February 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 00/17] ASoC: Intel: AVS - Audio DSP for cAVS
by Cezary Rojewski 01 Mar '22

01 Mar '22

A continuation of cleanup work of Intel SST solutions found in sound/soc/intel/. With two major chapters released last year catpt [1] and removal of haswell solution [2], time has come for Skylake-driver. Througout 2019, 2020 and 2021 Skylake-driver has had many fixes applied and even attempts of refactors as seen in fundamental overhaul [3], IPC flow adjustments [4] and LARGE_CONFIG overhaul [5] series. Unfortunately, story repeats itself - problems are found within the core of a driver. Painting it with different colors does not change the fact that is it still a house of cards. As changes needed to address those issues would make Skylake solution incompatible with its previous revisions, a decision has been made to provide a new solution instead. In time it would deprecate and replace Skylake-driver. That solution has been called AVS - from AudioDSP architecture name: Audio-Voice-Speech. It is meant to provide support for the exact same range of platforms as its predecessor: SKL, KBL, AML and APL. Note: this series is dependent upon HDA-series [6] which exposes several codec-organization functions allowing for reduced code size on avs-driver side. Note: this series does not add fully functional driver as its size would get out of control. Here, focus is put on adding IPC protocol and code loading code. Changes v2 -> v1: Almost all updates here are thanks to feedback from Pierre. - several comments and few kernel-docs have been added in areas which felt more or less unclear - avs_ipc_wait_busy_completion() now spins up to 'repeats_left' number of times before giving up - 'adsp_ba' field of struct avs_dev has been renamed to 'dsp_ba' - 'dops' field of struct avs_spec has been renamed to 'dsp_ops' - IPC abstraction has been simplified: SKL_ADSP_REG_HIP* regs are used directly - fixed allnoconfig with AVS enabled compilation issues - fixed code loading error paths: previously requested firmware is now released before function return to the caller - code and function arguments tied to D0IX support have been removed from this patchset and will be part of followup series adding that feature instead - enriched dev_err() messages in avs_dsp_get_core() and avs_dsp_put_core() - numerous wording fixes used in power/reset/stall DSP operations Changes RFC v1 [7]: -> v1: - separated HDA codec-organization patches, path and topology handling, PCM and complementary features such as recovery from this series to ease the review process - fixed EXPORT_SYMBOL_GPL for exported members of ASoC framework - result of stall() is now checked when sending ROM message - result of snd_hdac_ext_stream_set_spib() is now checked when loading basefw - if basefw is not ready, notification processing is now skipped - documented several topology parsing helpers Changes [internal] RFC v2 -> [public] RFC v1: - dropped any sysfs related changes from this series, moved to follow up one - dropped entire subscription-mechanism found in ipc.c. Handlers that are delegated to service certain firmware notifications are now called directly - fixed kernel doc for snd_soc_dapm_new_dai_widgets() as reported by ikp - prefixed snd_hda_codec_device_init() as suggested by Amadeo - improved comments for d0ix transitions for APL-based platforms as suggested by Pierre - a ton of spelling related fixes in most of the commit messages - fixed remaining warnings pointed by scan-build (variable assigned but not used) - replaced most of 'cAVS X.Y' expression usages with 