December 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v2] ASoC: soc-compress: prevent the potentially use of null pointer
by Jiasheng Jiang 02 Mar '22

02 Mar '22

There is one call trace that snd_soc_register_card() ->snd_soc_bind_card()->soc_init_pcm_runtime() ->snd_soc_dai_compress_new()->snd_soc_new_compress(). In the trace the 'codec_dai' transfers from card->dai_link, and we can see from the snd_soc_add_pcm_runtime() in snd_soc_bind_card() that, if value of card->dai_link->num_codecs is 0, then 'codec_dai' could be null pointer caused by index out of bound in 'asoc_rtd_to_codec(rtd, 0)'. And snd_soc_register_card() is called by various platforms. Therefore, it is better to add the check in the case of misusing. And because 'cpu_dai' has already checked in soc_init_pcm_runtime(), there is no need to check again. Adding the check as follow, then if 'codec_dai' is null, snd_soc_new_compress() will not pass through the check 'if (playback + capture != 1)', avoiding the leftover use of 'codec_dai'. Fixes: 467fece ("ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-compress.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b4f5935..67c3df1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -535,12 +535,14 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) - playback = 1; - if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) - capture = 1; + if (codec_dai) { + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + playback = 1; + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + capture = 1; + } /* * Compress devices are unidirectional so only one of the directions -- 2.7.4

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[PATCH v10 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 24 Feb '22

24 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. 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 (10): ASoC: qcom: SC7280: Update config for building codec dma drivers 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 | 275 +++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 445 +++++++++++++++ sound/soc/qcom/lpass.h | 145 +++++ 9 files changed, 1856 insertions(+), 87 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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[RESEND, v9 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 24 Feb '22

24 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. 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 (10): ASoC: qcom: SC7280: Update config for building codec dma drivers 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 | 275 +++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 441 +++++++++++++++ sound/soc/qcom/lpass.h | 162 ++++++ 9 files changed, 1869 insertions(+), 87 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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[RFC PATCH 1/3] ASoC: simple-card-utils: add support for componants provideing jack events via set_jack
by Carl Philipp Klemm 19 Feb '22

