change but quite a bit of modernization going on. We've also seen a lot of driver work, a lot of it cleanups but also some particular drivers. - Lots and lots of cleanups from Morimoto-san and Yue Haibing. - Lots of cleanups and enhancements to the Freescale, sunxi dnd Intel rivers. - Initial Sound Open Firmware suppot for i.MX8. - Removal of w90x900 and nuc900 drivers as the platforms are being removed. - New support for Cirrus Logic CS47L15 and CS47L92, Freescale i.MX 7ULP and 8MQ, Meson G12A and NXP UDA1334 ---------------------------------------------------------------- Amadeusz Sławiński (9): ASoC: Intel: Skylake: Combine snd_soc_skl_ipc and snd_soc_skl ASoC: Intel: Skylake: Do not disable FW notifications ASoC: Intel: Skylake: Remove static table index when parsing topology ASoC: Intel: Skylake: Use correct function to access iomem space ASoC: Intel: Fix use of potentially uninitialized variable ASoC: dapm: Expose snd_soc_dapm_new_control_unlocked properly ASoC: Intel: NHLT: Fix debug print format ASoC: Intel: Skylake: Print module type instead of id ASoC: Intel: Skylake: Release topology when we are done with it Andra Danciu (2): dt-bindings: sound: Add bindings for UDA1334 codec ASoC: codecs: Add uda1334 codec driver Andy Shevchenko (1): ASoC: Intel: Skylake: Print constant literals from format specifier Arnd Bergmann (1): ASoC: remove w90x900/nuc900 platform drivers Bard liao (1): ASoC: SOF: rename SOUNDWIRE to ALH Ben Zhang (1): ASoC: rt5677: keep analog power register at SND_SOC_BIAS_OFF Bjorn Andersson (1): ASoC: qcom: common: Include link-name in error messages Cezary Rojewski (13): ASoC: Intel: Skylake: Merge skl_sst and skl into skl_dev struct ASoC: Intel: Skylake: Remove MCPS available check ASoC: Intel: Skylake: Remove memory available check ASoC: Intel: Skylake: Make MCPS and CPS params obsolete ASoC: Intel: Skylake: Cleanup skl_module_cfg declaration ASoC: Intel: Update request-reply IPC model ASoC: Intel: Skylake: Limit large_config_get to single frame ASoC: Intel: Skylake: large_config_get overhaul ASoC: Intel: Baytrail: Fix implicit fallthrough warning ASoC: Intel: Haswell: Adjust machine device private context ASoC: Intel: haswell: Simplify device probe ASoC: Intel: bdw-rt5677: Simplify device probe ASoC: Intel: broadwell: Simplify device probe Charles Keepax (3): ASoC: cs47l35: Use define for DSPCLK enable bit ASoC: madera: Read device tree configuration ASoC: wm_adsp: Allow bus error handler to be called directly Cheng-Yi Chiang (2): ASoC: rockchip: rockchip_max98090: Set period size to 240 ASoC: hdmi-codec: Add an op to set callback function for plug event Christophe JAILLET (4): ASoC: pcm3168a: Fix a typo in the name of a constant ASoC: rt5665: Fix a typo in the name of a function ASoC: wm8955: Fix a typo in 'wm8995_pll_factors()' function name ASoC: cs4271: Fix a typo in the CS4171_NR_RATIOS Codrin Ciubotariu (4): ASoC: codecs: ad193x: Use regmap_multi_reg_write() when initializing ASoC: mchp-i2s-mcc: Fix unprepare of GCLK ASoC: mchp-i2s-mcc: Wait for RX/TX RDY only if controller is running ASoC: mchp-i2s-mcc: Fix simultaneous capture and playback in master mode Colin Ian King (6): ASoC: codec2codec: fix missing return of error return code ASoC: zx-tdm: remove redundant assignment to ts_width on error return path ASoC: codecs: ad193x: make two arrays static const, makes object smaller ASoC: rt1011: make array pd static const, makes object smaller ASoC: rt1305: make array pd static const, makes object smaller ASoC: rt1308: make array pd static const, makes object smaller Curtis Malainey (3): ASoC: rt5677: remove superfluous set ASoC: rt5677: Revert remove superfluous set ASoC: rt5677: Remove magic number register writes Dan Carpenter (1): ASoC: SOF: imx8: Fix an is IS_ERR() vs NULL check Daniel Baluta (20): ASoC: SOF: Introduce snd_sof_dsp_get_bar_index ops ASoC: SOF: loader: Use the BAR provided by FW ASoC: SOF: loader: Don't ignore SRAM block types ASoC: dt-bindings: Introduce compatible strings for 7ULP and 8MQ ASoC: fsl_sai: Add registers definition for multiple datalines ASoC: fsl_sai: Update Tx/Rx channel enable mask ASoC: fsl_sai: Add support for SAI new version ASoC: fsl_sai: Add support for imx7ulp/imx8mq ASoC: SOF: Remove call to snd_sof_dsp_mailbox_init ASoC: SOF: Intel: bdw: Use generic function for fw ready / mem windows creation ASoC: SOF: Intel: hda: Use generic function for fw ready / mem windows creation ASoC: SOF: Introduce snd_sof_dsp_get_mailbox_offset ASoC: SOF: Introduce snd_sof_dsp_get_window_offset ASoC: SOF: Intel: byt: Refactor fw ready / mem windows creation ASoC: SOF: topology: Add dummy support for i.MX8 DAIs ASoC: fsl_sai: Add support for imx8qm ASoC: dt-bindings: Introduce compatible string for imx8qm dt-bindings: dsp: fsl: Add DSP core binding support ASoC: SOF: Add OF DSP device support ASoC: SOF: imx: Add i.MX8 HW support Daniel Stuart (2): ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk Enric Balletbo i Serra (1): ASoC: rockchip-max98090: Remove MICBIAS as supply of input pin IN34 Fred Oh (1): ASoC: SOF: remove unused state variable in suspend function Greg Kroah-Hartman (3): ASoC: Intel: SoC: skylake: no need to check return value of debugfs_create functions ASoC: core: no need to check return value of debugfs_create functions ASoC: SOF: no need to check return value of debugfs_create functions Guennadi Liakhovetski (1): ASoC: SOF: Intel: hda: add a parameter to disable MSI Gustavo A. R. Silva (1): ASoC: ti: Mark expected switch fall-throughs Hariprasad Kelam (3): sound: soc: codecs: mt6358: change return type of mt6358_codec_init_reg ASoC: bcm: cygnus-pcm: Unneeded variable: "ret". ASoC: wcd9335: add irqflag IRQF_ONESHOT flag Janusz Jankowski (1): ASoC: SOF: Intel: ssp: BCLK delay parameter Janusz Krzysztofik (1): ASoC: ams-delta: Take control over audio mute GPIO pins Jaroslav Kysela (1): ASoC: SOF: Makefile - fix the top-level kernel module names (add snd- prefix) Jaska Uimonen (2): ASoC: SOF: topology: use set_get_data in process load ASoC: SOF: topology: fix get control data return type and arguments Jerome Brunet (17): ASoC: codec2codec: run callbacks in order ASoC: codec2codec: name link using stream direction ASoC: codec2codec: deal with params when necessary ASoC: meson: g12a-tohdmitx: override codec2codec params ASoC: create pcm for codec2codec links as well ASoC: codec2codec: remove ephemeral variables ASoC: codec2codec: fill some of the runtime stream parameters ASoC: meson: axg-tdm-formatter: free reset on device removal ASoC: meson: g12a-tohdmitx: require regmap mmio ASoC: meson: add sm1 compatibles ASoC: meson: add reset binding ASoC: meson: axg-frddr: expose all 8 outputs ASoC: meson: axg-toddr: expose all 8 inputs ASoC: meson: tdmin: expose all 16 inputs ASoC: meson: axg-frddr: add sm1 support ASoC: meson: axg-toddr: add sm1 support ASoC: meson: tdmout: add sm1 support Jiaxin Yu (5): ASoC: mediatek: mt6358: add delay after dmic clock on ASoC: mediatek: mt8183: switch tdm pins gpio function when playback on or off ASoC: mediatek: mt8183: fix tdm out data align issue ASoC: mediatek: mt8183: tdm hw support tdm out and 8ch i2s out ASoC: mediatek: mt8183: fix tdm out data is valid on rising edge Joe Perches (1): ASoC: wcd9335: Fix misuse of GENMASK macro Kai Vehmanen (5): ASoC: SOF: reset DMA state in prepare ASoC: SOF: use common code to send PCM_FREE IPC ASoC: SOF: ipc: use timeout configured at probe ASoC: SOF: core: increase default IPC timeouts ASoC: SOF: Intel: hda: reset link DMA state in prepare Katsuhiro Suzuki (8): ASoC: es8316: fix headphone mixer volume table ASoC: es8316: fix inverted L/R of headphone mixer volume ASoC: es8316: add DT-bindings ASoC: es8316: add clock control of MCLK ASoC: es8316: fix redundant codes of clock ASoC: es8316: support fixed and variable both clock rates ASoC: rockchip: ignore 0Hz sysclk SoC: simple-card-utils: set 0Hz to sysclk when shutdown Keyon Jie (5): ASoC: SOF: Intel: hda: correct ROM state mask ASoC: hdac_hda: fix page fault issue by removing race ASoC: Intel: sof-rt5682: add dmic dapm widget to support dmic PCM ASoC: Intel: skl-hda-dsp-generic: add dependency to dmic driver ASoC: Intel: skl-hda-dsp-generic: add dmic dapm widget and route Kunihiko Hayashi (1): ASoC: uniphier: Fix double reset assersion when transitioning to suspend state Kuninori Morimoto (101): ASoC: add soc-dai.c ASoC: soc-dai: mv soc_dai_hw_params() to soc-dai ASoC: soc-dai: add snd_soc_dai_hw_free() ASoC: soc-dai: add snd_soc_dai_startup() ASoC: soc-dai: add snd_soc_dai_shutdown() ASoC: soc-dai: add snd_soc_dai_prepare() ASoC: soc-dai: add snd_soc_dai_trigger() ASoC: soc-dai: add snd_soc_dai_bespoke_trigger() ASoC: soc-dai: add snd_soc_dai_delay() ASoC: soc-dai: add snd_soc_dai_suspend() ASoC: soc-dai: add snd_soc_dai_resume() ASoC: soc-dai: add snd_soc_dai_probe() ASoC: soc-dai: add snd_soc_dai_remove() ASoC: soc-dai: add snd_soc_dai_compress_new() ASoC: soc-core: use existing snd_soc_dai_digital_mute() ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c ASoC: soc-pcm: remove soc_rtdcom_ack() ASoC: soc-pcm: remove soc_rtdcom_copy_kernel() ASoC: soc-pcm: remove soc_fill_silence() ASoC: add soc-component.c ASoC: soc-component: add snd_soc_component_get/put() ASoC: soc-component: add snd_soc_component_open() ASoC: soc-component: add snd_soc_component_close() ASoC: soc-component: add snd_soc_component_prepare() ASoC: soc-component: add snd_soc_component_hw_params() ASoC: soc-component: add snd_soc_component_hw_free() ASoC: soc-component: add snd_soc_component_trigger() ASoC: soc-component: add snd_soc_component_suspend() ASoC: soc-component: add snd_soc_component_resume() ASoC: soc-component: add snd_soc_component_is_suspended() ASoC: soc-component: add snd_soc_component_probe() ASoC: soc-component: add snd_soc_component_remove() ASoC: soc-component: add snd_soc_component_of_xlate_dai_id() ASoC: soc-component: add snd_soc_component_of_xlate_dai_name() ASoC: soc-component: move snd_soc_component_seq_notifier() ASoC: soc-component: move snd_soc_component_stream_event() ASoC: soc-component: move snd_soc_component_set_bias_level() ASoC: soc-component: add snd_soc_pcm_component_pointer() ASoC: soc-component: add snd_soc_pcm_component_ioctrl() ASoC: soc-component: add snd_soc_pcm_component_copy_user() ASoC: soc-component: add snd_soc_pcm_component_page() ASoC: soc-component: add snd_soc_pcm_component_mmap() ASoC: soc-component: add snd_soc_pcm_component_pcm_new() ASoC: soc-component: add snd_soc_pcm_component_pcm_free() ASoC: rsnd: don't call clk_get_rate() under atomic context ASoC: soc-core: set component->debugfs_root NULL ASoC: soc-core: add comment for for_each_xxx ASoC: soc-core: check return value of snd_soc_add_dai_link() ASoC: soc-core: don't use for_each_card_links_safe() at snd_soc_find_dai_link() ASoC: soc-core: reuse rtdcom at snd_soc_rtdcom_add() ASoC: soc-core: tidyup for snd_soc_dapm_new_controls() ASoC: soc-core: tidyup for snd_soc_add_component_controls() ASoC: soc-core: tidyup for snd_soc_dapm_add_routes() ASoC: soc-core: tidyup for snd_soc_add_card_controls() ASoC: soc-core: call snd_soc_dapm_debugfs_init() at soc_init_card_debugfs() ASoC: soc-core: remove unneeded list_empty() check for snd_soc_try_rebind_card() ASoC: soc-core: tidyup for card->deferred_resume_work ASoC: soc-core: define soc_dpcm_debugfs_add() for non CONFIG_DEBUG_FS ASoC: soc-core: dai_link check under soc_dpcm_debugfs_add() ASoC: soc-core: support snd_soc_dai_link_component for aux_dev ASoC: simple-card: use snd_soc_dai_link_component for aux_dev ASoC: intel: cht_bsw_max98090_ti: use snd_soc_dai_link_component for aux_dev ASoC: mediatek: mt8183-da7219-max98357: use snd_soc_dai_link_component for aux_dev ASoC: meson: axg-card: use snd_soc_dai_link_component for aux_dev ASoC: rockchip: rockchip_max98090: use snd_soc_dai_link_component for aux_dev ASoC: samsung: neo1973_wm8753: use snd_soc_dai_link_component for aux_dev ASoC: samsung: speyside: use snd_soc_dai_link_component for aux_dev ASoC: samsung: tm2_wm5110: use snd_soc_dai_link_component for aux_dev ASoC: sunxi: sun4i-codec: use snd_soc_dai_link_component for aux_dev ASoC: ti: rx51: use snd_soc_dai_link_component for aux_dev ASoC: soc-core: remove legacy style of aux_dev ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: use snd_soc_dai_link_component for aux_dev ASoC: soc-core: add for_each_xxx macro for aux_dev ASoC: soc-core: merge snd_soc_initialize_card_lists() ASoC: soc-core: remove unneeded dai_link check from snd_soc_remove_dai_link() ASoC: soc-core: use device_register() ASoC: soc-core: add NOTE to snd_soc_rtdcom_lookup() ASoC: soc-core: soc_cleanup_card_resources() become void ASoC: soc-dai: use bit field for bus_control ASoC: soc-topology: use