'platform-based' equivalents as suggested by Pierre e.g.: cAVS 1.5 -> SKL-based Changes [internal] RFC v1 -> [internal] RFC v2: - fixed memleak caused by lack of kfree(vols) if memory allocation fails in avs_peakvol_create() as reported by Curtis - fixed missing 'i' iterator incrementation in avs_widget_ready() causing reference loss as reported by Curtis - replace hardcode: 0x40 usage with snd_hdac_calc_stream_format as suggested by Curtis. In consequence, readability for all code loading (CL) procedues has increased and such approach auto-documents the CL stream preparation - updated behavior of all index-fetching functions found in utils.c: avs_module_entry_index(), avs_module_id_entry_index() and follow ups: avs_get_module_entry(), avs_get_module_id_entry() to better conform to linux-kernel standard when no entry is found (return -ENOENT) rather than C++ standard (return -1, what in kernel case translated to -EPERM) as suggested by Curtis and Peter - several suggestions have been made regarding spacing, and so far, I've agreed and applied with all of them. None proposed seemed out of place or redundant - avs_path_stop() renamed to avs_path_pause() pipeline states are represented by RESET/PAUSED/RUNNING. avs_path_reset() and avs_path_run() were already there and avs_path_stop() just didn't look cohesive - added missing parsers for num_output_pin and num_input_pin which are required for custom modules such as WAVES or DSM - dropped 'priv_param_length' from custom module descriptor as this field is obsolete in firmware - parse_dictionary() has been split into parse_dictionary_header() and parse_dictionary_entries() to drop code duplication present in several parsing function which could not re-use entire parse_dictionary() - added avs_tplg_vendor_array_lookup_next() and avs_tplg_vendor_entry_next() to drop code duplicated present in several parsing functions. This change greatly impacted readability of all parsers - parsing helpers such avs_tplg_vendor_array_lookup() now return offset by updating specified in function argument list u32 *offset variable. This is to address problem when u32 offset would be greater than max int, which is the return type for these functions - AVS_DEFINE_PTR_PARSER() macro has been introduced to drop code duplication for all ptr-parsing users - all struct avs_path_module creators have had their declaration updated: function argument *owner ceased to exist as it could already be accessed by mod->owner - fixed the order of operation for conditional paths (e.g.: echo reference) so these are no longer controlled by "source" path and instead are impacted by state changes of source and sink paths both. Previously only source path e.g. playback sourcing echo reference would trigger RUNNING status for conditional path. Equivalent RUNNING on WoV path which is in this case sink path, would not do so, leading to order-of-operation problems. Behavior has been changed to: both source and sink need to be RUNNING for conditional path to be set to RUNNING too. PAUSED for either source or sink will cause PAUSED transition for conditional path. - to achieve the above, path states are now saved in 'state' i.e. new u32 field for struct avs_path - resigned from fw_filename field usage in favour of newly added tplg_filename for machine board descriptors as suggested by Pierre - platform descriptor fields have had their names update better reflect their purpose as suggested by Pierre - fixed comp_list missing locking when manipulated - all message senders now accept request as pointer as suggeseted by Peter - resigned of AZX_ usage for all ADSP-related registers, leaving