June 2020 - Alsa-devel - mailman.alsa-project.org

ALSA journalctl Error
by GitHub issues - edited 10 Jun '20

10 Jun '20

alsa-project/alsa-lib issue #57 was edited from danieloppenlander: I ran across an error in my journalctl on Ubuntu 20.04. It said to report a bug. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=6b505aa5138f2a974037ab5c3a89e1605807df97 Here is the journalctl message: ``` We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/57 Repository URL: https://github.com/alsa-project/alsa-lib

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ALSA journalctl error.
by GitHub issues - edited 10 Jun '20

10 Jun '20

alsa-project/alsa-lib issue #57 was edited from danieloppenlander: I ran across an error in my journalctl on Ubuntu 20.04. It said to report a bug. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=6b505aa5138f2a974037ab5c3a89e1605807df97 Here is the journalctl message: ``` We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/57 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write.
by GitHub issues - edited 10 Jun '20

10 Jun '20

alsa-project/alsa-lib issue #57 was edited from danieloppenlander: I ran across an error in my journalctl on Ubuntu 20.04. It said to report a bug. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=6b505aa5138f2a974037ab5c3a89e1605807df97 Here is the journalctl message: ``` We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/57 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 14:31:33 elrond pulseaudio[11933]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 14:31:33 elrond pulseaudio[11933]: ALSA woke us up to write new data to the device, but there was actually nothing to write.
by GitHub issues - opened 10 Jun '20

10 Jun '20

alsa-project/alsa-lib issue #57 was opened from danieloppenlander: I ran across an error in my journalctl on Ubuntu 20.04. It said to report a bug. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=6b505aa5138f2a974037ab5c3a89e1605807df97 Issue URL : https://github.com/alsa-project/alsa-lib/issues/57 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/2] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT10-A tablet
by Hans de Goede 09 Jun '20

09 Jun '20

The Toshiba Encore WT10-A tablet almost fully works with the default settings for Bay Trail CR devices. The only issue is that it uses a digital mic. connected the the DMIC1 input instead of an analog mic. Add a quirk for this model using the default settings with the input-map replaced with BYT_RT5640_DMIC1_MAP. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index fbfd53874b47..5c1a5e2aff6f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -754,6 +754,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_JD_NOT_INV | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT10-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.26.2

3 5

[PATCH v2] dt-bindings: tas2562: Convert the tas2562 binding to yaml
by Dan Murphy 09 Jun '20

09 Jun '20

Convert the TAS2562 text file to yaml format. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 34 ---------- .../devicetree/bindings/sound/tas2562.yaml | 65 +++++++++++++++++++ 2 files changed, 65 insertions(+), 34 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2562.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt deleted file mode 100644 index 94796b547184..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ /dev/null @@ -1,34 +0,0 @@ -Texas Instruments TAS2562 Smart PA - -The TAS2562 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: - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562", "ti,tas2563". - - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - - ti,imon-slot-no:- TDM TX current sense time slot. - -Optional properties: -- interrupt-parent: phandle to the interrupt controller which provides - the interrupt. -- interrupts: (GPIO) interrupt to which the chip is connected. -- shut-down: GPIO used to control the state of the device. - -Examples: -tas2562@4c { - #address-cells = <1>; - #size-cells = <0>; - compatible = "ti,tas2562"; - reg = <0x4c>; - - interrupt-parent = <&gpio1>; - interrupts = <14>; - - shut-down = <&gpio1 15 0>; - ti,imon-slot-no = <0>; -}; - diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml new file mode 100644 index 000000000000..c6168aa32954 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2562.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2562 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2562 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. + +properties: + compatible: + enum: + - ti,tas2562 + - ti,tas2563 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + shut-down: + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shut-down = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + }; + }; + -- 2.26.2

1 0

[RFC PATCH] ASoC: fsl_asrc_dma: Fix warning "Cannot create DMA dma:tx symlink"
by Shengjiu Wang 09 Jun '20