for_each_component_dais() at remove_dai() ASoC: soc-core: initialize component list ASoC: soc-core: initialize list at one place ASoC: soc-core: rename soc_post_component_init() to soc_rtd_init() ASoC: soc-core: add soc_rtd_free() ASoC: soc-core: move soc_probe_component() position ASoC: soc-core: dapm related setup at one place ASoC: soc-core: add snd_soc_dapm_init() ASoC: soc-core: move soc_probe_link_components() position ASoC: simple-card: indicate rebind issue ASoC: audio-graph: indicate rebind issue ASoC: soc-core: add comment to jack at soc_remove_component() ASoC: soc-core: self contained soc_probe_link_components() ASoC: soc-core: self contained soc_remove_link_components() ASoC: soc-core: self contained soc_remove_link_dais() ASoC: soc-core: move soc_probe_dai() next to soc_remove_dai() ASoC: soc-core: add new soc_link_init() ASoC: soc-core: self contained soc_probe_link_dais() ASoC: soc-core: move soc_probe_link_dais() next to soc_remove_link_dais() ASoC: soc-core: self contained soc_bind_aux_dev() ASoC: soc-core: add soc_unbind_aux_dev() ASoC: soc-core: self contained soc_unbind_aux_dev() Libin Yang (1): ASoC: intel: skl_hda_dsp_common: create HDMI jack kctl Lucas Stach (4): ASoC: fsl_sai: add of_match data ASoC: fsl_sai: derive TX FIFO watermark from FIFO depth ASoC: fsl_sai: mark regmap as fast_io ASoC: tlv320aic31xx: suppress error message for EPROBE_DEFER Mac Chiang (1): ASoC: Intel: boards: Add Cometlake machine driver support Mark Brown (8): Merge branch 'asoc-5.3' into asoc-5.4 Merge branch 'topic/hda-dmic' of https://git.kernel.org/.../tiwai/sound into asoc-5.4 ASoC: sof: Fix warning when IPC flood test is not enabled Merge branch 'topic/hda-bus-ops-cleanup' of https://git.kernel.org/.../tiwai/sound into asoc-5.4 Merge branch 'topic/hda-bus-ops-cleanup' of https://git.kernel.org/.../tiwai/sound into asoc-5.4 Merge branch 'asoc-5.3' into asoc-5.4 Merge branch 'asoc-5.3' into asoc-linus Merge branch 'asoc-5.4' into asoc-next Masanari Iida (1): ASoC: ti: Fix typos in ti/Kconfig Mauro Carvalho Chehab (1): ASoC: sunxi: fix a sound binding broken reference Maxime Ripard (28): ASoC: sun4i-i2s: Register regmap and PCM before our component ASoC: sun4i-i2s: Switch to devm for PCM register ASoC: sun4i-i2s: Don't use the oversample to calculate BCLK ASoC: sun4i-i2s: Replace call to params_channels by local variable ASoC: sun4i-i2s: Rework MCLK divider calculation ASoC: sun4i-i2s: Move the channel configuration to a callback ASoC: sun4i-i2s: Move the format configuration to a callback ASoC: sun4i-i2s: Use module clock as BCLK parent on newer SoCs ASoC: sun4i-i2s: Fix the MCLK and BCLK dividers on newer SoCs ASoC: sun4i-i2s: RX and TX counter registers are swapped ASoC: sun4i-i2s: Fix LRCK and BCLK polarity offsets on newer SoCs ASoC: sun4i-i2s: Fix the LRCK polarity ASoC: sun4i-i2s: Fix WSS and SR fields for the A83t ASoC: sun4i-i2s: Fix MCLK Enable bit offset on A83t ASoC: sun4i-i2s: Fix the LRCK period on A83t ASoC: sun4i-i2s: Remove duplicated quirks structure ASoC: sun4i-i2s: Pass the channels number as an argument ASoC: sun4i-i2s: Support more channels ASoC: sun4i-i2s: Add support for TDM slots ASoC: sun4i-i2s: Use the physical / slot width for the clocks ASoC: sun4i-i2s: Use the actual format width instead of an hardcoded one ASoC: sun4i-i2s: Replace call to params_width by local variable ASoC: sun4i-i2s: Add support for DSP formats ASoC: sunxi: Revert initial A83t support ASoC: sun4i: Revert A83t description ASoC: dt-bindings: sun4i-spdif: Fix dma-names warning ASoC: dt-bindings: Convert Allwinner A33 codec to a schema ASoC: dt-bindings: Convert Allwinner A64 analog codec to a schema Michał Mirosław (5): ASoC: wm8904: fix typo in DAPM kcontrol name ASoC: wm8904: implement input mode select as a mux ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig ASoC: atmel_ssc_dai: rework DAI format configuration ASoC: atmel_ssc_dai: implement left-justified data mode Oleksandr Suvorov (6): ASoC: Define a set of DAPM pre/post-up events ASoC: sgtl5000: Improve VAG power and mute control ASoC: sgtl5000: Fix definition of VAG Ramp Control ASoC: sgtl5000: add ADC mute control ASoC: sgtl5000: Fix of unmute outputs on probe ASoC: sgtl5000: Fix charge pump source assignment Pan Xiuli (5): ASoC: SOF: pci: mark last_busy value at runtime PM init ASoC: Intel: common: add ACPI matching tables for Tiger Lake ASoC: SOF: Intel: initial support for Tiger Lake. ASoC: Intel: common: add ACPI matching tables for EHL ASoC: SOF: Intel: initial support for Elkhart Lake Peter Ujfalusi (12): ASoC: pcm3168a: Allow all channels in case of parallel DIN/DOUT setup ASoC: ti: davinci-mcasp: Add support for RIGHT_J format ASoC: ti: davinci-mcasp: Improve serializer handling in multi AXR setups ASoC: ti: davinci-mcasp: Support for correct symmetric sample bits ASoC: pcm3168a: Retain the independence of DAC and ADC side of the codec ASoC: pcm3168a: Allow reconfiguration of tdm_slots and slot_width ASoC: core: Move pcm_mutex up to card level from snd_soc_pcm_runtime ASoC: ti: davinci-i2s: Move the XSYNCERR workaround to .prepare callback ASoC: ti: davinci-i2s: Add S32_LE as support format ASoC: ti: edma-pcm: Fix for legacy dma_slave_map based channel lookup ASoC: ti: davinci-evm: Do not fail if the dai_set_sysclk returns -ENOTSUPP ASoC: dmaengine: Make the pcm->name equal to pcm->id if the name is not set Pierre-Louis Bossart (14): ASoC: Intel: Skylake: move NHLT header to common directory ALSA: hda: move parts of NHLT code to new module ALSA: hda: intel-nhlt: handle NHLT VENDOR_DEFINED DMIC geometry ASoC: Intel: Skylake: use common NHLT module ALSA: hda/intel: stop probe if DMICS are detected on Skylake+ platforms ASoC: SOF: Intel: hda: fix MSI handling ASoC: soc-core: fix module_put() warning in soc_cleanup_component ASoC: SOF: Intel: hda: fixup HDaudio topology name with DMIC number ASoC: Intel: hsw: remove i386 build warning w/ size_t argument ASoC: SOF: fix HDA direct MMIO access ASoC: SOF: ipc: add ALH parameters ASoC: SOF: topology: initial support for Intel ALH DAI type ASoC: SOF: pcm: add ALH support ASoC: SOF/Intel: fix selection of SND_INTEL_NHTL Rander Wang (4): ASoC: SOF: Intel: hda: reduce ifdef usage for hda ASoC: SOF: Intel: hda: Enable jack detection ASoC: SOF: Intel: hda: fix link DMA config ASoC: SOF: Intel: hda: fix stream id setting Ranjani Sridharan (2): ASoC: SOF: Intel: hda: Make hdac_device device-managed ASoC: SOF: Intel: hda: Initialize HDA controller after i915 init Richard Fitzgerald (1): ASoC: cs47l15: Add codec driver for Cirrus Logic CS47L15 Sathya Prakash M R (1): ASoC: Intel: boards: Match Product Family instead of product Shengjiu Wang (8): ASoC: fsl_esai: Wrap some operations to be functions ASoC: fsl_esai: recover the channel swap after xrun ASoC: fsl_esai: Add compatible string for imx6ull ASoC: fsl_esai: Add new compatible string for imx6ull ASoC: imx-audmux: Add driver suspend and resume to support MEGA Fast ASoC: imx-audmix: register the card on a proper dev ASoC: cs42xx8: Force suspend/resume during system suspend/resume ASoC: fsl_ssi: Fix clock control issue in master mode Shuming Fan (6): ASoC: rt1308: simplify the EFUSE read function ASoC: rt1308: add silence detection and manual PDB control ASoC: rt1308: Convert headers to SPDX ASoC: rt1011: Add R0 temperature and TDM1 ADC2DAT Swap control ASoC: rt1011: Add a flag for the R0 calibration test ASoC: rt1011: add mutex protection to set_fmt/set_tdm_slot Srinivas Kandagatla (1): ASoC: wcd9335: Fix primary interpolator max rate Stephen Boyd (1): ASoC: Remove dev_err() usage after platform_get_irq() Stephen Rothwell (1): ALSA: hda: readl/writel need linux/io.h Stuart Henderson (1): ASoC: cs47l92: Add codec driver for Cirrus Logic CS47L92 Szymon Mielczarek (1): ASoC: dapm: Invalidate only paths reachable for a given stream Takashi Iwai (4): ALSA: hda: Remove page allocation redirection ALSA: hda: Direct MMIO accesses ASoC: SOF: Intel: Initialize hdaudio bus properly ASoC: hdac_hdmi: Offload dapm update at jack detection Timo Wischer (1): ASoC: rsnd: Support hw_free() callback at DAI level Tzung-Bi Shih (6): ASoC: mediatek: mt8183: make headset codec optional ASoC: mediatek: mt8183: remove unused DAPM-related variables ASoC: mediatek: mt8183: move private structure ASoC: mediatek: mt8183: move headset jack to card-specific storage ASoC: mediatek: mt8183: remove forward declaration of headset_init ASoC: mediatek: mt8183: fix space issues Vidyakumar Athota (1): ALSA: pcm: add support for 352.8KHz and 384KHz sample rate Wei Yongjun (1): ASoC: SOF: imx8: Fix return value check in imx8_probe() YueHaibing (66): ASoC: tegra20_das: use devm_platform_ioremap_resource() to simplify code ASoC: tegra: use devm_platform_ioremap_resource() to simplify code ASoC: rockchip: use devm_platform_ioremap_resource() to simplify code ASoC: ep93xx-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: mt8173: use devm_platform_ioremap_resource() to simplify code ASoC: mt6797: use devm_platform_ioremap_resource() to simplify code ASoC: imx-audmux: use devm_platform_ioremap_resource() to simplify code ASoC: fsl_audmix: use devm_platform_ioremap_resource() to simplify code ASoC: bcm2835-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: sirf: use devm_platform_ioremap_resource() to simplify code ASoC: mxs-saif: use devm_platform_ioremap_resource() to simplify code ASoC: spear: use devm_platform_ioremap_resource() to simplify code ASoC: kirkwood-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: xtfpga-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-pdm: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-spdifin: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-spdifout: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-fifo: use devm_platform_ioremap_resource() to simplify code ASoC: xlnx: use devm_platform_ioremap_resource() to simplify code ASoC: sun8i-codec-analog: use devm_platform_ioremap_resource() to simplify code ASoC: sunxi: sun50i-codec-analog: use devm_platform_ioremap_resource() to simplify code ASoC: sun8i-codec: use devm_platform_ioremap_resource() to simplify code ASoC: stm32: sai: use devm_platform_ioremap_resource() to simplify code ASoC: codecs: msm8916-wcd: use devm_platform_ioremap_resource() to simplify code ASoC: codecs: jz4725b: use devm_platform_ioremap_resource() to simplify code ASoC: mmp-sspa: use devm_platform_ioremap_resource() to simplify code ASoC: jz4740: use devm_platform_ioremap_resource() to simplify code ASoC: inno_rk3036: use devm_platform_ioremap_resource() to simplify code ASoC: uniphier: evea: use devm_platform_ioremap_resource() to simplify code ASoC: uniphier: aio-dma: use devm_platform_ioremap_resource() to simplify code ASoC: psc-ac97: use devm_platform_ioremap_resource() to simplify code ASoC: au1x: psc-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: meson: g12a-tohdmitx: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-tdm-formatter: use devm_platform_ioremap_resource() to simplify code ASoC: tscs454: remove unused variable 'PLL_48K_RATE' ASoC: SOF: Intel: Add missing include file hdac_hda.h ASoC: mt6351: remove unused variable 'mt_lineout_control' ASoC: ml26124: remove unused variable 'ngth' ASoC: max98926: remove two unused variables ASoC: max9850: remove unused variable 'max9850_reg' ASoC: es8328: Fix copy-paste error in es8328_right_line_controls ASoC: cs4349: Use PM ops 'cs4349_runtime_pm' ASoC: wm8737: Fix copy-paste error in wm8737_snd_controls ASoC: soc-core: Fix -Wunused-const-variable warning ASoC: mt2701: remove unused variables ASoC: mediatek: mt8183-da7219-max98357: remove unused variable ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: remove unused variables ASoC: cs42l73: remove unused variables 'vsp_output_mux' and 'xsp_output_mux' ASoC: cs4349: Use PM ops 'cs4349_runtime_pm' ASoC: max98371: remove unused variable 'max98371_noload_gain_tlv' ASoC: rt1011: remove unused variable 'dac_vol_tlv' and 'adc_vol_tlv' ASoC: tlv320aic23: remove unused variable 'tlv320aic23_rec_src' ASoC: tlv320aic31xx: remove unused variable 'cm_m_enum' ASoC: es8328: remove unused variable 'pga_tlv' ASoC: es8328: Fix copy-paste error in es8328_right_line_controls ASoC: cs42l56: remove unused variable 'adc_swap_enum' ASoC: 88pm860x: remove unused variables 'pcm_switch_controls' and 'aif1_mux' ASoC: AMD: Fix Kconfig warning without GPIOLIB ASoC: wm8988: fix typo in wm8988_right_line_controls ASoC: mediatek: mt2701: Fix -Wunused-const-variable warnings ASoC: SOF: imx8: Make some functions static ASoC: amd: use devm_platform_ioremap_resource() to simplify code ASoC: ep93xx: use devm_platform_ioremap_resource() to simplify code ASoC: sirf-audio: use devm_platform_ioremap_resource() to simplify code ASoC: tegra: use devm_platform_ioremap_resource() to simplify code ASoC: SOF: imx8: Fix COMPILE_TEST error Zhu Yingjiang (3): ASoC: SOF: Intel: hda: set position buffer in init chip ASoC: SOF: Intel: hda: use SOF defined init chip in resume ASoC: SOF: Intel: hda: remove duplicated clear WAKESTS fengchunguo (1): ASoC: max98383: fix i2c probe failure Documentation/devicetree/bindings/dsp/fsl,dsp.yaml | 88 + .../bindings/sound/allwinner,sun4i-a10-spdif.yaml | 4 +- .../sound/allwinner,sun50i-a64-codec-analog.yaml | 39 + .../bindings/sound/allwinner,sun8i-a33-codec.yaml | 57 + .../devicetree/bindings/sound/amlogic,axg-fifo.txt | 9 +- .../devicetree/bindings/sound/amlogic,axg-pdm.txt | 6 +- .../bindings/sound/amlogic,axg-spdifin.txt | 6 +- .../bindings/sound/amlogic,axg-spdifout.txt | 6 +- .../bindings/sound/amlogic,axg-tdm-formatters.txt | 4 +- .../bindings/sound/amlogic,g12a-tohdmitx.txt | 5 +- .../devicetree/bindings/sound/everest,es8316.txt | 23 + .../devicetree/bindings/sound/fsl,esai.txt | 7 +- .../devicetree/bindings/sound/fsl-sai.txt | 4 +- .../bindings/sound/sun50i-codec-analog.txt | 14 - .../devicetree/bindings/sound/sun8i-a33-codec.txt | 63 - .../devicetree/bindings/sound/uda1334.txt | 17 + arch/arm/mach-omap1/board-ams-delta.c | 10 +- include/sound/hdaudio.h | 69 +- include/sound/hdaudio_ext.h | 1 - include/sound/hdmi-codec.h | 17 + .../skl-nhlt.h => include/sound/intel-nhlt.h | 51 +- include/sound/pcm.h | 5 + include/sound/soc-acpi-intel-match.h | 2 + include/sound/soc-component.h | 387 ++++ include/sound/soc-dai.h | 28 +- include/sound/soc-dapm.h | 14 +- include/sound/soc-dpcm.h | 9 +- include/sound/soc.h | 348 +--- include/sound/sof/dai-intel.h | 12 + include/sound/sof/dai.h | 5 +- include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/tokens.h | 9 + sound/core/pcm_native.c | 2 +- sound/hda/Kconfig | 8 + sound/hda/Makefile | 3 + sound/hda/ext/hdac_ext_bus.c | 60 +- sound/hda/hdac_bus.c | 37 +- sound/hda/hdac_controller.c | 18 +- sound/hda/hdac_stream.c | 8 +- sound/hda/intel-nhlt.c | 107 + sound/pci/hda/Kconfig | 11 + sound/pci/hda/hda_codec.c | 8 +- sound/pci/hda/hda_controller.c | 6 +- sound/pci/hda/hda_controller.h | 3 +- sound/pci/hda/hda_intel.c | 105 +- sound/pci/hda/hda_tegra.c | 84 +- sound/soc/Kconfig | 1 - sound/soc/Makefile | 3 +- sound/soc/amd/Kconfig | 2 +- sound/soc/amd/acp-pcm-dma.c | 3 +- sound/soc/atmel/Kconfig | 30 +- sound/soc/atmel/atmel-classd.c | 7 +- sound/soc/atmel/atmel-pdmic.c | 7 +- sound/soc/atmel/atmel_ssc_dai.c | 293 +-- sound/soc/atmel/mchp-i2s-mcc.c | 111 +- sound/soc/au1x/psc-ac97.c | 5 +- sound/soc/au1x/psc-i2s.c | 5 +- sound/soc/bcm/bcm2835-i2s.c | 4 +- sound/soc/bcm/cygnus-pcm.c | 6 +- sound/soc/bcm/cygnus-ssp.c | 7 +- sound/soc/cirrus/ep93xx-ac97.c | 4 +- sound/soc/cirrus/ep93xx-i2s.c | 4 +- sound/soc/codecs/88pm860x-codec.c | 15 - sound/soc/codecs/Kconfig | 21 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ad193x.c | 19 +- sound/soc/codecs/cs4271.c | 6 +- sound/soc/codecs/cs42l56.c | 8 - sound/soc/codecs/cs42l73.c | 6 - sound/soc/codecs/cs42xx8.c | 2 + sound/soc/codecs/cs4349.c | 1 + sound/soc/codecs/cs47l15.c | 1490 ++++++++++++++ sound/soc/codecs/cs47l35.c | 2 +- sound/soc/codecs/cs47l90.c | 9 +- sound/soc/codecs/cs47l92.c | 2039 ++++++++++++++++++++ sound/soc/codecs/es8316.c | 73 +- sound/soc/codecs/es8328.c | 3 +- sound/soc/codecs/hdac_hda.c | 4 + sound/soc/codecs/hdac_hdmi.c | 57 +- sound/soc/codecs/hdmi-codec.c | 46 + sound/soc/codecs/inno_rk3036.c | 4 +- sound/soc/codecs/jz4725b.c | 4 +- sound/soc/codecs/jz4740.c | 4 +- sound/soc/codecs/madera.c | 531 ++++- sound/soc/codecs/madera.h | 10 + sound/soc/codecs/max98371.c | 4 - sound/soc/codecs/max98373.c | 34 +- sound/soc/codecs/max98373.h | 1 + sound/soc/codecs/max9850.c | 13 - sound/soc/codecs/max98926.c | 9 - sound/soc/codecs/ml26124.c | 1 - sound/soc/codecs/msm8916-wcd-analog.c | 12 +- sound/soc/codecs/msm8916-wcd-digital.c | 4 +- sound/soc/codecs/mt6351.c | 5 - sound/soc/codecs/mt6358.c | 10 +- sound/soc/codecs/pcm3168a.c | 133 +- sound/soc/codecs/rk3328_codec.c | 4 +- sound/soc/codecs/rt1011.c | 56 +- sound/soc/codecs/rt1011.h | 3 +- sound/soc/codecs/rt1305.c | 3 +- sound/soc/codecs/rt1308.c | 51 +- sound/soc/codecs/rt1308.h | 6 +- sound/soc/codecs/rt5665.c | 8 +- sound/soc/codecs/rt5677.c | 20 +- sound/soc/codecs/sgtl5000.c | 248 ++- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/sirf-audio-codec.c | 4 +- sound/soc/codecs/tlv320aic23.c | 2 - sound/soc/codecs/tlv320aic31xx.c | 8 +- sound/soc/codecs/tscs454.c | 1 - sound/soc/codecs/twl6040.c | 4 +- sound/soc/codecs/uda1334.c | 295 +++ sound/soc/codecs/wcd-clsh-v2.c | 2 +- sound/soc/codecs/wcd9335.c | 23 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8904.c | 56 +- sound/soc/codecs/wm8955.c | 4 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm_adsp.c | 6 +- sound/soc/codecs/wm_adsp.h | 4 +- sound/soc/fsl/fsl_asrc.c | 4 +- sound/soc/fsl/fsl_audmix.c | 4 +- sound/soc/fsl/fsl_esai.c | 267 ++- sound/soc/fsl/fsl_sai.c | 358 ++-- sound/soc/fsl/fsl_sai.h | 85 +- sound/soc/fsl/fsl_spdif.c | 4 +- sound/soc/fsl/fsl_ssi.c | 22 +- sound/soc/fsl/imx-audmix.c | 4 +- sound/soc/fsl/imx-audmux.c | 58 +- sound/soc/fsl/imx-ssi.c | 4 +- sound/soc/generic/audio-graph-card.c | 19 + sound/soc/generic/simple-card-utils.c | 7 + sound/soc/generic/simple-card.c | 22 +- sound/soc/intel/Kconfig | 1 + sound/soc/intel/baytrail/sst-baytrail-ipc.c | 65 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 1 + sound/soc/intel/boards/Kconfig | 28 +- sound/soc/intel/boards/bdw-rt5677.c | 6 +- sound/soc/intel/boards/broadwell.c | 6 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 31 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 101 +- sound/soc/intel/boards/haswell.c | 6 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 5 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 + sound/soc/intel/boards/sof_rt5682.c | 7 +- sound/soc/intel/common/Makefile | 1 + sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 12 + sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 18 + sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 24 + sound/soc/intel/common/soc-intel-quirks.h | 5 + sound/soc/intel/common/sst-acpi.c | 3 +- sound/soc/intel/common/sst-ipc.c | 71 +- sound/soc/intel/common/sst-ipc.h | 28 +- sound/soc/intel/haswell/sst-haswell-ipc.c | 164 +- sound/soc/intel/skylake/Makefile | 12 +- sound/soc/intel/skylake/bxt-sst.c | 50 +- sound/soc/intel/skylake/cnl-sst-dsp.h | 7 +- sound/soc/intel/skylake/cnl-sst.c | 43 +- sound/soc/intel/skylake/skl-debug.c | 44 +- sound/soc/intel/skylake/skl-messages.c | 264 ++- sound/soc/intel/skylake/skl-nhlt.c | 109 +- sound/soc/intel/skylake/skl-pcm.c | 74 +- sound/soc/intel/skylake/skl-ssp-clk.c | 5 +- sound/soc/intel/skylake/skl-sst-dsp.c | 10 +- sound/soc/intel/skylake/skl-sst-dsp.h | 29 +- sound/soc/intel/skylake/skl-sst-ipc.c | 160 +- sound/soc/intel/skylake/skl-sst-ipc.h | 55 +- sound/soc/intel/skylake/skl-sst-utils.c | 37 +- sound/soc/intel/skylake/skl-sst.c | 54 +- sound/soc/intel/skylake/skl-topology.c | 494 ++--- sound/soc/intel/skylake/skl-topology.h | 43 +- sound/soc/intel/skylake/skl.c | 73 +- sound/soc/intel/skylake/skl.h | 105 +- sound/soc/kirkwood/kirkwood-i2s.c | 8 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 3 +- sound/soc/mediatek/common/mtk-btcvsd.c | 4 +- sound/soc/mediatek/mt2701/mt2701-afe-common.h | 21 - sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 38 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 5 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 8 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 15 +- sound/soc/mediatek/mt8183/mt8183-dai-tdm.c | 177 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 150 +- sound/soc/mediatek/mt8183/mt8183-reg.h | 8 +- sound/soc/meson/Kconfig | 1 + sound/soc/meson/axg-card.c | 11 +- sound/soc/meson/axg-fifo.c | 6 +- sound/soc/meson/axg-fifo.h | 1 + sound/soc/meson/axg-frddr.c | 105 +- sound/soc/meson/axg-pdm.c | 4 +- sound/soc/meson/axg-spdifin.c | 4 +- sound/soc/meson/axg-spdifout.c | 4 +- sound/soc/meson/axg-tdm-formatter.c | 6 +- sound/soc/meson/axg-tdmin.c | 47 +- sound/soc/meson/axg-tdmout.c | 103 +- sound/soc/meson/axg-toddr.c | 83 +- sound/soc/meson/g12a-tohdmitx.c | 38 +- sound/soc/mxs/mxs-saif.c | 13 +- sound/soc/nuc900/Kconfig | 29 - sound/soc/nuc900/Makefile | 12 - sound/soc/nuc900/nuc900-ac97.c | 391 ---- sound/soc/nuc900/nuc900-audio.c | 73 - sound/soc/nuc900/nuc900-audio.h | 108 -- sound/soc/nuc900/nuc900-pcm.c | 321 --- sound/soc/pxa/mmp-sspa.c | 4 +- sound/soc/qcom/common.c | 22 +- sound/soc/qcom/lpass-platform.c | 5 +- sound/soc/rockchip/rockchip_i2s.c | 3 + sound/soc/rockchip/rockchip_max98090.c | 18 +- sound/soc/samsung/neo1973_wm8753.c | 3 +- sound/soc/samsung/speyside.c | 3 +- sound/soc/samsung/tm2_wm5110.c | 10 +- sound/soc/sh/rcar/adg.c | 21 +- sound/soc/sh/rcar/core.c | 16 +- sound/soc/sh/rcar/rsnd.h | 12 +- sound/soc/sirf/sirf-usp.c | 4 +- sound/soc/soc-component.c | 561 ++++++ sound/soc/soc-compress.c | 57 +- sound/soc/soc-core.c | 1269 +++++------- sound/soc/soc-dai.c | 407 ++++ sound/soc/soc-dapm.c | 361 ++-- sound/soc/soc-generic-dmaengine-pcm.c | 6 + sound/soc/soc-jack.c | 18 - sound/soc/soc-pcm.c | 546 ++---- sound/soc/soc-topology.c | 8 +- sound/soc/soc-utils.c | 199 -- sound/soc/sof/Kconfig | 11 + sound/soc/sof/Makefile | 8 +- sound/soc/sof/core.c | 4 +- sound/soc/sof/debug.c | 50 +- sound/soc/sof/imx/Kconfig | 23 + sound/soc/sof/imx/Makefile | 4 + sound/soc/sof/imx/imx8.c | 394 ++++ sound/soc/sof/intel/Kconfig | 33 + sound/soc/sof/intel/apl.c | 4 +- sound/soc/sof/intel/bdw.c | 166 +- sound/soc/sof/intel/byt.c | 174 +- sound/soc/sof/intel/cnl.c | 36 +- sound/soc/sof/intel/hda-bus.c | 86 +- sound/soc/sof/intel/hda-codec.c | 50 +- sound/soc/sof/intel/hda-ctrl.c | 21 +- sound/soc/sof/intel/hda-dai.c | 38 +- sound/soc/sof/intel/hda-dsp.c | 100 +- sound/soc/sof/intel/hda-ipc.c | 150 +- sound/soc/sof/intel/hda.c | 132 +- sound/soc/sof/intel/hda.h | 18 +- sound/soc/sof/ipc.c | 8 +- sound/soc/sof/loader.c | 182 +- sound/soc/sof/ops.h | 46 +- sound/soc/sof/pcm.c | 66 +- sound/soc/sof/pm.c | 6 +- sound/soc/sof/sof-of-dev.c | 143 ++ sound/soc/sof/sof-pci-dev.c | 47 + sound/soc/sof/sof-priv.h | 19 +- sound/soc/sof/topology.c | 275 ++- sound/soc/sof/trace.c | 9 +- sound/soc/spear/spdif_in.c | 5 +- sound/soc/sprd/sprd-mcdt.c | 4 +- sound/soc/sti/sti_uniperif.c | 4 +- sound/soc/stm/stm32_i2s.c | 5 +- sound/soc/stm/stm32_sai.c | 8 +- sound/soc/stm/stm32_spdifrx.c | 4 +- sound/soc/sunxi/sun4i-codec.c | 14 +- sound/soc/sunxi/sun4i-i2s.c | 671 ++++--- sound/soc/sunxi/sun50i-codec-analog.c | 4 +- sound/soc/sunxi/sun8i-codec-analog.c | 4 +- sound/soc/sunxi/sun8i-codec.c | 4 +- sound/soc/tegra/tegra20_das.c | 4 +- sound/soc/tegra/tegra30_ahub.c | 5 +- sound/soc/tegra/tegra30_i2s.c | 4 +- sound/soc/ti/Kconfig | 4 +- sound/soc/ti/ams-delta.c | 31 +- sound/soc/ti/davinci-evm.c | 2 +- sound/soc/ti/davinci-i2s.c | 90 +- sound/soc/ti/davinci-mcasp.c | 185 +- sound/soc/ti/edma-pcm.c | 17 +- sound/soc/ti/n810.c | 1 + sound/soc/ti/rx51.c | 15 +- sound/soc/uniphier/aio-cpu.c | 31 +- sound/soc/uniphier/aio-dma.c | 8 +- sound/soc/uniphier/aio.h | 1 + sound/soc/uniphier/evea.c | 4 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 - sound/soc/xilinx/xlnx_i2s.c | 4 +- sound/soc/xilinx/xlnx_spdif.c | 3 +- sound/soc/xtensa/xtfpga-i2s.c | 5 +- sound/soc/zte/zx-tdm.c | 1 - 287 files changed, 12569 insertions(+), 7014 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/fsl,dsp.yaml create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml create mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.txt delete mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt delete mode 100644 Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/uda1334.txt rename sound/soc/intel/skylake/skl-nhlt.h => include/sound/intel-nhlt.h (65%) create mode 100644 include/sound/soc-component.h create mode 100644 sound/hda/intel-nhlt.c create mode 100644 sound/soc/codecs/cs47l15.c create mode 100644 sound/soc/codecs/cs47l92.c create mode 100644 sound/soc/codecs/uda1334.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-ehl-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-tgl-match.c delete mode 100644 sound/soc/nuc900/Kconfig delete mode 100644 sound/soc/nuc900/Makefile delete mode 100644 sound/soc/nuc900/nuc900-ac97.c delete mode 100644 sound/soc/nuc900/nuc900-audio.c delete mode 100644 sound/soc/nuc900/nuc900-audio.h delete mode 100644 sound/soc/nuc900/nuc900-pcm.c create mode 100644 sound/soc/soc-component.c create mode 100644 sound/soc/soc-dai.c create mode 100644 sound/soc/sof/imx/Kconfig create mode 100644 sound/soc/sof/imx/Makefile create mode 100644 sound/soc/sof/imx/imx8.c create mode 100644 sound/soc/sof/sof-of-dev.c

