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

[PATCH 00/19] ASoC: SOF: Intel/IPC4: Support for external firmware libraries
by Peter Ujfalusi 19 Oct '22

19 Oct '22

Hi, In IPC4 all DSP loadable executable is a 'library' containing modules. The main or basefw is also a library which contains multiple modules. IPC4 allows to use loadable libraries to extend the functionality of the booted basefw. This series adds support for loading external libraries in case they are needed by the loaded topology file. The libraries must be placed to a specific firmware directory (fw_lib_prefix), which is: intel/avs-lib|sof-ipc4-lib/ followed by the platform name and in case of community key use a 'community' directory. For example for upx-i11 (community key): intel/avs-lib/tgl/community is the default path. The name of the library should be the UUID of the module it contains since the library loading is going to look for the file as <module_UUID>.bin In case there is a need to bundle multiple modules into single library, symlinks can be used to point to the file: module_boundle.bin <UUID1>.bin -> module_boundle.bin <UUID2>.bin -> module_boundle.bin <UUID3>.bin -> module_boundle.bin But note that in this case all modules will be loaded to the DSP since only the whole library can be loaded, not individual modules. Regards, Peter --- Peter Ujfalusi (18): ASoC: SOF: Introduce container struct for SOF firmware ASoC: SOF: amd: Use the basefw firmware container directly ASoC: SOF: Intel: hda-loader: Use the basefw firmware container directly ASoC: SOF: Intel: hda-loader-skl: Use the basefw firmware container directly ASoC: SOF: Drop the firmware and fw_offset from snd_sof_pdata ASoC: SOF: ipc: ops: Add support for optional init and exit callbacks ASoC: SOF: ipc4-loader: Save the maximum number of libraries supported ASoC: SOF: ipc4: Convert the firmware handling (loader) to library convention ASoC: SOF: IPC4: Add helper for looking up module by UUID ASoC: SOF: Add path definition for external firmware libraries ASoC: SOF: Intel: Set the default firmware library path for IPC4 ASoC: SOF: ipc4: Define platform dependent library loading callback ASoC: SOF: Intel: hda: Add flag to indicate that the firmware is IMR booted ASoC: SOF: Intel: Add ipc4 library loading implementation ASoC: SOF: loader: Add support for IPC dependent post firmware boot ops ASoC: SOF: ipc4: Stop using the query_fw_configuration fw_loader ops ASoC: SOF: loader: Remove the query_fw_configuration ops ASoC: SOF: ipc4-loader: Support for loading external libraries Ranjani Sridharan (1): ASoC: SOF: loader: Set complete state before post_fw_run op include/sound/sof.h | 10 +- include/sound/sof/ipc4/header.h | 4 + sound/soc/sof/amd/acp-loader.c | 6 +- sound/soc/sof/intel/apl.c | 3 + sound/soc/sof/intel/cnl.c | 3 + sound/soc/sof/intel/hda-loader-skl.c | 7 +- sound/soc/sof/intel/hda-loader.c | 83 +++++++++- sound/soc/sof/intel/hda.h | 4 + sound/soc/sof/intel/icl.c | 3 + sound/soc/sof/intel/mtl.c | 3 + sound/soc/sof/intel/pci-apl.c | 6 + sound/soc/sof/intel/pci-cnl.c | 9 ++ sound/soc/sof/intel/pci-icl.c | 6 + sound/soc/sof/intel/pci-mtl.c | 3 + sound/soc/sof/intel/pci-tgl.c | 21 +++ sound/soc/sof/intel/tgl.c | 3 + sound/soc/sof/ipc.c | 6 + sound/soc/sof/ipc3-loader.c | 26 ++- sound/soc/sof/ipc4-loader.c | 233 ++++++++++++++++++++++++--- sound/soc/sof/ipc4-priv.h | 65 ++++++-- sound/soc/sof/ipc4-topology.c | 17 +- sound/soc/sof/ipc4.c | 41 +++++ sound/soc/sof/loader.c | 25 ++- sound/soc/sof/sof-pci-dev.c | 26 +++ sound/soc/sof/sof-priv.h | 27 +++- 25 files changed, 540 insertions(+), 100 deletions(-) -- 2.38.0

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[PATCH] ASOC: amd: acp: Add TDM slots setting support for ACP I2S controller
by Venkata Prasad Potturu 19 Oct '22

19 Oct '22

Modify set tdm slot function to set ACP I2S controller slots in tdm mode based on number of slots. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/acp-i2s.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index ac416572db0d..09b6511c0a26 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -51,7 +51,7 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas struct device *dev = dai->component->dev; struct acp_dev_data *adata = snd_soc_dai_get_drvdata(dai); struct acp_stream *stream; - int slot_len; + int slot_len, no_of_slots; switch (slot_width) { case SLOT_WIDTH_8: @@ -71,6 +71,20 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas return -EINVAL; } + switch (slots) { + case 1 ... 7: + no_of_slots = slots; + break; + case 8: + no_of_slots = 0; + break; + default: + dev_err(dev, "Unsupported slots %d\n", slots); + return -EINVAL; + } + + slots = no_of_slots; + spin_lock_irq(&adata->acp_lock); list_for_each_entry(stream, &adata->stream_list, list) { if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) -- 2.25.1