19 Feb '22

This allows componants that want a jack to report state on to do so by calling set_jack on components implementing this function. Im not entirely sure this is the right way to do this so RFC Signed-off-by: Carl Philipp Klemm <philipp(a)uvos.xyz> --- include/sound/simple_card_utils.h | 6 ++-- sound/soc/generic/simple-card-utils.c | 47 +++++++++++++++++++-------- 2 files changed, 36 insertions(+), 17 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 51b3b485a92e..547ad537613d 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -57,8 +57,8 @@ struct asoc_simple_priv { struct prop_nums num; unsigned int mclk_fs; } *dai_props; - struct asoc_simple_jack hp_jack; - struct asoc_simple_jack mic_jack; + struct asoc_simple_jack *hp_jack; + struct asoc_simple_jack *mic_jack; struct snd_soc_dai_link *dai_link; struct asoc_simple_dai *dais; struct snd_soc_dai_link_component *dlcs; @@ -173,7 +173,7 @@ int asoc_simple_parse_pin_switches(struct snd_soc_card *card, char *prefix); int asoc_simple_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, + struct asoc_simple_jack **sjack, int is_hp, char *prefix, char *pin); int asoc_simple_init_priv(struct asoc_simple_priv *priv, struct link_info *li); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 10c63b73900c..1899feba16cc 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -395,6 +395,7 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); struct asoc_simple_dai *dai; + struct snd_soc_component *component; int i, ret; for_each_prop_dai_codec(props, i, dai) { @@ -412,6 +413,21 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; + for_each_rtd_components(rtd, i, component) { + if (component->driver->set_jack) { + if (!priv->hp_jack) { + priv->hp_jack = devm_kzalloc(priv->snd_card.dev, + sizeof(*priv->hp_jack), GFP_KERNEL); + snd_soc_card_jack_new(&priv->snd_card, + "Headphones", + SND_JACK_HEADPHONE, + &priv->hp_jack->jack, + NULL, 0); + } + snd_soc_component_set_jack(component, &priv->hp_jack->jack, NULL); + } + } + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_dai_init); @@ -554,7 +570,7 @@ int asoc_simple_parse_pin_switches(struct snd_soc_card *card, EXPORT_SYMBOL_GPL(asoc_simple_parse_pin_switches); int asoc_simple_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, + struct asoc_simple_jack **sjack, int is_hp, char *prefix, char *pin) { @@ -569,8 +585,6 @@ int asoc_simple_init_jack(struct snd_soc_card *card, if (!prefix) prefix = ""; - sjack->gpio.gpio = -ENOENT; - if (is_hp) { snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); pin_name = pin ? pin : "Headphones"; @@ -588,21 +602,26 @@ int asoc_simple_init_jack(struct snd_soc_card *card, return -EPROBE_DEFER; if (gpio_is_valid(det)) { - sjack->pin.pin = pin_name; - sjack->pin.mask = mask; + struct asoc_simple_jack *sjack_d; + + sjack = devm_kzalloc(dev, sizeof(*(*sjack)), GFP_KERNEL); + sjack_d = *sjack; + + sjack_d->pin.pin = pin_name; + sjack_d->pin.mask = mask; - sjack->gpio.name = gpio_name; - sjack->gpio.report = mask; - sjack->gpio.gpio = det; - sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); - sjack->gpio.debounce_time = 150; + sjack_d->gpio.name = gpio_name; + sjack_d->gpio.report = mask; + sjack_d->gpio.gpio = det; + sjack_d->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); + sjack_d->gpio.debounce_time = 150; snd_soc_card_jack_new(card, pin_name, mask, - &sjack->jack, - &sjack->pin, 1); + &sjack_d->jack, + &sjack_d->pin, 1); - snd_soc_jack_add_gpios(&sjack->jack, 1, - &sjack->gpio); + snd_soc_jack_add_gpios(&sjack_d->jack, 1, + &sjack_d->gpio); } return 0; -- 2.34.1

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[PATCH 0/1] sound: add quirk for Huawei D15
by Mauro Carvalho Chehab 18 Feb '22

18 Feb '22

Hi Pierre-Louis, Based on my tests, Huawei D15 (Intel) uses SSP0 on es8336. Add a quirk for it. Please notice that, currently, only the internal speaker is working. The topology for the internal microphone and for the headphones is wrong. Enabling/disabling the other two quirks (GPIO and/or DMIC) doesn't cause any audible results, nor change the devices listed on pavucontrol (tested with pipewire-pulse). Mauro Carvalho Chehab (1): ASoC: Intel: sof_es8336: add quirk for Huawei D15 2021 sound/soc/intel/boards/sof_es8336.c | 8 ++++++++ 1 file changed, 8 insertions(+) -- 2.33.1

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[PATCH 1/4] ALSA: hda/realtek: fix mute/micmute LEDs for HP 855 G8
by Jeremy Szu 08 Feb '22

08 Feb '22

The HP EliteBook 855 G8 Notebook PC is using ALC285 codec which needs ALC285_FIXUP_HP_MUTE_LED fixup to make it works. After applying the fixup, the mute/micmute LEDs work good. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 552e2cb73291..9d68f591c6bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8291,6 +8291,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.31.1

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[RFC 00/37] ASoC: Intel: AVS - Audio DSP for cAVS
by Cezary Rojewski 07 Feb '22