2 1

[alsa-devel] [PATCH] ASoC: SOF: Intel: work around snd_hdac_aligned_read link failure
by Arnd Bergmann 10 Sep '19

10 Sep '19

When CONFIG_SND_HDA_ALIGNED_MMIO is selected by another driver (i.e. Tegra) that selects CONFIG_SND_HDA_CORE as a loadable module, but SND_SOC_SOF_HDA_COMMON is built-in, we get a link failure from some functions that access the hda register: sound/soc/sof/intel/hda.o: In function `hda_ipc_irq_dump': hda.c:(.text+0x784): undefined reference to `snd_hdac_aligned_read' sound/soc/sof/intel/hda-stream.o: In function `hda_dsp_stream_threaded_handler': hda-stream.c:(.text+0x12e4): undefined reference to `snd_hdac_aligned_read' hda-stream.c:(.text+0x12f8): undefined reference to `snd_hdac_aligned_write' Add an explicit 'select' statement as a workaround. This is not a great solution, but it's the easiest way I could come up with. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sof/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..9180184026e1 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -248,6 +248,7 @@ config SND_SOC_SOF_HDA_COMMON tristate select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_HDA_LINK_BASELINE + select SND_HDA_CORE if SND_HDA_ALIGNED_MMIO help This option is not user-selectable but automagically handled by 'select' statements at a higher level -- 2.20.0

3 5

[alsa-devel] omap-mcbsp: TX Buffer Overflow
by Tomas Novotny 10 Sep '19

10 Sep '19

Hi, we have AM3703 based board similar to BeagleBoard. I'm hitting this error after upgrade to latest LTS 4.19.71 (upgraded from 4.1): omap-mcbsp 49022000.mcbsp: TX Buffer Overflow! This appears during or after playing of short (~2s) ding-dong wav. That error exists for longer time, because handling of tx buffer overflow irq was introduced in 2016: 4e85e7776eba ("ASoC: omap-mcbsp: Enable TX/RX under and overflow interrupts"). I've cherry-picked it to 4.1 and I see the error there also. The sound seems clear and ok to me, but we are using low quality speaker. There are two workarounds to get rid of the message: 1) Change 'dma_op_mode' sysfs attribute from 'element' to 'threshold'. I found that just by coincidence when checking sysfs attributes. 2) Compile kernel with CONFIG_VIDEO_OMAP3=y. Found on Logic PD forum [1]. Does anybody have any idea what's going wrong? Or why these (somehow) unrelated workarounds help? Thanks, Tomas [1] https://support.logicpd.com/TDGForum/tabid/124/aft/2277/Default.aspx