them only for HOST memory space related operations - fixed disable path for core DSP operations: power/reset/stall as reported by Peter - safety when locking between received responses (reply vs notification) has been lowered as suggested by Pierre. Most usages are not performed in IRQ context and none is done in hard-IRQ one - s/master/main/ plus AVS_MAIN_CORE_MASK has replaced ->master_mask - several functions have had their logging updated - logs have been moved to lower level functions as suggested by Pierre - hdac_ext_stream usage has been streamlined to estream, hdac_streams are represented by hstream instead - hw_params() are resilient to scenarios when they are called mutliple times as reported by Pierre - avs_dsp_enable() now collapses if any of its steps fails as reported by Pierre and Peter - avs_module_ida_empty() now returns value of type bool as suggested by Bard [1]: https://www.spinics.net/lists/alsa-devel/msg116440.html [2]: https://www.spinics.net/lists/alsa-devel/msg116901.html [3]: https://www.spinics.net/lists/alsa-devel/msg94199.html [4]: https://www.spinics.net/lists/alsa-devel/msg92588.html [5]: https://lore.kernel.org/all/20190808181549.12521-1-cezary.rojewski@intel.co… [6]: https://lore.kernel.org/alsa-devel/20220207114906.3759800-1-cezary.rojewski… [7]: https://lore.kernel.org/all/20211208111301.1817725-1-cezary.rojewski@intel.… Cezary Rojewski (17): ALSA: hda: Add helper macros for DSP capable devices ASoC: Export DAI register and widget ctor and dctor functions ASoC: Intel: Introduce AVS driver ASoC: Intel: avs: Inter process communication ASoC: Intel: avs: Add code loading requests ASoC: Intel: avs: Add pipeline management requests ASoC: Intel: avs: Add module management requests ASoC: Intel: avs: Add power management requests ASoC: Intel: avs: Add ROM requests ASoC: Intel: avs: Add basefw runtime-parameter requests ASoC: Intel: avs: Firmware resources management utilities ASoC: Intel: avs: Declare module configuration types ASoC: Intel: avs: Dynamic firmware resources management ASoC: Intel: avs: General code loading flow ASoC: Intel: avs: Implement CLDMA transfer ASoC: Intel: avs: Code loading over CLDMA ASoC: Intel: avs: Code loading over HDA include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 50 +++ include/sound/soc-dapm.h | 1 + sound/soc/intel/Kconfig | 12 + sound/soc/intel/Makefile | 1 + sound/soc/intel/avs/Makefile | 6 + sound/soc/intel/avs/avs.h | 248 +++++++++++ sound/soc/intel/avs/cldma.c | 317 ++++++++++++++ sound/soc/intel/avs/cldma.h | 29 ++ sound/soc/intel/avs/core.c | 62 +++ sound/soc/intel/avs/dsp.c | 303 +++++++++++++ sound/soc/intel/avs/ipc.c | 393 +++++++++++++++++ sound/soc/intel/avs/loader.c | 604 +++++++++++++++++++++++++ sound/soc/intel/avs/messages.c | 697 +++++++++++++++++++++++++++++ sound/soc/intel/avs/messages.h | 753 ++++++++++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 75 ++++ sound/soc/intel/avs/utils.c | 301 +++++++++++++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 15 + 19 files changed, 3870 insertions(+) create mode 100644 sound/soc/intel/avs/Makefile create mode 100644 sound/soc/intel/avs/avs.h create mode 100644 sound/soc/intel/avs/cldma.c create mode 100644 sound/soc/intel/avs/cldma.h create mode 100644 sound/soc/intel/avs/core.c create mode 100644 sound/soc/intel/avs/dsp.c create mode 100644 sound/soc/intel/avs/ipc.c create mode 100644 sound/soc/intel/avs/loader.c create mode 100644 sound/soc/intel/avs/messages.c create mode 100644 sound/soc/intel/avs/messages.h create mode 100644 sound/soc/intel/avs/registers.h create mode 100644 sound/soc/intel/avs/utils.c -- 2.25.1