07 Feb '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. Several functions found within HDAudio and ASoC framework have been exported and are re-used later by the avs-driver to prevent code being duplicated in this solution. All of these act as driver dependencies and are found at the beginning of the series to make it clear what's needed for avs-driver to compile. Note: this series is dependent upon NHLT-series [6] which was recently merged by Takashi to his for-next branch yet is unavailable on current broonie/for-next. Note: this series does not add fully functional driver as its size would get out of control. Focus is put on adding new code. A SND_SOC_INTEL_SKYLAKE_FAMILY=n dependency is added so no probing attempts are performed. Until all patches are merged, Skylake-driver remains the recommended option. Series which will later follow this one, focus on filling all the remaining functionality gaps and provide filesystems support. Also, a range of machine boards will be added to match range of configurations supported by the avs-driver. Once avs-driver and its collaterals are merged, skylake-driver deprecation and removal process begins, along with all other components that are connected to it e.g.: skylake-driver-only machine boards. As AudioDSP firmware which avs-driver communicates with supports a wide range of audio formats, module configurations and multi-pipeline streams, couple of new concepts are introduce to enable all those functionalities: - Path template and path variants - Runtime path - Conditional path - Granular sound cards (as opposed to 'super-cards') These are later better explained by their respective patches: 'Topology parsing', 'Path management', 'Conditional-path support' and 'Machine board registration'. A 'path' represents a DSP side of audio stream in runtime - is a logical container for pipelines which are themselves composed of modules - processing units. Due to high range of possible audio format combinations, there can be more variants of given path (and thus, its pipelines and modules) than total number of pipelines and module instances which firmware supports concurrently, all the instance IDs are allocated dynamically with help of IDA interface. 'Path templates' come from topology file and describe a pattern which is later used to actually create runtime 'path'. To support modern audio usecases such as WaveOnVoice and EchoReference, conditional paths concept came into existence. These work similarly to standard paths except are a consequence of other paths being created or deleted, rather then being created when userspace app opens specific FE for streaming. Their state machine is controlled by source and sink paths which created them in the first place. Granular machine boards is a contrast to 'super-card' idea which is currently widely used throughout Intel ASoC drivers. Major reasons are: complexity reduction (each board now focuses on a single, concrete device) and overall reduction of topology file size when entire configuration is taken into account. This has functional benefits too: one card failing won't prevent others from probing and being operative. 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/all/YaDq7L1Mu++3UBL7@sirena.org.uk/T/ Cezary Rojewski (37): ALSA: hda: Add snd_hdac_ext_bus_link_at() helper ALSA: hda: Update and expose snd_hda_codec_device_init() ALSA: hda: Update and expose codec register procedures ALSA: hda: Expose codec cleanup and power-save functions 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: Topology parsing ASoC: Intel: avs: Path management ASoC: Intel: avs: Conditional-path support 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 ASoC: Intel: avs: Generic soc component driver ASoC: Intel: avs: Generic PCM FE operations ASoC: Intel: avs: non-HDA PCM BE operations ASoC: Intel: avs: HDA PCM BE operations ASoC: Intel: avs: Coredump and recovery flow ASoC: Intel: avs: Prepare for firmware tracing ASoC: Intel: avs: D0ix power state support ASoC: Intel: avs: Event tracing ASoC: Intel: avs: Machine board registration ASoC: Intel: avs: PCI driver implementation ASoC: Intel: avs: Power management ASoC: Intel: avs: SKL-based platforms support ASoC: Intel: avs: APL-based platforms support .../ABI/testing/sysfs-bus-pci-devices-avs | 24 + include/sound/hda_codec.h | 11 +- include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 50 + include/sound/soc-acpi.h | 2 + include/sound/soc-dapm.h | 1 + include/uapi/sound/intel/avs/tokens.h | 147 ++ sound/hda/ext/hdac_ext_controller.c | 31 +- sound/pci/hda/hda_codec.c | 93 +- sound/pci/hda/hda_local.h | 2 - sound/soc/codecs/hdac_hda.c | 2 +- sound/soc/intel/Kconfig | 19 + sound/soc/intel/Makefile | 1 + sound/soc/intel/avs/Makefile | 12 + sound/soc/intel/avs/apl.c | 244 +++ sound/soc/intel/avs/avs.h | 331 ++++ sound/soc/intel/avs/board_selection.c | 459 +++++ sound/soc/intel/avs/cldma.c | 328 ++++ sound/soc/intel/avs/cldma.h | 29 + sound/soc/intel/avs/core.c | 737 +++++++ sound/soc/intel/avs/dsp.c | 326 ++++ sound/soc/intel/avs/ipc.c | 612 ++++++ sound/soc/intel/avs/loader.c | 672 +++++++ sound/soc/intel/avs/messages.c | 674 +++++++ sound/soc/intel/avs/messages.h | 813 ++++++++ sound/soc/intel/avs/path.c | 1287 +++++++++++++ sound/soc/intel/avs/path.h | 85 + sound/soc/intel/avs/pcm.c | 1240 ++++++++++++ sound/soc/intel/avs/registers.h | 83 + sound/soc/intel/avs/skl.c | 127 ++ sound/soc/intel/avs/topology.c | 1700 +++++++++++++++++ sound/soc/intel/avs/topology.h | 207 ++ sound/soc/intel/avs/trace.c | 34 + sound/soc/intel/avs/trace.h | 158 ++ sound/soc/intel/avs/utils.c | 305 +++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 15 + 37 files changed, 10824 insertions(+), 40 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-pci-devices-avs create mode 100644 include/uapi/sound/intel/avs/tokens.h create mode 100644 sound/soc/intel/avs/Makefile create mode 100644 sound/soc/intel/avs/apl.c create mode 100644 sound/soc/intel/avs/avs.h create mode 100644 sound/soc/intel/avs/board_selection.c 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/path.c create mode 100644 sound/soc/intel/avs/path.h create mode 100644 sound/soc/intel/avs/pcm.c create mode 100644 sound/soc/intel/avs/registers.h create mode 100644 sound/soc/intel/avs/skl.c create mode 100644 sound/soc/intel/avs/topology.c create mode 100644 sound/soc/intel/avs/topology.h create mode 100644 sound/soc/intel/avs/trace.c create mode 100644 sound/soc/intel/avs/trace.h create mode 100644 sound/soc/intel/avs/utils.c -- 2.25.1