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[PATCH] ASoC: twl4030: make read-only array ramp_base static const
by Colin Ian King 19 Oct '22

19 Oct '22

Don't populate the read-only array ramp_base on the stack but instead make it static const. Also makes the object code a little smaller. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/twl4030.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e48768233e20..9c50ac356c89 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -700,8 +700,10 @@ static void headset_ramp(struct snd_soc_component *component, int ramp) struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); struct twl4030_board_params *board_params = twl4030->board_params; /* Base values for ramp delay calculation: 2^19 - 2^26 */ - unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304, - 8388608, 16777216, 33554432, 67108864}; + static const unsigned int ramp_base[] = { + 524288, 1048576, 2097152, 4194304, + 8388608, 16777216, 33554432, 67108864 + }; unsigned int delay; hs_gain = twl4030_read(component, TWL4030_REG_HS_GAIN_SET); -- 2.37.3

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[PATCH 0/3] ASoC: Intel/SOF: simplify S3 resume flows
by Pierre-Louis Bossart 19 Oct '22

19 Oct '22

All Intel drivers for cAVS platforms contain a sequence for S3 resume which doesn't seem justified nor necessary. Forensic Git investigation in internal repositories did not provide any rationale for the implementation, and tests show no impact when those sequences are removed. This sequence was identified as problematic during a large HDaudio cleanup where all programming sequences were revisited before extensions are added. Pierre-Louis Bossart (3): ASoC: Intel: Skylake: simplify S3 resume flows ASoC: Intel: avs: simplify S3 resume flows ASoC: SOF: Intel: hda-dsp: simplify S3 resume flows sound/soc/intel/avs/core.c | 11 ----------- sound/soc/intel/skylake/skl.c | 9 --------- sound/soc/sof/intel/hda-dsp.c | 14 -------------- 3 files changed, 34 deletions(-) -- 2.34.1

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[PATCH] ASoC: SOF: ipc3: Log the tx message before sending it
by Peter Ujfalusi 19 Oct '22

19 Oct '22

It makes more sense to log the message before it is sent to the DSP. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/ipc3.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index b28af3a48b70..1fef4dcc0936 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -329,6 +329,8 @@ static int ipc3_tx_msg_unlocked(struct snd_sof_ipc *ipc, struct snd_sof_dev *sdev = ipc->sdev; int ret; + ipc3_log_header(sdev->dev, "ipc tx", hdr->cmd); + ret = sof_ipc_send_msg(sdev, msg_data, msg_bytes, reply_bytes); if (ret) { @@ -338,8 +340,6 @@ static int ipc3_tx_msg_unlocked(struct snd_sof_ipc *ipc, return ret; } - ipc3_log_header(sdev->dev, "ipc tx", hdr->cmd); - /* now wait for completion */ return ipc3_wait_tx_done(ipc, reply_data); } -- 2.38.0

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[PATCH][next][V2] ASoC: codecs: jz4725b: Fix spelling mistake "Sourc" -> "Source", "Routee" -> "Route"
by Colin Ian King 19 Oct '22

19 Oct '22

There are two spelling mistakes in codec routing description. Fix it. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- V2: Fix "Routee" -> "Route" too --- sound/soc/codecs/jz4725b.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 685ba1d3a644..64b14b1c74b9 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -359,7 +359,7 @@ static const struct snd_soc_dapm_route jz4725b_codec_dapm_routes[] = { {"Mixer to ADC", NULL, "Mixer"}, {"ADC Source Capture Route", "Mixer", "Mixer to ADC"}, - {"ADC Sourc Capture Routee", "Line In", "Line In"}, + {"ADC Source Capture Route", "Line In", "Line In"}, {"ADC Source Capture Route", "Mic 1", "Mic 1"}, {"ADC Source Capture Route", "Mic 2", "Mic 2"}, {"ADC", NULL, "ADC Source Capture Route"}, -- 2.37.3

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[PATCH][next] ASoC: codecs: jz4725b: Fix spelling mistake "Sourc" -> "Source"
by Colin Ian King 19 Oct '22

19 Oct '22

There is a spelling mistake in codec routing description. Fix it. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/jz4725b.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 685ba1d3a644..64b14b1c74b9 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -359,7 +359,7 @@ static const struct snd_soc_dapm_route jz4725b_codec_dapm_routes[] = { {"Mixer to ADC", NULL, "Mixer"}, {"ADC Source Capture Route", "Mixer", "Mixer to ADC"}, - {"ADC Sourc Capture Routee", "Line In", "Line In"}, + {"ADC Source Capture Routee", "Line In", "Line In"}, {"ADC Source Capture Route", "Mic 1", "Mic 1"}, {"ADC Source Capture Route", "Mic 2", "Mic 2"}, {"ADC", NULL, "ADC Source Capture Route"}, -- 2.37.3