09 Jun '20

The issue log is: [ 48.021506] CPU: 0 PID: 664 Comm: aplay Not tainted 5.7.0-rc1-13120-g12b434cbbea0 #343 [ 48.031063] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 48.037638] [<c0110dd8>] (unwind_backtrace) from [<c010b8ec>] (show_stack+0x10/0x14) [ 48.045413] [<c010b8ec>] (show_stack) from [<c0557fc0>] (dump_stack+0xe4/0x118) [ 48.052757] [<c0557fc0>] (dump_stack) from [<c032aeb4>] (sysfs_warn_dup+0x50/0x64) [ 48.060357] [<c032aeb4>] (sysfs_warn_dup) from [<c032b1a8>] (sysfs_do_create_link_sd+0xc8/0xd4) [ 48.069086] [<c032b1a8>] (sysfs_do_create_link_sd) from [<c05dc668>] (dma_request_chan+0xb0/0x210) [ 48.078068] [<c05dc668>] (dma_request_chan) from [<c05dc7d0>] (dma_request_slave_channel+0x8/0x14) [ 48.087060] [<c05dc7d0>] (dma_request_slave_channel) from [<c09d5cd4>] (fsl_asrc_dma_hw_params+0x1dc/0x434) [ 48.096831] [<c09d5cd4>] (fsl_asrc_dma_hw_params) from [<c09c143c>] (soc_pcm_hw_params+0x4b0/0x650) [ 48.105903] [<c09c143c>] (soc_pcm_hw_params) from [<c09c36a8>] (dpcm_fe_dai_hw_params+0x70/0xe4) [ 48.114715] [<c09c36a8>] (dpcm_fe_dai_hw_params) from [<c099b274>] (snd_pcm_hw_params+0x158/0x418) [ 48.123701] [<c099b274>] (snd_pcm_hw_params) from [<c099c5a0>] (snd_pcm_ioctl+0x734/0x183c) [ 48.132079] [<c099c5a0>] (snd_pcm_ioctl) from [<c029ff94>] (ksys_ioctl+0x2ac/0xb98) [ 48.139765] [<c029ff94>] (ksys_ioctl) from [<c0100080>] (ret_fast_syscall+0x0/0x28) [ 48.147440] Exception stack(0xed3c5fa8 to 0xed3c5ff0) [ 48.152515] 5fa0: bec28670 00e92870 00000004 c25c4111 bec28670 0002000f [ 48.160716] 5fc0: bec28670 00e92870 00e92820 00000036 0000bb80 00000000 0002c2f8 00000003 [ 48.168913] 5fe0: b6f4d3fc bec2853c b6ee7655 b6e22cf8 [ 48.174236] fsl-esai-dai 2024000.esai: Cannot create DMA dma:tx symlink The dma channel is already requested by Back-End cpu dai driver, if fsl_asrc_dma requests dma chan with same dma:tx symlink, then this warning comes out. The warning is added by commit 71723a96b8b1 ("dmaengine: Create symlinks between DMA channels and slaves") commit bad83565eafe ("dmaengine: Cleanups for the slave <-> channel symlink support") As the dma channel is requested by Back-End, we need to reuse the channel and avoid to request a new one, then the issue can be fixed. In order to get the dma channel which is already requested in Back-End. we export two functions (snd_soc_lookup_component_nolocked and soc_component_to_pcm), if we can get the dma channel, then reuse it. if can't, then request a new one. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- include/sound/dmaengine_pcm.h | 11 ++++++ include/sound/soc.h | 2 ++ sound/soc/fsl/fsl_asrc_common.h | 2 ++ sound/soc/fsl/fsl_asrc_dma.c | 49 +++++++++++++++++++++------ sound/soc/soc-core.c | 3 +- sound/soc/soc-generic-dmaengine-pcm.c | 12 ------- 6 files changed, 55 insertions(+), 24 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index b65220685920..8c5e38180fb0 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -161,4 +161,15 @@ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, #define SND_DMAENGINE_PCM_DRV_NAME "snd_dmaengine_pcm" +struct dmaengine_pcm { + struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1]; + const struct snd_dmaengine_pcm_config *config; + struct snd_soc_component component; + unsigned int flags; +}; + +static inline struct dmaengine_pcm *soc_component_to_pcm(struct snd_soc_component *p) +{ + return container_of(p, struct dmaengine_pcm, component); +} #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index 74868436ac79..565612a8d690 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -444,6 +444,8 @@ int devm_snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *component_driver, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_component(struct device *dev); +struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev, + const char *driver_name); struct snd_soc_component *snd_soc_lookup_component(struct device *dev, const char *driver_name); diff --git a/sound/soc/fsl/fsl_asrc_common.h b/sound/soc/fsl/fsl_asrc_common.h index 77665b15c8db..09512bc79b80 100644 --- a/sound/soc/fsl/fsl_asrc_common.h +++ b/sound/soc/fsl/fsl_asrc_common.h @@ -32,6 +32,7 @@ enum asrc_pair_index { * @dma_chan: inputer and output DMA channels * @dma_data: private dma data * @pos: hardware pointer position + * @req_dma_chan_dev_to_dev: flag for release dev_to_dev chan * @private: pair private area */ struct fsl_asrc_pair { @@ -45,6 +46,7 @@ struct fsl_asrc_pair { struct dma_chan *dma_chan[2]; struct imx_dma_data dma_data; unsigned int pos; + bool req_dma_chan_dev_to_dev; void *private; }; diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index d6a3fc5f87e5..3ad862225326 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -135,6 +135,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct snd_dmaengine_dai_dma_data *dma_params_be = NULL; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; + struct snd_soc_component *component_be = NULL; struct fsl_asrc *asrc = pair->asrc; struct dma_slave_config config_fe, config_be; enum asrc_pair_index index = pair->index; @@ -142,7 +143,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, int stream = substream->stream; struct imx_dma_data *tmp_data; struct snd_soc_dpcm *dpcm; - struct dma_chan *tmp_chan; + struct dma_chan *tmp_chan = NULL, *tmp_chan_new = NULL; struct device *dev_be; u8 dir = tx ? OUT : IN; dma_cap_mask_t mask; @@ -160,6 +161,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, substream_be = snd_soc_dpcm_get_substream(be, stream); dma_params_be = snd_soc_dai_get_dma_data(dai, substream_be); dev_be = dai->dev; + component_be = snd_soc_lookup_component_nolocked(dev_be, SND_DMAENGINE_PCM_DRV_NAME); break; } @@ -205,10 +207,16 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, */ if (!asrc->use_edma) { /* Get DMA request of Back-End */ - tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); + if (component_be) + tmp_chan = soc_component_to_pcm(component_be)->chan[substream->stream]; + if (!tmp_chan) { + tmp_chan_new = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); + tmp_chan = tmp_chan_new; + } tmp_data = tmp_chan->private; pair->dma_data.dma_request = tmp_data->dma_request; - dma_release_channel(tmp_chan); + if (tmp_chan_new) + dma_release_channel(tmp_chan_new); /* Get DMA request of Front-End */ tmp_chan = asrc->get_dma_channel(pair, dir); @@ -220,9 +228,26 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, pair->dma_chan[dir] = dma_request_channel(mask, filter, &pair->dma_data); + pair->req_dma_chan_dev_to_dev = true; } else { - pair->dma_chan[dir] = - asrc->get_dma_channel(pair, dir); + /* With EDMA, there is two dma channels can be used for p2p, + * one is from ASRC, one is from another peripheral + * (ESAI or SAI) previously we select the dma channel of ASRC, + * but find an issue for ideal ratio case, there is no control + * for data copy speed, the speed is faster than sample + * frequency. + * + * So we switch to dma channel of peripheral (ESAI or SAI), + * that copy speed of DMA is controlled by data consumption + * speed in the peripheral FIFO. + */ + pair->req_dma_chan_dev_to_dev = false; + if (component_be) + pair->dma_chan[dir] = soc_component_to_pcm(component_be)->chan[substream->stream];; + if (!pair->dma_chan[dir]) { + pair->dma_chan[dir] = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); + pair->req_dma_chan_dev_to_dev = true; + } } if (!pair->dma_chan[dir]) { @@ -273,19 +298,21 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, static int fsl_asrc_dma_hw_free(struct snd_soc_component *component, struct snd_pcm_substream *substream) { + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; + u8 dir = tx ? OUT : IN; snd_pcm_set_runtime_buffer(substream, NULL); - if (pair->dma_chan[IN]) - dma_release_channel(pair->dma_chan[IN]); + if (pair->dma_chan[!dir]) + dma_release_channel(pair->dma_chan[!dir]); - if (pair->dma_chan[OUT]) - dma_release_channel(pair->dma_chan[OUT]); + if (pair->dma_chan[dir] && pair->req_dma_chan_dev_to_dev) + dma_release_channel(pair->dma_chan[dir]); - pair->dma_chan[IN] = NULL; - pair->dma_chan[OUT] = NULL; + pair->dma_chan[!dir] = NULL; + pair->dma_chan[dir] = NULL; return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b07eca2c6ccc..d4c73e86d058 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -310,7 +310,7 @@ struct snd_soc_component *snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd, } EXPORT_SYMBOL_GPL(snd_soc_rtdcom_lookup); -static struct snd_soc_component +struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev, const char *driver_name) { struct snd_soc_component *component; @@ -329,6 +329,7 @@ static struct snd_soc_component return found_component; } +EXPORT_SYMBOL_GPL(snd_soc_lookup_component_nolocked); struct snd_soc_component *snd_soc_lookup_component(struct device *dev, const char *driver_name) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index f728309a0833..80a4e71f2d95 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -21,18 +21,6 @@ */ #define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(31) -struct dmaengine_pcm { - struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1]; - const struct snd_dmaengine_pcm_config *config; - struct snd_soc_component component; - unsigned int flags; -}; - -static struct dmaengine_pcm *soc_component_to_pcm(struct snd_soc_component *p) -{ - return container_of(p, struct dmaengine_pcm, component); -} - static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, struct snd_pcm_substream *substream) { -- 2.21.0