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[PATCH] ASoC: Intel: atom: Remove redundant check to simplify the code
by Tang Bin 25 Jan '22

25 Jan '22

In the function sst_platform_get_resources(), if platform_get_irq() failed, the return should not be zero, as the example in platform.c is * int irq = platform_get_irq(pdev, 0) * if (irq < 0) * return irq; So remove the redundant check to simplify the code. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 3be64430c..696d547c5 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -226,8 +226,8 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) /* Find the IRQ */ ctx->irq_num = platform_get_irq(pdev, ctx->pdata->res_info->acpi_ipc_irq_index); - if (ctx->irq_num <= 0) - return ctx->irq_num < 0 ? ctx->irq_num : -EIO; + if (ctx->irq_num < 0) + return ctx->irq_num; return 0; } -- 2.20.1.windows.1

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[PATCH v1 1/1] ASoC: madera: Replace kernel.h with the necessary inclusions
by Andy Shevchenko 24 Jan '22

24 Jan '22

When kernel.h is used in the headers it adds a lot into dependency hell, especially when there are circular dependencies are involved. Replace kernel.h inclusion with the list of what is really being used. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- include/sound/madera-pdata.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/madera-pdata.h b/include/sound/madera-pdata.h index e3060f48f108..58398d80c3de 100644 --- a/include/sound/madera-pdata.h +++ b/include/sound/madera-pdata.h @@ -9,7 +9,7 @@ #ifndef MADERA_CODEC_PDATA_H #define MADERA_CODEC_PDATA_H -#include <linux/kernel.h> +#include <linux/types.h> #define MADERA_MAX_INPUT 6 #define MADERA_MAX_MUXED_CHANNELS 4 -- 2.34.1

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[PATCH] ALSA: hda: intel-dsp-config: add JasperLake support
by Brent Lu 13 Jan '22

13 Jan '22

Add rules to select SOF driver for Jasper Lake systems if digital microphone is present or the system is a Chromebook. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/hda/intel-dsp-config.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index b9ac9e9e45a4..be47a1ca12b6 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -299,6 +299,27 @@ static const struct config_entry config_table[] = { }, #endif +/* Jasper Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_JASPERLAKE) + { + .flags = FLAG_SOF, + .device = 0x4dc8, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4dc8, + }, +#endif + /* Tigerlake */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) { -- 2.25.1

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