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[PATCH v5 0/2] Add support for SoundWire1.6 audio cgcr register control
by Srinivasa Rao Mandadapu 01 Mar '22

01 Mar '22

This patch series is to add v1.6.0 compatible name for qcom soundwire driver and corresponding dt bindings. Changes Since V4: -- Dropped audio cgcr control patch due to dependency on clock patches. -- Update dt-bindings as per new reset control properties. Changes Since V3: -- Add v1.6.0 compatible name and soundwire data structure. -- Change macro define name properly. -- Update bindings for new property. -- Change commit message description. -- Change signedoff by sequence. Changes since v2: -- Update error check after ioremap. Changes since v1: -- Add const name to mask value. Srinivasa Rao Mandadapu (2): soundwire: qcom: Add compatible name for v1.6.0 dt-bindings: soundwire: qcom: Add bindings for audio CSR reset control property Documentation/devicetree/bindings/soundwire/qcom,sdw.txt | 12 ++++++++++++ drivers/soundwire/qcom.c | 6 ++++++ 2 files changed, 18 insertions(+) -- 2.7.4

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[PATCH v3] ASoC: codecs: wsa881x: add runtime pm support
by Srinivas Kandagatla 01 Mar '22

01 Mar '22

WSA SoundWire Controller does not support Clock stop and performs a soft reset on suspend resume path. Its recommended that WSA881x codecs connected to this are also reset using a hard reset during suspend resume. So this codec driver performs a hard reset during suspend resume cycle. --- Hi Mark, Looks like the runtime pm series was partly applied, so am sending just this leftover patch with requested changes. Changes since v2: removed setting up simple_clk_stop_capable flag update log srini sound/soc/codecs/wsa881x.c | 53 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 0222370ff95d..616b26c70c3b 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -11,6 +11,7 @@ #include <linux/of_gpio.h> #include <linux/regmap.h> #include <linux/slab.h> +#include <linux/pm_runtime.h> #include <linux/soundwire/sdw.h> #include <linux/soundwire/sdw_registers.h> #include <linux/soundwire/sdw_type.h> @@ -198,6 +199,7 @@ #define WSA881X_OCP_CTL_TIMER_SEC 2 #define WSA881X_OCP_CTL_TEMP_CELSIUS 25 #define WSA881X_OCP_CTL_POLL_TIMER_SEC 60 +#define WSA881X_PROBE_TIMEOUT 1000 #define WSA881X_PA_GAIN_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -747,6 +749,12 @@ static int wsa881x_put_pa_gain(struct snd_kcontrol *kc, unsigned int mask = (1 << fls(max)) - 1; int val, ret, min_gain, max_gain; + ret = pm_runtime_get_sync(comp->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(comp->dev); + return ret; + } + max_gain = (max - ucontrol->value.integer.value[0]) & mask; /* * Gain has to set incrementally in 4 steps @@ -773,6 +781,9 @@ static int wsa881x_put_pa_gain(struct snd_kcontrol *kc, usleep_range(1000, 1010); } + pm_runtime_mark_last_busy(comp->dev); + pm_runtime_put_autosuspend(comp->dev); + return 1; } @@ -1101,6 +1112,7 @@ static int wsa881x_probe(struct sdw_slave *pdev, const struct sdw_device_id *id) { struct wsa881x_priv *wsa881x; + struct device *dev = &pdev->dev; wsa881x = devm_kzalloc(&pdev->dev, sizeof(*wsa881x), GFP_KERNEL); if (!wsa881x) @@ -1132,12 +1144,52 @@ static int wsa881x_probe(struct sdw_slave *pdev, return PTR_ERR(wsa881x->regmap); } + pm_runtime_set_autosuspend_delay(dev, 3000); + pm_runtime_use_autosuspend(dev); + pm_runtime_mark_last_busy(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + return devm_snd_soc_register_component(&pdev->dev, &wsa881x_component_drv, wsa881x_dais, ARRAY_SIZE(wsa881x_dais)); } +static int __maybe_unused wsa881x_runtime_suspend(struct device *dev) +{ + struct regmap *regmap = dev_get_regmap(dev, NULL); + struct wsa881x_priv *wsa881x = dev_get_drvdata(dev); + + gpiod_direction_output(wsa881x->sd_n, 0); + + regcache_cache_only(regmap, true); + regcache_mark_dirty(regmap); + + return 0; +} + +static int __maybe_unused wsa881x_runtime_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct regmap *regmap = dev_get_regmap(dev, NULL); + struct wsa881x_priv *wsa881x = dev_get_drvdata(dev); + + gpiod_direction_output(wsa881x->sd_n, 1); + + wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(WSA881X_PROBE_TIMEOUT)); + + regcache_cache_only(regmap, false); + regcache_sync(regmap); + + return 0; +} + +static const struct dev_pm_ops wsa881x_pm_ops = { + SET_RUNTIME_PM_OPS(wsa881x_runtime_suspend, wsa881x_runtime_resume, NULL) +}; + static const struct sdw_device_id wsa881x_slave_id[] = { SDW_SLAVE_ENTRY(0x0217, 0x2010, 0), SDW_SLAVE_ENTRY(0x0217, 0x2110, 0), @@ -1151,6 +1203,7 @@ static struct sdw_driver wsa881x_codec_driver = { .id_table = wsa881x_slave_id, .driver = { .name = "wsa881x-codec", + .pm = &wsa881x_pm_ops, } }; module_sdw_driver(wsa881x_codec_driver); -- 2.21.0

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[PATCH v16 0/9] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 28 Feb '22