3 3

[alsa-devel] [PATCH] ASoC: fsl_sai: Fix noise when using EDMA
by Daniel Baluta 09 Sep '19

09 Sep '19

From: Mihai Serban <mihai.serban(a)nxp.com> EDMA requires the period size to be multiple of maxburst. Otherwise the remaining bytes are not transferred and thus noise is produced. We can handle this issue by adding a constraint on SNDRV_PCM_HW_PARAM_PERIOD_SIZE to be multiple of tx/rx maxburst value. Cc: NXP Linux Team <linux-imx(a)nxp.com> Signed-off-by: Mihai Serban <mihai.serban(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 15 +++++++++++++++ sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 16 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 728307acab90..fe126029f4e3 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -612,6 +612,16 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, FSL_SAI_CR3_TRCE_MASK, FSL_SAI_CR3_TRCE); + /* + * some DMA controllers need period size to be a multiple of + * tx/rx maxburst + */ + if (sai->soc_data->use_constraint_period_size) + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + tx ? sai->dma_params_tx.maxburst : + sai->dma_params_rx.maxburst); + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints); @@ -1011,30 +1021,35 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .use_imx_pcm = false, .fifo_depth = 32, .reg_offset = 0, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .use_imx_pcm = true, .fifo_depth = 32, .reg_offset = 0, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .use_imx_pcm = true, .fifo_depth = 16, .reg_offset = 8, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .use_imx_pcm = true, .fifo_depth = 128, .reg_offset = 8, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_imx_pcm = true, .fifo_depth = 64, .reg_offset = 0, + .use_constraint_period_size = true, }; static const struct of_device_id fsl_sai_ids[] = { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index b89b0ca26053..3a3f6f8e5595 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -157,6 +157,7 @@ struct fsl_sai_soc_data { bool use_imx_pcm; + bool use_constraint_period_size; unsigned int fifo_depth; unsigned int reg_offset; }; -- 2.17.1

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[alsa-devel] about pop chime on pcm api
by Yang, Xiaotao (X.) 09 Sep '19

09 Sep '19

Hi: Why should I call api<snd_pcm_close()> between the interval if I want to play a wav file two times repeatly, Otherwise pop chime occurs? Eg: (fake code as follows): Following codes works good. Pcm.open();//open pcm device Pcm.play(demo.wav);//set parameter and call snd_pcm_writei() to play Pcm.close();//close pcm device Pcm.open(); Pcm.play(demo.wav); Pcm.close(); But following codes works with pop chime Pcm.open(); Pcm.play(demo.wav); Pcm.play(demo.wav); Pcm.close();

2 1

[alsa-devel] Applied "ASoC: hdmi-codec: Add an op to set callback function for plug event" to the asoc tree
by Mark Brown 09 Sep '19

09 Sep '19

The patch ASoC: hdmi-codec: Add an op to set callback function for plug event has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 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 6fa5963c37a2e3335eba0b7455e35a01318ebc15 Mon Sep 17 00:00:00 2001 From: Cheng-Yi Chiang <cychiang(a)chromium.org> Date: Wed, 17 Jul 2019 16:33:23 +0800 Subject: [PATCH] ASoC: hdmi-codec: Add an op to set callback function for plug event Add an op in hdmi_codec_ops so codec driver can register callback function to handle plug event. Driver in DRM can use this callback function to report connector status. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Link: https://lore.kernel.org/r/20190717083327.47646-2-cychiang@chromium.org Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/hdmi-codec.h | 17 +++++++++++++ sound/soc/codecs/hdmi-codec.c | 46 +++++++++++++++++++++++++++++++++++ 2 files changed, 63 insertions(+) diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h index 7fea496f1f34..83b17682e01c 100644 --- a/include/sound/hdmi-codec.h +++ b/include/sound/hdmi-codec.h @@ -47,6 +47,9 @@ struct hdmi_codec_params { int channels; }; +typedef void (*hdmi_codec_plugged_cb)(struct device *dev, + bool plugged); + struct hdmi_codec_pdata; struct hdmi_codec_ops { /* @@ -88,6 +91,14 @@ struct hdmi_codec_ops { */ int (*get_dai_id)(struct snd_soc_component *comment, struct device_node *endpoint); + + /* + * Hook callback function to handle connector plug event. + * Optional + */ + int (*hook_plugged_cb)(struct device *dev, void *data, + hdmi_codec_plugged_cb fn, + struct device *codec_dev); }; /* HDMI codec initalization data */ @@ -99,6 +110,12 @@ struct hdmi_codec_pdata { void *data; }; +struct snd_soc_component; +struct snd_soc_jack; + +int hdmi_codec_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack); + #define HDMI_CODEC_DRV_NAME "hdmi-audio-codec" #endif /* __HDMI_CODEC_H__ */ diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 0bf1c8cad108..b5fd8f08726e 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -7,6 +7,7 @@ #include <linux/module.h> #include <linux/string.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -274,6 +275,8 @@ struct hdmi_codec_priv { struct snd_pcm_chmap *chmap_info; unsigned int chmap_idx; struct mutex lock; + struct snd_soc_jack *jack; + unsigned int jack_status; }; static const struct snd_soc_dapm_widget hdmi_widgets[] = { @@ -663,6 +666,49 @@ static int hdmi_dai_probe(struct snd_soc_dai *dai) return 0; } +static void hdmi_codec_jack_report(struct hdmi_codec_priv *hcp, + unsigned int jack_status) +{ + if (hcp->jack && jack_status != hcp->jack_status) { + snd_soc_jack_report(hcp->jack, jack_status, SND_JACK_LINEOUT); + hcp->jack_status = jack_status; + } +} + +static void plugged_cb(struct device *dev, bool plugged) +{ + struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); + + if (plugged) + hdmi_codec_jack_report(hcp, SND_JACK_LINEOUT); + else + hdmi_codec_jack_report(hcp, 0); +} + +/** + * hdmi_codec_set_jack_detect - register HDMI plugged callback + * @component: the hdmi-codec instance + * @jack: ASoC jack to report (dis)connection events on + */ +int hdmi_codec_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + int ret = -EOPNOTSUPP; + + if (hcp->hcd.ops->hook_plugged_cb) { + hcp->jack = jack; + ret = hcp->hcd.ops->hook_plugged_cb(component->dev->parent, + hcp->hcd.data, + plugged_cb, + component->dev); + if (ret) + hcp->jack = NULL; + } + return ret; +} +EXPORT_SYMBOL_GPL(hdmi_codec_set_jack_detect); + static int hdmi_dai_spdif_probe(struct snd_soc_dai *dai) { struct hdmi_codec_daifmt *cf = dai->playback_dma_data; -- 2.20.1

1 0

[alsa-devel] Alsa conformance test - a new test for ALSA drivers
by Yu-Hsuan Hsu 09 Sep '19

09 Sep '19

Hi all, I have created an new tool "Alsa conformance test"( https://chromium.googlesource.com/chromiumos/platform/audiotest/+/master/al…), which is used for verifying ALSA drivers. Here are some main functions it can measure: 1. Correctness of hw_params 2. Stability of rate 3. Stability of step 4. Runtime of each ALSA API 5. Underrun and Overrun I recommend to use the script "alsa_conformance_test.py" inside. Just give it a device and then it will test the below functions automatically. 1. To make sure whether all the parameters provided by an ALSA driver can be used. 2. To make sure whether the measured rate is the same as the expected rate. We have used this tool to find out many issues in ChromeOS. I hope it can help development in Linux too. If you have any problems or want more features, please create a bug in crbug.com and assign to me ( yuhsuan(a)chromium.org). Thanks! Best, Yu-Hsuan

2 3

[alsa-devel] [PATCH] SoC: simple-card-utils: set 0Hz to sysclk when shutdown
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch set 0Hz to sysclk when shutdown the card. Some codecs set rate constraints that derives from sysclk. This mechanism works correctly if machine drivers give fixed frequency. But simple-audio and audio-graph card set variable clock rate if 'mclk-fs' property exists. In this case, rate constraints will go bad scenario. For example a codec accepts three limited rates (mclk / 256, mclk / 384, mclk / 512). Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - Codec set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 24kHz (1/512) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk to 8.192MHz (= fs * mclk-fs = 32k * 256) - Codec set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 16kHz (1/512) - Play 48kHz again, but it's NOT acceptable because constraints do not allow 48kHz So codecs treat 0Hz sysclk as signal of applying no constraints to avoid this problem. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/generic/simple-card-utils.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 556b1a789629..9b794775df53 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -213,10 +213,17 @@ EXPORT_SYMBOL_GPL(asoc_simple_startup); void asoc_simple_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + if (dai_props->mclk_fs) { + snd_soc_dai_set_sysclk(codec_dai, 0, 0, SND_SOC_CLOCK_IN); + snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_OUT); + } + asoc_simple_clk_disable(dai_props->cpu_dai); asoc_simple_clk_disable(dai_props->codec_dai); -- 2.23.0.rc1

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[alsa-devel] [PATCH] ASoC: rockchip: ignore 0Hz sysclk
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch ignores sysclk setting if it is 0Hz. Some codecs treat 0Hz sysclk as signal of applying no constraints. This driver does not have such feature but current implementation outputs 'Failed to set mclk' error message if machine driver sets 0Hz sysclk to this driver. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/rockchip/rockchip_i2s.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 88ebaf6e1880..af2d5a6124c8 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -419,6 +419,9 @@ static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, struct rk_i2s_dev *i2s = to_info(cpu_dai); int ret; + if (freq == 0) + return 0; + ret = clk_set_rate(i2s->mclk, freq); if (ret) dev_err(i2s->dev, "Fail to set mclk %d\n", ret); -- 2.23.0.rc1

2 1

[alsa-devel] [PATCH 1/2] ASoC: es8316: fix redundant codes of clock
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch removes redundant null checks for optional MCLK clock. And fix DT binding document for changing clock property to optional from required. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/everest,es8316.txt | 3 ++ sound/soc/codecs/es8316.c | 31 ++++++++----------- 2 files changed, 16 insertions(+), 18 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.txt b/Documentation/devicetree/bindings/sound/everest,es8316.txt index aefcff9c48a2..1bf03c5f2af4 100644 --- a/Documentation/devicetree/bindings/sound/everest,es8316.txt +++ b/Documentation/devicetree/bindings/sound/everest,es8316.txt @@ -6,6 +6,9 @@ Required properties: - compatible : should be "everest,es8316" - reg : the I2C address of the device for I2C + +Optional properties: + - clocks : a list of phandle, should contain entries for clock-names - clock-names : should include as follows: "mclk" : master clock (MCLK) of the device diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index d07d50f51b28..1424ecd6952c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -373,11 +373,9 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (freq == 0) return 0; - if (es8316->mclk) { - ret = clk_set_rate(es8316->mclk, freq); - if (ret) - return ret; - } + ret = clk_set_rate(es8316->mclk, freq); + if (ret) + return ret; /* Limit supported sample rates to ones that can be autodetected * by the codec running in slave mode. @@ -712,20 +710,18 @@ static int es8316_probe(struct snd_soc_component *component) es8316->component = component; - es8316->mclk = devm_clk_get(component->dev, "mclk"); - if (PTR_ERR(es8316->mclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; + es8316->mclk = devm_clk_get_optional(component->dev, "mclk"); if (IS_ERR(es8316->mclk)) { - dev_err(component->dev, "clock is invalid, ignored\n"); - es8316->mclk = NULL; + dev_err(component->dev, "unable to get mclk\n"); + return PTR_ERR(es8316->mclk); } + if (!es8316->mclk) + dev_warn(component->dev, "assuming static mclk\n"); - if (es8316->mclk) { - ret = clk_prepare_enable(es8316->mclk); - if (ret) { - dev_err(component->dev, "unable to enable clock\n"); - return ret; - } + ret = clk_prepare_enable(es8316->mclk); + if (ret) { + dev_err(component->dev, "unable to enable mclk\n"); + return ret; } /* Reset codec and enable current state machine */ @@ -754,8 +750,7 @@ static void es8316_remove(struct snd_soc_component *component) { struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - if (es8316->mclk) - clk_disable_unprepare(es8316->mclk); + clk_disable_unprepare(es8316->mclk); } static const struct snd_soc_component_driver soc_component_dev_es8316 = { -- 2.23.0.rc1