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[PATCH] ASoC: sun50i-dmic: avoid unused variable warning for sun50i_dmic_of_match
by Ban Tao 19 Oct '22

19 Oct '22

In configurations with CONFIG_OF=n, we get a harmless build warning: sound/soc/sunxi/sun50i-dmic.c:268:34: warning: unused variable 'sun50i_dmic_of_match' [-Wunused-const-variable] Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/sunxi/sun50i-dmic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index 86cff5a..62509cb 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -391,7 +391,7 @@ static const struct dev_pm_ops sun50i_dmic_pm = { static struct platform_driver sun50i_dmic_driver = { .driver = { .name = "sun50i-dmic", - .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .of_match_table = sun50i_dmic_of_match, .pm = &sun50i_dmic_pm, }, .probe = sun50i_dmic_probe, -- 2.7.4

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[PATCH v1] ASoC: simple-card: Fix up checks for HW param fixups
by Aidan MacDonald 19 Oct '22

19 Oct '22

The "convert-xxx" properties only have an effect for DPCM DAI links. A DAI link is only created as DPCM if the device tree requires it; part of this involves checking for the use of "convert-xxx" properties. When the convert-sample-format property was added, the checks got out of sync. A DAI link that specified only convert-sample-format but did not pass any of the other DPCM checks would not go into DPCM mode and the convert-sample-format property would be silently ignored. Fix this by adding a function to do the "convert-xxx" property checks, instead of open-coding it in simple-card and audio-graph-card. And add "convert-sample-format" to the check function so that DAI links using it will be initialized correctly. Fixes: 047a05366f4b ("ASoC: simple-card-utils: Fixup DAI sample format") Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> --- include/sound/simple_card_utils.h | 1 + sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/simple-card-utils.c | 15 +++++++++++++++ sound/soc/generic/simple-card.c | 3 +-- 4 files changed, 18 insertions(+), 3 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index a0b827f0c2f6..cfed741e0181 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -177,6 +177,7 @@ void asoc_simple_convert_fixup(struct asoc_simple_data *data, struct snd_pcm_hw_params *params); void asoc_simple_parse_convert(struct device_node *np, char *prefix, struct asoc_simple_data *data); +bool asoc_simple_convert_is_required(const struct asoc_simple_data *data); int asoc_simple_parse_routing(struct snd_soc_card *card, char *prefix); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index b327372f2e4a..7804625b131e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -417,7 +417,7 @@ static inline bool parse_as_dpcm_link(struct asoc_simple_priv *priv, * or has convert-xxx property */ if ((of_get_child_count(codec_port) > 1) || - (adata->convert_rate || adata->convert_channels)) + asoc_simple_convert_is_required(adata)) return true; return false; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index bef16833c487..7a708be24656 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -85,6 +85,21 @@ void asoc_simple_parse_convert(struct device_node *np, } EXPORT_SYMBOL_GPL(asoc_simple_parse_convert); +/** + * asoc_simple_convert_is_required() - Query if HW param conversion was requested + * @data: Link data. + * + * Returns true if any HW param conversion was requested for this DAI link with + * any "convert-xxx" properties. + */ +bool asoc_simple_convert_is_required(const struct asoc_simple_data *data) +{ + return data->convert_rate || + data->convert_channels || + data->convert_sample_format; +} +EXPORT_SYMBOL_GPL(asoc_simple_convert_is_required); + int asoc_simple_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 78419e18717d..675991cdee87 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -393,8 +393,7 @@ static int __simple_for_each_link(struct asoc_simple_priv *priv, * or has convert-xxx property */ if (dpcm_selectable && - (num > 2 || - adata.convert_rate || adata.convert_channels)) { + (num > 2 || asoc_simple_convert_is_required(&adata))) { /* * np * |1(CPU)|0(Codec) li->cpu -- 2.38.0

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sof-es8336.conf
by GitHub issues - edited 18 Oct '22

18 Oct '22

alsa-project/alsa-ucm-conf issue #231 was edited from junocomp: The only way for the `Headphone Mixer Volume` to be enabled at startup the following commands needs to be added. `sset Headphone 3` When adding this to `sof-essx8336.conf` and reload alsa I get the following error ``` sudo alsaucm reload ALSA lib parser.c:1194:(parse_sequence) error: sequence command 'sset' is ignored ALSA lib parser.c:1045:(parse_sequence) error: string type is expected for sequence command ALSA lib parser.c:2289:(parse_controls_boot) Unable to parse BootSequence ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -22 alsaucm: error failed to open sound card hw:0: Invalid argument ``` When running `amixer -c 0 sset Headphone 3` manually it works fine Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/231 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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