28 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V15: -- Bisect patches to avoid build failure in other architectures. -- Remove redundant variables lpass variant structure. Changes Since V14: -- Split common wrapper function to separate wrapper for each handle in platform driver. -- Update cdc dma buffer handling with memremap with ioremap. -- Remove redundant error prints. -- Update irq flag. Changes Since V13: -- Change bulk cdc clock voting to individual clock voting. -- Remove redundant code, conditional check and prints. -- Fix typo errors. Changes Since V12: -- Fix arguments type mismatch. Changes Since V11: -- Fix kernel robot issue on arguments type mismatch. Changes Since V10: -- Split bulk clock voting to individual clock voting as per use case in cdc-dma driver. -- Add missing codec dma clocks. -- Update rxtx lpm buffer size. Changes Since V9: -- Change individual clock voting to bulk clock voting of lpass-sc7280 platform driver. -- Remove redundant clocks in lpass variant structure. -- Add mclk for MI2S based headset path. -- Remove unused lpass variant structure members in lpass header. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (9): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 301 ++++++++++ sound/soc/qcom/lpass-cpu.c | 253 ++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 628 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 438 ++++++++++++++ sound/soc/qcom/lpass.h | 141 +++++ 9 files changed, 1890 insertions(+), 88 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

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[PATCH v5 0/2] Add power domains support for digital macro codecs
by Srinivasa Rao Mandadapu 28 Feb '22

28 Feb '22

This patch set is to add power domains support for RX, TX and VA macros. Changes since v4: -- Fix uninitialized variable return error Changes since v3: -- Add dt-bindings support. Changes since v2: -- Remove redundant local variable. -- Update pds error handling sequence. -- Update module description. -- Clean up pds init function. -- Remove redundant arguments. Changes since v1: -- Add missing macros in Kconfig. Srinivasa Rao Mandadapu (2): ASoC: codecs: Add power domains support in digital macro codecs ASoC: qcom: dt-bindings: Add bindings for power domains in lpass digital codecs .../bindings/sound/qcom,lpass-rx-macro.yaml | 8 +++ .../bindings/sound/qcom,lpass-tx-macro.yaml | 8 +++ .../bindings/sound/qcom,lpass-va-macro.yaml | 8 +++ sound/soc/codecs/Kconfig | 7 +++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lpass-macro-common.c | 67 ++++++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 17 ++++++ sound/soc/codecs/lpass-rx-macro.c | 10 +++- sound/soc/codecs/lpass-tx-macro.c | 9 +++ sound/soc/codecs/lpass-va-macro.c | 10 ++++ sound/soc/qcom/Kconfig | 1 + 11 files changed, 146 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/lpass-macro-common.c create mode 100644 sound/soc/codecs/lpass-macro-common.h -- 2.7.4

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[PATCH V2 1/4] ASoC: amd: vg: fix for pm resume callback sequence
by Vijendar Mukunda 28 Feb '22