2 3

[alsa-devel] [RFC PATCH] ASoC: ams-delta: Take control over audio mute GPIO pins
by Janusz Krzysztofik 09 Sep '19

09 Sep '19

Since commit 1137ceee76ba ("ARM: OMAP1: ams-delta: Don't request unused GPIOs"), on-board audio has appeared muted. It has been discovered that believed to be unused GPIO pins "hookflash1" and "hookflash2" need to be set low for audible sound in handsfree and handset mode respectively. According to Amstrad E3 wiki, the purpose of both pins hasn't been clearly identified. Original Amstrad software used to produce a high pulse on them when the phone was taken off hook or recall was pressed. With the current findings, we can assume the pins provide a kind of audio mute function, separately for handset and handsfree operation modes. Commit 2afdb4c41d78 ("ARM: OMAP1: ams-delta: Fix audio permanently muted") attempted to fix the issue temporarily by hogging the GPIO pin "hookflash1" renamed to "audio_mute", however the fix occurred incomplete as it restored audible sound only for handsfree mode. Stop hogging that pin, rename the pins to "handsfree_mute" and "handset_mute" respectively and implement appropriate DAPM event callbacks for "Speaker" and "Earpiece" DAPM widgets. Fixes: 1137ceee76ba ("ARM: OMAP1: ams-delta: Don't request unused GPIOs") Signed-off-by: Janusz Krzysztofik <jmkrzyszt(a)gmail.com> --- arch/arm/mach-omap1/board-ams-delta.c | 10 +++++---- sound/soc/ti/ams-delta.c | 31 +++++++++++++++++++++++++-- 2 files changed, 35 insertions(+), 6 deletions(-) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index e47a6fbcfd6e..a2aa7a12b374 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -246,8 +246,8 @@ static struct platform_device latch2_gpio_device = { #define LATCH2_PIN_SCARD_CMDVCC 11 #define LATCH2_PIN_MODEM_NRESET 12 #define LATCH2_PIN_MODEM_CODEC 13 -#define LATCH2_PIN_AUDIO_MUTE 14 -#define LATCH2_PIN_HOOKFLASH 15 +#define LATCH2_PIN_HANDSFREE_MUTE 14 +#define LATCH2_PIN_HANDSET_MUTE 15 static struct regulator_consumer_supply modem_nreset_consumers[] = { REGULATOR_SUPPLY("RESET#", "serial8250.1"), @@ -476,6 +476,10 @@ static struct gpiod_lookup_table ams_delta_audio_gpio_table = { "hook_switch", 0), GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_MODEM_CODEC, "modem_codec", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_HANDSFREE_MUTE, + "handsfree_mute", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_HANDSET_MUTE, + "handset_mute", 0), { }, }, }; @@ -590,8 +594,6 @@ static int gpiochip_match_by_label(struct gpio_chip *chip, void *data) static struct gpiod_hog ams_delta_gpio_hogs[] = { GPIO_HOG(LATCH2_LABEL, LATCH2_PIN_KEYBRD_DATAOUT, "keybrd_dataout", GPIO_ACTIVE_HIGH, GPIOD_OUT_LOW), - GPIO_HOG(LATCH2_LABEL, LATCH2_PIN_AUDIO_MUTE, "audio_mute", - GPIO_ACTIVE_HIGH, GPIOD_OUT_LOW), {}, }; diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index dee8fc70a64f..8e2fb81ad05c 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -23,14 +23,31 @@ #include "omap-mcbsp.h" #include "../codecs/cx20442.h" +static struct gpio_desc *handset_mute; +static struct gpio_desc *handsfree_mute; + +static int ams_delta_event_handset(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handset_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int ams_delta_event_handsfree(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handsfree_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + /* Board specific DAPM widgets */ static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { /* Handset */ SND_SOC_DAPM_MIC("Mouthpiece", NULL), - SND_SOC_DAPM_HP("Earpiece", NULL), + SND_SOC_DAPM_HP("Earpiece", ams_delta_event_handset), /* Handsfree/Speakerphone */ SND_SOC_DAPM_MIC("Microphone", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Speaker", ams_delta_event_handsfree), }; /* How they are connected to codec pins */ @@ -542,6 +559,16 @@ static int ams_delta_probe(struct platform_device *pdev) card->dev = &pdev->dev; + handset_mute = devm_gpiod_get(card->dev, "handset_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handset_mute)) + return PTR_ERR(handset_mute); + + handsfree_mute = devm_gpiod_get(card->dev, "handsfree_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handsfree_mute)) + return PTR_ERR(handsfree_mute); + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); -- 2.21.0

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[alsa-devel] [PATCH] ASoC: rt1011: make array pd static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 100 bytes. Before: text data bss dec hex filename 51463 13016 128 64607 fc5f sound/soc/codecs/rt1011.o After: text data bss dec hex filename 51299 13080 128 64507 fbfb sound/soc/codecs/rt1011.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1011.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index f9235764ad32..be1e276e3631 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1519,7 +1519,8 @@ static const struct snd_soc_dapm_route rt1011_dapm_routes[] = { static int rt1011_get_clk_info(int sclk, int rate) { - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; if (sclk <= 0 || rate <= 0) return -EINVAL; -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: codecs: ad193x: make two arrays static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the arrays on the stack but instead make them static const. Makes the object code smaller by 37 bytes. Before: text data bss dec hex filename 16253 7200 0 23453 5b9d sound/soc/codecs/ad193x.o After: text data bss dec hex filename 16056 7360 0 23416 5b78 sound/soc/codecs/ad193x.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/ad193x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index fb04c9379b71..980e024a5720 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -416,7 +416,7 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = { /* codec register values to set after reset */ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { - const struct reg_sequence reg_init[] = { + static const struct reg_sequence reg_init[] = { { 0, 0x99 }, /* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */ { 1, 0x04 }, /* PLL_CLK_CTRL1: no on-chip Vref */ { 2, 0x40 }, /* DAC_CTRL0: TDM mode */ @@ -432,7 +432,7 @@ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { 12, 0x00 }, /* DAC_L4_VOL: no attenuation */ { 13, 0x00 }, /* DAC_R4_VOL: no attenuation */ }; - const struct reg_sequence reg_adc_init[] = { + static const struct reg_sequence reg_adc_init[] = { { 14, 0x03 }, /* ADC_CTRL0: high-pass filter enable */ { 15, 0x43 }, /* ADC_CTRL1: sata delay=1, adc aux mode */ { 16, 0x00 }, /* ADC_CTRL2: reset */ -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: rt1308: make array pd static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 82 bytes. Before: text data bss dec hex filename 26548 7288 64 33900 846c sound/soc/codecs/rt1308.o After: text data bss dec hex filename 26370 7384 64 33818 841a sound/soc/codecs/rt1308.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1308.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index 6f2ee6809dbb..b75931a69a1c 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -436,7 +436,8 @@ static const struct snd_soc_dapm_route rt1308_dapm_routes[] = { static int rt1308_get_clk_info(int sclk, int rate) { - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; if (sclk <= 0 || rate <= 0) return -EINVAL; -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: rt1305: make array pd static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 93 bytes. Before: text data bss dec hex filename 38961 9784 64 48809 bea9 sound/soc/codecs/rt1305.o After: text data bss dec hex filename 38804 9848 64 48716 be4c sound/soc/codecs/rt1305.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1305.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index 9909369483f0..e27742abfa76 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -608,7 +608,8 @@ static const struct snd_soc_dapm_route rt1305_dapm_routes[] = { static int rt1305_get_clk_info(int sclk, int rate) { - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; if (sclk <= 0 || rate <= 0) return -EINVAL; -- 2.20.1

2 1

[alsa-devel] [PATCH resend 00/11] ASoC: soc-core cleanup
by Kuninori Morimoto 09 Sep '19

09 Sep '19

Hi Mark These are resend of soc-core cleanup patches which are not yet accepted/reviewed. These are based on latest mark/for-v5.4 branch. Kuninori Morimoto (11): ASoC: soc-core: add comment to jack at soc_remove_component() ASoC: soc-core: self contained soc_probe_link_components() ASoC: soc-core: self contained soc_remove_link_components() ASoC: soc-core: self contained soc_remove_link_dais() ASoC: soc-core: move soc_probe_dai() next to soc_remove_dai() ASoC: soc-core: add new soc_link_init() ASoC: soc-core: self contained soc_probe_link_dais() ASoC: soc-core: move soc_probe_link_dais() next to soc_remove_link_dais() ASoC: soc-core: self contained soc_bind_aux_dev() ASoC: soc-core: add soc_unbind_aux_dev() ASoC: soc-core: self contained soc_unbind_aux_dev() sound/soc/soc-core.c | 264 +++++++++++++++++++++++++++------------------------ 1 file changed, 139 insertions(+), 125 deletions(-) -- 2.7.4

2 23

[alsa-devel] [PATCH v2] drm: bridge/dw_hdmi: add audio sample channel status setting
by Cheng-Yi Chiang 09 Sep '19