2 1

[PATCH] soc/meson: add missing free_irq() in error path
by Pavel Machek 09 Jun '20

09 Jun '20

free_irq() is missing in case of error, fix that. Signed-off-by: Pavel Machek (CIP) <pavel(a)denx.de> diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 2e9b56b29d31..b2e867113226 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -249,7 +249,7 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Enable pclk to access registers and clock the fifo ip */ ret = clk_prepare_enable(fifo->pclk); if (ret) - return ret; + goto free_irq; /* Setup status2 so it reports the memory pointer */ regmap_update_bits(fifo->map, FIFO_CTRL1, @@ -269,8 +269,14 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Take memory arbitror out of reset */ ret = reset_control_deassert(fifo->arb); if (ret) - clk_disable_unprepare(fifo->pclk); + goto free_clk; + + return 0; +free_clk: + clk_disable_unprepare(fifo->pclk); +free_irq: + free_irq(fifo->irq, ss); return ret; } EXPORT_SYMBOL_GPL(axg_fifo_pcm_open); -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

3 4

Re: [PATCH 4/5] soundwire: qcom: avoid dependency on CONFIG_SLIMBUS
by Srinivas Kandagatla 09 Jun '20

09 Jun '20

On 08/06/2020 21:43, Jonathan Marek wrote: > The driver may be used without slimbus, so don't depend on slimbus. > > Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> > --- > drivers/soundwire/Kconfig | 1 - > drivers/soundwire/qcom.c | 5 +++++ > 2 files changed, 5 insertions(+), 1 deletion(-) > > diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig > index fa2b4ab92ed9..d121cf739090 100644 > --- a/drivers/soundwire/Kconfig > +++ b/drivers/soundwire/Kconfig > @@ -33,7 +33,6 @@ config SOUNDWIRE_INTEL > > config SOUNDWIRE_QCOM > tristate "Qualcomm SoundWire Master driver" > - depends on SLIMBUS > depends on SND_SOC Why not move this to imply SLIMBUS this will give more flexibility! > help > SoundWire Qualcomm Master driver. > diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c > index 14334442615f..ac81c64768ea 100644 > --- a/drivers/soundwire/qcom.c > +++ b/drivers/soundwire/qcom.c > @@ -769,13 +769,18 @@ static int qcom_swrm_probe(struct platform_device *pdev) > if (!ctrl) > return -ENOMEM; > > +#ifdef CONFIG_SLIMBUS > if (dev->parent->bus == &slimbus_bus) { > +#else > + if (false) { > +#endif May be you can do bit more cleanup here, which could endup like: ctrl->regmap = dev_get_regmap(dev->parent, NULL); if (!ctrl->regmap) { res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res) { ctrl->mmio = devm_ioremap_resource(dev, res); if (IS_ERR(ctrl->mmio)) { dev_err(dev, "No valid mem resource found\n"); return PTR_ERR(ctrl->mmio); } ctrl->reg_read = qcom_swrm_cpu_reg_read; ctrl->reg_write = qcom_swrm_cpu_reg_write; } else { dev_err(dev, "No valid slim resource found\n"); return -EINVAL; } } else { ctrl->reg_read = qcom_swrm_ahb_reg_read; ctrl->reg_write = qcom_swrm_ahb_reg_write; } thanks, srini > ctrl->reg_read = qcom_swrm_ahb_reg_read; > ctrl->reg_write = qcom_swrm_ahb_reg_write; > ctrl->regmap = dev_get_regmap(dev->parent, NULL); > if (!ctrl->regmap) > return -EINVAL; > } else { > + > res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > > ctrl->reg_read = qcom_swrm_cpu_reg_read; >

1 0

Re: [PATCH 0/5] soundwire: qcom: add mmio support
by Srinivas Kandagatla 09 Jun '20

09 Jun '20

Thanks Jonathan for the patches, On 08/06/2020 21:43, Jonathan Marek wrote: > This adds initial support for soundwire device on sm8250. > One thing off my list!! > Tested with the "wsa" sdw device, which is simpler than the others. WSA881x? did you test both enumeration and streaming? Are you planing to add any new WSA or WCD codec support for this SoC? thanks, srini > > Jonathan Marek (5): > soundwire: qcom: fix abh/ahb typo > soundwire: qcom: add support for mmio soundwire devices > soundwire: qcom: add v1.5.1 compatible > soundwire: qcom: avoid dependency on CONFIG_SLIMBUS > soundwire: qcom: enable CPU interrupts for mmio devices > > drivers/soundwire/Kconfig | 1 - > drivers/soundwire/qcom.c | 42 +++++++++++++++++++++++++++++++++++---- > 2 files changed, 38 insertions(+), 5 deletions(-) >

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