28 Feb '22

The previous condition is used to cross check only the active stream status for I2S HS instance playback and capture use cases. Modified logic to invoke sequence for two i2s controller instances. This also fixes warnings reported by kernel robot: "warning: variable 'frmt_val' set but not used" "warning: variable 'reg_val' set but not used" Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v1 -> v2: fixed kernel rebot warnings sound/soc/amd/vangogh/acp5x-pcm-dma.c | 66 +++++++++++++-------------- 1 file changed, 33 insertions(+), 33 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index f10de38976cb..3aa2d9a36880 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -426,51 +426,51 @@ static int acp5x_audio_remove(struct platform_device *pdev) static int __maybe_unused acp5x_pcm_resume(struct device *dev) { struct i2s_dev_data *adata; - u32 val, reg_val, frmt_val; + struct i2s_stream_instance *rtd; + u32 val; - reg_val = 0; - frmt_val = 0; adata = dev_get_drvdata(dev); if (adata->play_stream && adata->play_stream->runtime) { - struct i2s_stream_instance *rtd = - adata->play_stream->runtime->private_data; + rtd = adata->play_stream->runtime->private_data; config_acp5x_dma(rtd, SNDRV_PCM_STREAM_PLAYBACK); - switch (rtd->i2s_instance) { - case I2S_HS_INSTANCE: - reg_val = ACP_HSTDM_ITER; - frmt_val = ACP_HSTDM_TXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = ACP_I2STDM_ITER; - frmt_val = ACP_I2STDM_TXFRMT; + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_HSTDM_ITER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_HSTDM_TXFRMT); + val = acp_readl(adata->acp5x_base + ACP_HSTDM_ITER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_HSTDM_ITER); + } + } + if (adata->i2ssp_play_stream && adata->i2ssp_play_stream->runtime) { + rtd = adata->i2ssp_play_stream->runtime->private_data; + config_acp5x_dma(rtd, SNDRV_PCM_STREAM_PLAYBACK); + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_I2STDM_ITER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_I2STDM_TXFRMT); + val = acp_readl(adata->acp5x_base + ACP_I2STDM_ITER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_I2STDM_ITER); } - acp_writel((rtd->xfer_resolution << 3), - rtd->acp5x_base + reg_val); } if (adata->capture_stream && adata->capture_stream->runtime) { - struct i2s_stream_instance *rtd = - adata->capture_stream->runtime->private_data; + rtd = adata->capture_stream->runtime->private_data; config_acp5x_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); - switch (rtd->i2s_instance) { - case I2S_HS_INSTANCE: - reg_val = ACP_HSTDM_IRER; - frmt_val = ACP_HSTDM_RXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = ACP_I2STDM_IRER; - frmt_val = ACP_I2STDM_RXFRMT; + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_HSTDM_IRER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_HSTDM_RXFRMT); + val = acp_readl(adata->acp5x_base + ACP_HSTDM_IRER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_HSTDM_IRER); } - acp_writel((rtd->xfer_resolution << 3), - rtd->acp5x_base + reg_val); } - if (adata->tdm_mode == TDM_ENABLE) { - acp_writel(adata->tdm_fmt, adata->acp5x_base + frmt_val); - val = acp_readl(adata->acp5x_base + reg_val); - acp_writel(val | 0x2, adata->acp5x_base + reg_val); + if (adata->i2ssp_capture_stream && adata->i2ssp_capture_stream->runtime) { + rtd = adata->i2ssp_capture_stream->runtime->private_data; + config_acp5x_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_I2STDM_IRER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_I2STDM_RXFRMT); + val = acp_readl(adata->acp5x_base + ACP_I2STDM_IRER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_I2STDM_IRER); + } } acp_writel(1, adata->acp5x_base + ACP_EXTERNAL_INTR_ENB); return 0; -- 2.17.1

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[PATCH] ASoC: dt-bindings: renesas,rz-ssi: Document RZ/V2L SoC
by Lad Prabhakar 28 Feb '22

28 Feb '22

Document RZ/V2L SSI bindings. RZ/V2L SSI is identical to one found on the RZ/G2L SoC. No driver changes are required as generic compatible string "renesas,rz-ssi" will be used as a fallback. Signed-off-by: Lad Prabhakar <prabhakar.mahadev-lad.rj(a)bp.renesas.com> Reviewed-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- DTSI changes have been posted as part of series [0]. [0] https://patchwork.kernel.org/project/linux-renesas-soc/ patch/20220227203744.18355-10-prabhakar.mahadev-lad.rj(a)bp.renesas.com/ --- Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml index 414ff8035a4e..7e8d252f7bca 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml @@ -4,7 +4,7 @@ $id: http://devicetree.org/schemas/sound/renesas,rz-ssi.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# -title: Renesas RZ/G2L ASoC Sound Serial Interface (SSIF-2) +title: Renesas RZ/{G2L,V2L} ASoC Sound Serial Interface (SSIF-2) maintainers: - Biju Das <biju.das.jz(a)bp.renesas.com> @@ -14,6 +14,7 @@ properties: items: - enum: - renesas,r9a07g044-ssi # RZ/G2{L,LC} + - renesas,r9a07g054-ssi # RZ/V2L - const: renesas,rz-ssi reg: -- 2.17.1

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[PATCH] ASoC: twl4030: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the twl4030 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/twl4030.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e059711ff293..0ba3546ef870 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1840,13 +1840,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) old_format = twl4030_read(component, TWL4030_REG_AUDIO_IF); format = old_format; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: format &= ~(TWL4030_AIF_SLAVE_EN); format &= ~(TWL4030_CLK256FS_EN); break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: format |= TWL4030_AIF_SLAVE_EN; format |= TWL4030_CLK256FS_EN; break; @@ -2038,9 +2037,8 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, old_format = twl4030_read(component, TWL4030_REG_VOICE_IF); format = old_format; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: format &= ~(TWL4030_VIF_SLAVE_EN); break; case SND_SOC_DAIFMT_CBS_CFS: -- 2.30.2