09 Sep '19

From: Yakir Yang <ykk(a)rock-chips.com> When transmitting IEC60985 linear PCM audio, we configure the Aduio Sample Channel Status information of all the channel status bits in the IEC60958 frame. Refer to 60958-3 page 10 for frequency, original frequency, and wordlength setting. This fix the issue that audio does not come out on some monitors (e.g. LG 22CV241) Note that these registers are only for interfaces: I2S audio interface, General Purpose Audio (GPA), or AHB audio DMA (AHBAUDDMA). For S/PDIF interface this information comes from the stream. Currently this function dw_hdmi_set_channel_status is only called from dw-hdmi-i2s-audio in I2S setup. Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Original patch by Yakir Yang is at https://lore.kernel.org/patchwork/patch/539653/ Change from v1 to v2: 1. Remove the version check because this will only be called by dw-hdmi-i2s-audio, and the registers are available in I2S setup. 2. Set these registers in dw_hdmi_i2s_hw_params. 3. Fix the sample width setting so it can use 16 or 24 bits. .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 1 + drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 70 +++++++++++++++++++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.h | 20 ++++++ include/drm/bridge/dw_hdmi.h | 2 + 4 files changed, 93 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index 34d8e837555f..b801a28b0f17 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -102,6 +102,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, } dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); + dw_hdmi_set_channel_status(hdmi, hparms->sample_width); dw_hdmi_set_channel_count(hdmi, hparms->channels); dw_hdmi_set_channel_allocation(hdmi, hparms->cea.channel_allocation); diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index bd65d0479683..d1daa369c8ae 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -582,6 +582,76 @@ static unsigned int hdmi_compute_n(unsigned int freq, unsigned long pixel_clk) return n; } +/* + * When transmitting IEC60958 linear PCM audio, these registers allow to + * configure the channel status information of all the channel status + * bits in the IEC60958 frame. For the moment this configuration is only + * used when the I2S audio interface, General Purpose Audio (GPA), + * or AHB audio DMA (AHBAUDDMA) interface is active + * (for S/PDIF interface this information comes from the stream). + */ +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, + unsigned int sample_width) +{ + u8 aud_schnl_samplerate; + u8 aud_schnl_8; + u8 word_length_bits; + + switch (hdmi->sample_rate) { + case 32000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_32K; + break; + case 44100: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_44K1; + break; + case 48000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_48K; + break; + case 88200: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_88K2; + break; + case 96000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_96K; + break; + case 176400: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_176K4; + break; + case 192000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_192K; + break; + case 768000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_768K; + break; + default: + dev_warn(hdmi->dev, "Unsupported audio sample rate (%u)\n", + hdmi->sample_rate); + return; + } + + /* set channel status register */ + hdmi_modb(hdmi, aud_schnl_samplerate, HDMI_FC_AUDSCHNLS7_SMPRATE_MASK, + HDMI_FC_AUDSCHNLS7); + + /* + * Set original frequency to be the same as frequency. + * Use one-complement value as stated in IEC60958-3 page 13. + */ + aud_schnl_8 = (~aud_schnl_samplerate) << + HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_OFFSET; + + /* + * Refer to IEC60958-3 page 12. We can accept 16 bits or 24 bits. + * Otherwise, set the register to 0t o indicate using default value. + */ + word_length_bits = (sample_width == 16) ? 0x2 : + ((sample_width == 24) ? 0xb : 0); + + aud_schnl_8 |= word_length_bits << HDMI_FC_AUDSCHNLS8_WORDLEGNTH_OFFSET; + + hdmi_writeb(hdmi, aud_schnl_8, HDMI_FC_AUDSCHNLS8); +} +EXPORT_SYMBOL_GPL(dw_hdmi_set_channel_status); + static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, unsigned long pixel_clk, unsigned int sample_rate) { diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h index 6988f12d89d9..619ebc1c8354 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -158,6 +158,17 @@ #define HDMI_FC_SPDDEVICEINF 0x1062 #define HDMI_FC_AUDSCONF 0x1063 #define HDMI_FC_AUDSSTAT 0x1064 +#define HDMI_FC_AUDSV 0x1065 +#define HDMI_FC_AUDSU 0x1066 +#define HDMI_FC_AUDSCHNLS0 0x1067 +#define HDMI_FC_AUDSCHNLS1 0x1068 +#define HDMI_FC_AUDSCHNLS2 0x1069 +#define HDMI_FC_AUDSCHNLS3 0x106a +#define HDMI_FC_AUDSCHNLS4 0x106b +#define HDMI_FC_AUDSCHNLS5 0x106c +#define HDMI_FC_AUDSCHNLS6 0x106d +#define HDMI_FC_AUDSCHNLS7 0x106e +#define HDMI_FC_AUDSCHNLS8 0x106f #define HDMI_FC_DATACH0FILL 0x1070 #define HDMI_FC_DATACH1FILL 0x1071 #define HDMI_FC_DATACH2FILL 0x1072 @@ -706,6 +717,15 @@ enum { /* HDMI_FC_AUDSCHNLS7 field values */ HDMI_FC_AUDSCHNLS7_ACCURACY_OFFSET = 4, HDMI_FC_AUDSCHNLS7_ACCURACY_MASK = 0x30, + HDMI_FC_AUDSCHNLS7_SMPRATE_MASK = 0x0f, + HDMI_FC_AUDSCHNLS7_SMPRATE_192K = 0xe, + HDMI_FC_AUDSCHNLS7_SMPRATE_176K4 = 0xc, + HDMI_FC_AUDSCHNLS7_SMPRATE_96K = 0xa, + HDMI_FC_AUDSCHNLS7_SMPRATE_768K = 0x9, + HDMI_FC_AUDSCHNLS7_SMPRATE_88K2 = 0x8, + HDMI_FC_AUDSCHNLS7_SMPRATE_32K = 0x3, + HDMI_FC_AUDSCHNLS7_SMPRATE_48K = 0x2, + HDMI_FC_AUDSCHNLS7_SMPRATE_44K1 = 0x0, /* HDMI_FC_AUDSCHNLS8 field values */ HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_MASK = 0xf0, diff --git a/include/drm/bridge/dw_hdmi.h b/include/drm/bridge/dw_hdmi.h index cf528c289857..12144d2f80f4 100644 --- a/include/drm/bridge/dw_hdmi.h +++ b/include/drm/bridge/dw_hdmi.h @@ -156,6 +156,8 @@ void dw_hdmi_setup_rx_sense(struct dw_hdmi *hdmi, bool hpd, bool rx_sense); void dw_hdmi_set_sample_rate(struct dw_hdmi *hdmi, unsigned int rate); void dw_hdmi_set_channel_count(struct dw_hdmi *hdmi, unsigned int cnt); +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, + unsigned int sample_width); void dw_hdmi_set_channel_allocation(struct dw_hdmi *hdmi, unsigned int ca); void dw_hdmi_audio_enable(struct dw_hdmi *hdmi); void dw_hdmi_audio_disable(struct dw_hdmi *hdmi); -- 2.23.0.187.g17f5b7556c-goog

3 5

[alsa-devel] [PATCH v2 0/6] ] ASoC: atmel: extend SSC support
by Michał Mirosław 08 Sep '19

08 Sep '19

This series improves support for various configurations using SSC module as implemented in Atmel SAMA5Dx SoCs. Patches are: 1. enable SSC in Kconfig for audio-graph-card support 2. DRY mode setting code 3. implement left-justified data mode 4-6. enable shared FSYNC source for slave mode Patches against tiwai/sound/for-next tree. You can also pull from https://rere.qmqm.pl/git/linux branch: atmel-ssc Michał Mirosław (6): ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig ASoC: atmel_ssc_dai: rework DAI format configuration ASoC: atmel_ssc_dai: implement left-justified data mode dt-bindings: misc: atmel-ssc: LRCLK from TF/RF pin option misc: atmel-ssc: get LRCLK pin selection from DT ASoC: atmel_ssc_dai: Enable shared FSYNC source in frame-slave mode .../devicetree/bindings/misc/atmel-ssc.txt | 5 + drivers/misc/atmel-ssc.c | 9 + include/linux/atmel-ssc.h | 2 + sound/soc/atmel/Kconfig | 30 +- sound/soc/atmel/atmel_ssc_dai.c | 305 ++++++------------ 5 files changed, 137 insertions(+), 214 deletions(-) -- 2.20.1

5 16

[alsa-devel] [PATCH] alsa: lx6464es - add support for LX6464ESe pci express variant
by Tim Blechmann 08 Sep '19

08 Sep '19

The pci express variant of the digigram lx6464es card has a different device ID, but works without changes to the driver. Thanks to Nikolas Slottke for reporting and testing. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- include/linux/pci_ids.h | 2 ++ sound/pci/lx6464es/lx6464es.c | 8 ++++++++ 2 files changed, 10 insertions(+) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index 7fa3f1498b34..08e15891ed2e 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -1933,6 +1933,8 @@ #define PCI_VENDOR_ID_DIGIGRAM 0x1369 #define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM 0xc001 #define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM 0xc002 +#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_SERIAL_SUBSYSTEM 0xc021 +#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_CAE_SERIAL_SUBSYSTEM 0xc022 #define PCI_VENDOR_ID_KAWASAKI 0x136b #define PCI_DEVICE_ID_MCHIP_KL5A72002 0xff01 diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 9655b08a1c52..3f07e346ad41 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -65,6 +65,14 @@ static const struct pci_device_id snd_lx6464es_ids[] = { PCI_VENDOR_ID_DIGIGRAM, PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM), }, /* LX6464ES-CAE */ + { PCI_DEVICE_SUB(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES, + PCI_VENDOR_ID_DIGIGRAM, + PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_SERIAL_SUBSYSTEM), + }, /* LX6464ESe */ + { PCI_DEVICE_SUB(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES, + PCI_VENDOR_ID_DIGIGRAM, + PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_CAE_SERIAL_SUBSYSTEM), + }, /* LX6464ESe-CAE */ { 0, }, }; -- 2.18.0

2 1

[alsa-devel] Low sound and no mic in Lenovo Yoga / Ideapad S940 (Realtek ALC298)
by José Mesa Orihuela 07 Sep '19

07 Sep '19

I have tested 3 differents linux distros (manjaro, ubuntu, popos) with the same results: * Low sound volume / ouput * No mic (seems detected but doesn't work) This problem looks similar to C930. I have tried all the solutions (pin remapping, changing model...) without success. https://github.com/droserasprout/lenovo-yoga-c930-linux#speaker Alsa-info.sh log: http://alsa-project.org/db/?f=701015751be757989bb93d70a9a06f7e89d64e24 Linux lnx 4.19.66-1-MANJARO#1SMP PREEMPT Fri Aug 9 18:01:53 UTC 2019 x86_64 GNU/Lin More logs: |00:1f.3 Multimedia audio controller [0401]: Intel Corporation Cannon Point-LP High Definition Audio Controller [8086:9dc8] (rev 30) | |card 0: PCH [HDA Intel PCH], device 0: ALC298 Analog [ALC298 Analog] | |snd_hda_codec_hdmi 69632 1 snd_sof_intel_hda_common 73728 1 sof_pci_dev snd_sof_intel_hda 20480 1 snd_sof_intel_hda_common snd_sof 118784 4 snd_sof_intel_hda_common,snd_sof_intel_byt,snd_sof_intel_ipc,sof_pci_dev snd_hda_codec_realtek 126976 1 snd_hda_codec_generic 94208 1 snd_hda_codec_realtek snd_soc_hdac_hda 24576 2 snd_sof_intel_hda_common,snd_soc_skl ledtrig_audio 16384 2 snd_hda_codec_generic,snd_hda_codec_realtek snd_hda_ext_core 36864 4 snd_sof_intel_hda_common,snd_soc_hdac_hda,snd_soc_skl,snd_sof_intel_hda snd_soc_acpi_intel_match 28672 3 snd_sof_intel_hda_common,sof_pci_dev,snd_soc_skl snd_soc_core 290816 4 snd_sof,snd_sof_intel_hda_common,snd_soc_hdac_hda,snd_soc_skl snd_hda_intel 49152 8 snd_hda_codec 159744 5 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec_realtek,snd_soc_hdac_hda snd_hda_core 102400 10 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_ext_core,snd_hda_codec,snd_hda_codec_realtek,snd_sof_intel_hda_common,snd_soc_hdac_hda,snd_soc_skl,snd_sof_intel_hda snd_hwdep 20480 1 snd_hda_codec snd_pcm 139264 11 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_ext_core,snd_hda_codec,snd_sof,snd_sof_intel_hda_common,snd_soc_core,snd_soc_skl,snd_hda_core,snd_pcm_dmaengine snd 110592 25 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hwdep,snd_hda_intel,snd_hda_codec,snd_hda_codec_realtek,snd_timer,snd_compress,snd_soc_core,snd_pcm | Any ideas to solve this problem? Regards

1 0

[alsa-devel] [PATCH] ALSA: firewire-lib: remove WARN_ON() at destruction of AMDTP domain
by Takashi Sakamoto 07 Sep '19

07 Sep '19

The destructor of AMDTP domain has WARN_ON() for the list of associated AMDTP stream. Although this reminds a case that developers forget to program consumer drivers to stop AMDTP domain, it hits when AMDTP domain is not initialized yet. This occurs when initialization of sound card fails as well and it's superfluous. This commit removes the WARN_ON. Although the API to AMDTP domain does nothing, it's left for future usage. Fixes: 3ec3d7a3ff106 ("ALSA: firewire-lib: add AMDTP domain structure to handle several isoc contexts") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 88270257e896..e50e28f77e74 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1153,7 +1153,8 @@ EXPORT_SYMBOL_GPL(amdtp_domain_init); */ void amdtp_domain_destroy(struct amdtp_domain *d) { - WARN_ON(!list_empty(&d->streams)); + // At present nothing to do. + return; } EXPORT_SYMBOL_GPL(amdtp_domain_destroy); -- 2.20.1

2 1

[alsa-devel] [RFC PATCH v2 0/6] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 06 Sep '19

06 Sep '19

Hi all, here's a second RFC round for this series that adapts SOF (and one example machine driver) to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. If the approach looks ok, I'll remove RFC from v3 and update the remaining machine drivers (this v2 still changes only one machine driver and thus the RFC tag). v2 changes: - Codec selection via module parameter. Concerns were raised that a Kconfig option will not be sufficient as the patch changes user-space mixer controls. Distro kernels will want to use same kernel config for all platforms and a kconfig control would apply to all drivers. To address this, a combination of a module parameter, and a default value set via kconfig, was added to the patchset. This allows distros to support different combinations with a single kernel build. To switch codecs for a specific machine, UCM file update can be deployed together with a change to modprobe parameter. This is somewhat complicated, so further feedback is very welcome whether this is enough. - In machine drivers that are shared with SST and SOF, added runtime detection for which codec is used. For SST, the old hdac-hdmi is always used. - One patch dropped from series. This patch is no longer needed (pcm handles now passed to codec driver). Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) - Pre-reviews at: https://github.com/thesofproject/linux/pull/1155 Kai Vehmanen (6): ALSA: hda - add mst_no_extra_pcms flag to hda_codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ALSA: hda/hdmi - implement mst_no_extra_pcms flag ALSA: hda/hdmi - allow control creation without a linked pcm ASoC: SOF: Intel: add support for snd-hda-codec-hdmi include/sound/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 23 +++-- sound/soc/codecs/hdac_hda.c | 95 +++++++++++++++++--- sound/soc/codecs/hdac_hda.h | 12 ++- sound/soc/intel/boards/skl_hda_dsp_common.c | 28 +++++- sound/soc/intel/boards/skl_hda_dsp_common.h | 69 ++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 7 -- sound/soc/sof/intel/Kconfig | 10 +++ sound/soc/sof/intel/hda-codec.c | 14 ++- sound/soc/sof/intel/hda.h | 6 +- 10 files changed, 231 insertions(+), 34 deletions(-) -- 2.17.1