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[PATCH] ASoC: ti: davinci-i2s: Add check for clk_enable()
by Jiasheng Jiang 28 Feb '22

28 Feb '22

As the potential failure of the clk_enable(), it should be better to check it and return error if fails. Fixes: 5f9a50c3e55e ("ASoC: Davinci: McBSP: add device tree support for McBSP") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/ti/davinci-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 6dca51862dd7..0363a088d2e0 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -708,7 +708,9 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) return -ENODEV; - clk_enable(dev->clk); + ret = clk_enable(dev->clk); + if (ret) + goto err_put_clk; dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); @@ -730,6 +732,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); err_release_clk: clk_disable(dev->clk); +err_put_clk: clk_put(dev->clk); return ret; } -- 2.25.1

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[PATCH] ASoC: amd: pcm-dma: Use platform_get_irq() to get the interrupt
by Meng Tang 28 Feb '22

28 Feb '22

platform_get_resource(pdev, IORESOURCE_IRQ, ..) relies on static allocation of IRQ resources in DT core code, this causes an issue when using hierarchical interrupt domains using "interrupts" property in the node as this bypassed the hierarchical setup and messed up the irq chaining. In preparation for removal of static setup of IRQ resource from DT core code use platform_get_irq(). Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/soc/amd/acp-pcm-dma.c | 11 ++++------- sound/soc/amd/raven/acp3x-pcm-dma.c | 8 ++------ sound/soc/amd/renoir/acp3x-pdm-dma.c | 7 ++----- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 7 ++----- 4 files changed, 10 insertions(+), 23 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 8fa2e2fde4f1..1cd2e70a57df 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1217,9 +1217,8 @@ static const struct snd_soc_component_driver acp_asoc_platform = { static int acp_audio_probe(struct platform_device *pdev) { - int status; + int status, irq; struct audio_drv_data *audio_drv_data; - struct resource *res; const u32 *pdata = pdev->dev.platform_data; if (!pdata) { @@ -1249,13 +1248,11 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->asic_type = *pdata; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + irq = platform_get_irq(pdev, 0); + if (irq < 0) return -ENODEV; - } - status = devm_request_irq(&pdev->dev, res->start, dma_irq_handler, + status = devm_request_irq(&pdev->dev, irq, dma_irq_handler, 0, "ACP_IRQ", &pdev->dev); if (status) { dev_err(&pdev->dev, "ACP IRQ request failed\n"); diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 75c06697fa09..e4f8dbf0d11d 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -394,13 +394,9 @@ static int acp3x_audio_probe(struct platform_device *pdev) if (!adata->acp3x_base) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + adata->i2s_irq = platform_get_irq(pdev, 0); + if (adata->i2s_irq < 0) return -ENODEV; - } - - adata->i2s_irq = res->start; dev_set_drvdata(&pdev->dev, adata); status = devm_snd_soc_register_component(&pdev->dev, diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 9dd22a2fa2e5..88a242538461 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -399,13 +399,10 @@ static int acp_pdm_audio_probe(struct platform_device *pdev) if (!adata->acp_base) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + adata->pdm_irq = platform_get_irq(pdev, 0); + if (adata->pdm_irq < 0) return -ENODEV; - } - adata->pdm_irq = res->start; adata->capture_stream = NULL; dev_set_drvdata(&pdev->dev, adata); diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index f10de38976cb..c8cd1777a63c 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -388,13 +388,10 @@ static int acp5x_audio_probe(struct platform_device *pdev) if (!adata->acp5x_base) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + adata->i2s_irq = platform_get_irq(pdev, 0); + if (adata->i2s_irq < 0) return -ENODEV; - } - adata->i2s_irq = res->start; dev_set_drvdata(&pdev->dev, adata); status = devm_snd_soc_register_component(&pdev->dev, &acp5x_i2s_component, -- 2.20.1

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