3 10

[alsa-devel] [PATCH] ALSA: firewire-tascam: add note for FE-8
by Takashi Sakamoto 06 Sep '19

06 Sep '19

TASCAM FE-8 is the rest of model in TASCAM FireWire series. This device has no functionality to process audio signal and MIDI messages. Instead, it transfers control messages to host system corresponding to operations for some faders, buttons and knobs on its surface. Unlike the other devices in this series, the control messages are transmitted by asynchronous transactions. Some registers of device are used for registration of destination address for the transaction. The transaction includes quadlet-aligned data up to 32 quadlets. Userspace applications can receive the transaction and parse it for control message via Linux FireWire subsystem, without any support by ALSA firewire-tascam driver. Therefore the driver gives no support for it. This commit removes placeholder for FE-8 and add some comment for its functionalities as notes. $ python2 linux-firewire-utils/src/crpp < ~/git/am-config-rom/tascam/tascam-fe8.img ROM header and bus information block ----------------------------------------------------------------- 400 040f4798 bus_info_length 4, crc_length 15, crc 18328 (should be 14256) 404 31333934 bus_name "1394" 408 20ff7002 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 40c 00022eff company_id 00022e | 410 a094dcb7 device_id ffa094dcb7 | EUI-64 00022effa094dcb7 root directory ----------------------------------------------------------------- 414 0004bccc directory_length 4, crc 48332 418 0300022e vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 8d000006 --> eui-64 leaf at 438 424 d1000001 --> unit directory at 428 unit directory at 428 ----------------------------------------------------------------- 428 0003eda4 directory_length 3, crc 60836 42c 1200022e specifier id 430 13800001 version 434 d4000004 --> dependent info directory at 444 eui-64 leaf at 438 ----------------------------------------------------------------- 438 0002461e leaf_length 2, crc 17950 43c 00022eff company_id 00022e | 440 a094dcb7 device_id ffa094dcb7 | EUI-64 00022effa094dcb7 dependent info directory at 444 ----------------------------------------------------------------- 444 0002ae47 directory_length 2, crc 44615 448 81000002 --> descriptor leaf at 450 44c 82000006 --> bus dependent info leaf at 464 descriptor leaf at 450 ----------------------------------------------------------------- 450 0004a79e leaf_length 4, crc 42910 454 00000000 textual descriptor 458 00000000 minimal ASCII 45c 54415343 "TASC" 460 414d0000 "AM" bus dependent info leaf at 464 ----------------------------------------------------------------- 464 0004a7d8 leaf_length 4, crc 42968 468 00000000 46c 00000000 470 46452d38 474 00000000 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/tascam/tascam.c | 4 +++- sound/firewire/tascam/tascam.h | 20 ++++++++++++++++++++ 2 files changed, 23 insertions(+), 1 deletion(-) diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index 231052db5680..addc464503bc 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -39,6 +39,9 @@ static const struct snd_tscm_spec model_specs[] = { .midi_capture_ports = 2, .midi_playback_ports = 4, }, + // This kernel module doesn't support FE-8 because the most of features + // can be implemented in userspace without any specific support of this + // module. }; static int identify_model(struct snd_tscm *tscm) @@ -214,7 +217,6 @@ static const struct ieee1394_device_id snd_tscm_id_table[] = { .vendor_id = 0x00022e, .specifier_id = 0x00022e, }, - /* FE-08 requires reverse-engineering because it just has faders. */ {} }; MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table); diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 64a2e4d2bbfe..15bd335fa07f 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -129,6 +129,26 @@ struct snd_tscm { #define TSCM_OFFSET_MIDI_RX_QUAD 0x4000 +// Although FE-8 supports the above registers, it has no I/O interfaces for +// audio samples and music messages. Otherwise it supports another notification +// for status and control message as well as LED brightening. The message +// consists of quadlet-aligned data up to 32 quadlets. The first byte of message +// is fixed to 0x40. The second byte is between 0x00 to 0x1f and represent each +// control: +// fader: 0x00-0x07 +// button: 0x0d, 0x0e +// knob: 0x14-0x1b +// sensing: 0x0b +// +// The rest two bytes represent state of the controls; e.g. current value for +// fader and knob, bitmasks for button and sensing. +// Just after turning on, 32 quadlets messages with 0x00-0x1f are immediately +// sent in one transaction. After, several quadlets are sent in one transaction. +// +// TSCM_OFFSET_FE8_CTL_TX_ON 0x0310 +// TSCM_OFFSET_FE8_CTL_TX_ADDR_HI 0x0314 +// TSCM_OFFSET_FE8_CTL_TX_ADDR_LO 0x0318 + enum snd_tscm_clock { SND_TSCM_CLOCK_INTERNAL = 0, SND_TSCM_CLOCK_WORD = 1, -- 2.20.1

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[alsa-devel] [RFC PATCH 1/3] ALSA:hda: Simplify and clear calculating SDxFMT
by Pawel Harlozinski 06 Sep '19

06 Sep '19

Set SDxFMT based only on given format, due to maxbps not always being set. Split cases for formats 20,24,32 bits. For format SNDRV_PCM_FORMAT_FLOAT_LE width is equal 32 so it will end up with same mask. Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/hda/hdac_device.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b26cc93e7e10..add758e11b85 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -759,14 +759,13 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, val |= AC_FMT_BITS_16; break; case 20: + val |= AC_FMT_BITS_20; + break; case 24: + val |= AC_FMT_BITS_24; + break; case 32: - if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) - val |= AC_FMT_BITS_32; - else if (maxbps >= 24) - val |= AC_FMT_BITS_24; - else - val |= AC_FMT_BITS_20; + val |= AC_FMT_BITS_32; break; default: return 0; -- 2.17.1

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[alsa-devel] [PATCH] soundwire: add back ACPI dependency
by Arnd Bergmann 06 Sep '19

06 Sep '19

Soundwire gained a warning for randconfig builds without CONFIG_ACPI during the linux-5.3-rc cycle: drivers/soundwire/slave.c:16:12: error: unused function 'sdw_slave_add' [-Werror,-Wunused-function] Add the CONFIG_ACPI dependency at the top level now. Fixes: 8676b3ca4673 ("soundwire: fix regmap dependencies and align with other serial links") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- drivers/soundwire/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index f518273cfbe3..c73bfbaa2659 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -5,6 +5,7 @@ menuconfig SOUNDWIRE tristate "SoundWire support" + depends on ACPI help SoundWire is a 2-Pin interface with data and clock line ratified by the MIPI Alliance. SoundWire is used for transporting data -- 2.20.0

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[alsa-devel] alsa-devel ML archived on lore.kernel.org
by Takashi Iwai 06 Sep '19

06 Sep '19

Hi, per popular demands, now lore.kernel.org also archives the posts on alsa-devel ML: https://lore.kernel.org/alsa-devel/ This includes the posts in the past that are found on alsa-project.org ML archive, too. A good thing is that this allows us to refer to the thread via the URL like: https://lore.kernel.org/r/$MESSAGEID thanks, Takashi

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[alsa-devel] [PATCH] ALSA: hda/hdmi: remove redundant assignment to variable pcm_idx
by Colin King 05 Sep '19

05 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Variable pcm_idx is being initialized with a value that is never read and is being re-assigned immediately afterwards. The assignment is redundant and hence can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 83b8b9d27711..bca5de78e9ad 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1440,7 +1440,7 @@ static bool update_eld(struct hda_codec *codec, struct hdmi_spec *spec = codec->spec; bool old_eld_valid = pin_eld->eld_valid; bool eld_changed; - int pcm_idx = -1; + int pcm_idx; /* for monitor disconnection, save pcm_idx firstly */ pcm_idx = per_pin->pcm_idx; -- 2.20.1

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[alsa-devel] [PATCH] ASoC: qcom: common: Include link-name in error messages
by Bjorn Andersson 05 Sep '19

05 Sep '19

Reading out the link-name earlier and including it in the various error messages makes it much more convenient to figure out what links have unmet dependencies. Signed-off-by: Bjorn Andersson <bjorn.andersson(a)linaro.org> --- sound/soc/qcom/common.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 2c7348ddbbb3..6c20bdd850f3 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -53,12 +53,18 @@ int qcom_snd_parse_of(struct snd_soc_card *card) link->num_cpus = 1; link->num_platforms = 1; + ret = of_property_read_string(np, "link-name", &link->name); + if (ret) { + dev_err(card->dev, "error getting codec dai_link name\n"); + goto err; + } + cpu = of_get_child_by_name(np, "cpu"); platform = of_get_child_by_name(np, "platform"); codec = of_get_child_by_name(np, "codec"); if (!cpu) { - dev_err(dev, "Can't find cpu DT node\n"); + dev_err(dev, "%s: Can't find cpu DT node\n", link->name); ret = -EINVAL; goto err; } @@ -66,7 +72,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) ret = of_parse_phandle_with_args(cpu, "sound-dai", "#sound-dai-cells", 0, &args); if (ret) { - dev_err(card->dev, "error getting cpu phandle\n"); + dev_err(card->dev, "%s: error getting cpu phandle\n", link->name); goto err; } link->cpus->of_node = args.np; @@ -74,7 +80,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) ret = snd_soc_of_get_dai_name(cpu, &link->cpus->dai_name); if (ret) { - dev_err(card->dev, "error getting cpu dai name\n"); + dev_err(card->dev, "%s: error getting cpu dai name\n", link->name); goto err; } @@ -83,14 +89,14 @@ int qcom_snd_parse_of(struct snd_soc_card *card) "sound-dai", 0); if (!link->platforms->of_node) { - dev_err(card->dev, "platform dai not found\n"); + dev_err(card->dev, "%s: platform dai not found\n", link->name); ret = -EINVAL; goto err; } ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); if (ret < 0) { - dev_err(card->dev, "codec dai not found\n"); + dev_err(card->dev, "%s: codec dai not found\n", link->name); goto err; } link->no_pcm = 1; @@ -110,12 +116,6 @@ int qcom_snd_parse_of(struct snd_soc_card *card) } link->ignore_suspend = 1; - ret = of_property_read_string(np, "link-name", &link->name); - if (ret) { - dev_err(card->dev, "error getting codec dai_link name\n"); - goto err; - } - link->nonatomic = 1; link->dpcm_playback = 1; link->dpcm_capture = 1; -- 2.18.0

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[alsa-devel] [PATCH 0/8] ASoC: meson: add sm1 support
by Jerome Brunet 05 Sep '19

05 Sep '19

This patchset prepare then add the audio support on the amlogic sm1 SoC family in ASoC. This has been tested on sei610 platform. Since this platform does not have spdif, this patchset does not include the change necessary to support spdif input or output on the sm1. Jerome Brunet (8): ASoC: meson: add sm1 compatibles ASoC: meson: add reset binding ASoC: meson: axg-frddr: expose all 8 outputs ASoC: meson: axg-toddr: expose all 8 inputs ASoC: meson: tdmin: expose all 16 inputs ASoC: meson: axg-frddr: add sm1 support ASoC: meson: axg-toddr: add sm1 support ASoC: meson: tdmout: add sm1 support .../bindings/sound/amlogic,axg-fifo.txt | 9 +- .../bindings/sound/amlogic,axg-pdm.txt | 6 +- .../bindings/sound/amlogic,axg-spdifin.txt | 6 +- .../bindings/sound/amlogic,axg-spdifout.txt | 6 +- .../sound/amlogic,axg-tdm-formatters.txt | 4 +- .../bindings/sound/amlogic,g12a-tohdmitx.txt | 5 +- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/axg-fifo.h | 1 + sound/soc/meson/axg-frddr.c | 105 ++++++++++++++++-- sound/soc/meson/axg-tdmin.c | 47 +++++--- sound/soc/meson/axg-tdmout.c | 103 +++++++++++++---- sound/soc/meson/axg-toddr.c | 83 ++++++++++++-- 12 files changed, 321 insertions(+), 56 deletions(-) -- 2.21.0

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