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

arecord leaves doubled audio at end after a SIGTERM
by GitHub issues - edited 30 Nov '22

30 Nov '22

alsa-project/alsa-utils issue #179 was edited from chconnor: Hello: I issue a command like this: `arecord -t wav -c 1 -f S16_LE -r44100 -D hw:2,0 test.wav ` It works fine, but then I hit ctrl-C to terminate. This works, but the file has "junk" at the end that repeats the last fraction of a second. E.g. in this demo file I ctrl-C when I say the word "one", and the recording is like "three, two, on-on": https://caseyconnor.org/pub/audio/arecord_bug.wav It's as if it is mistakenly writing a buffer an extra time? (In my total ignorance and total lack of qualifications, I traced the code to [here](https://github.com/alsa-project/alsa-lib/blob/d28e8cb29485cc93f741b01…, but I get lost and can't see if anything is awry...) Issue URL : https://github.com/alsa-project/alsa-utils/issues/179 Repository URL: https://github.com/alsa-project/alsa-utils

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arecord leaves doubled audio at end after a SIGTERM
by GitHub issues - opened 30 Nov '22

30 Nov '22

alsa-project/alsa-utils issue #179 was opened from chconnor: Hello: I issue a command like this: `arecord -t wav -c 1 -f S16_LE -r44100 -D hw:2,0 test.wav ` It works fine, but then I hit ctrl-C to terminate. This works, but the file has "junk" at the end that repeats the last fraction of a second. E.g. in this demo file I ctrl-C when I say the word "one", and the recording is like "three, two, on-on": http://caseyconnor.org/pub/audio/arecord_bug.wav It's as if it is mistakenly writing a buffer an extra time? (In my total ignorance and total lack of qualifications, I traced the code to [here](https://github.com/alsa-project/alsa-lib/blob/d28e8cb29485cc93f741b01…, but I get lost and can't see if anything is awry...) Issue URL : https://github.com/alsa-project/alsa-utils/issues/179 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v2 0/2] ASoC: Intel: avs: rt5682: Refactor jack handling
by Cezary Rojewski 29 Nov '22

29 Nov '22

Leftover from recent series [1]. Following changes are proposed for the rt5682 sound card driver: 1) Move jack unassignment from platform_device->remove() to dai_link->exit(). This is done to make jack init and deinit flows symmetric 2) Remove platform_device->remove() function 3) Simplify card->suspend_pre() and card->resume_post() by making use of snd_soc_card_get_codec_dai() helper Changes in v2: - fixed compilation of the first patch in the series by moving offending pieces to the second patch [1]: https://lore.kernel.org/alsa-devel/20221125184032.2565979-1-cezary.rojewski… Amadeusz Sławiński (2): ASoC: Intel: avs: rt5682: Add define for codec DAI name ASoC: Intel: avs: rt5682: Refactor jack handling sound/soc/intel/avs/boards/rt5682.c | 31 ++++++++++++----------------- 1 file changed, 13 insertions(+), 18 deletions(-) -- 2.25.1

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[PATCH -next] ASoC: pcm512x: Fix PM disable depth imbalance in pcm512x_probe
by Zhang Qilong 29 Nov '22

29 Nov '22

The pm_runtime_enable will increase power disable depth. Thus a pairing decrement is needed on the error handling path to keep it balanced according to context. We fix it by going to err_pm instead of err_clk. Fixes:f086ba9d5389c ("ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/codecs/pcm512x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 767463e82665..89059a673cf0 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1634,7 +1634,7 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) if (val > 6) { dev_err(dev, "Invalid pll-in\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } pcm512x->pll_in = val; } @@ -1643,7 +1643,7 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) if (val > 6) { dev_err(dev, "Invalid pll-out\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } pcm512x->pll_out = val; } @@ -1652,12 +1652,12 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) dev_err(dev, "Error: both pll-in and pll-out, or none\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } if (pcm512x->pll_in && pcm512x->pll_in == pcm512x->pll_out) { dev_err(dev, "Error: pll-in == pll-out\n"); ret = -EINVAL; - goto err_clk; + goto err_pm; } } #endif -- 2.25.1

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[PATCH V3 0/6] ASoC: codecs: Add Awinic AW883XX audio amplifier driver
by wangweidong.a＠awinic.com 29 Nov '22

29 Nov '22

From: Weidong Wang <wangweidong.a(a)awinic.com> The Awinic AW883XX is an I2S/TDM input, high efficiency digital Smart K audio amplifier with an integrated 10.25V smart boost convert Add a DT schema for describing Awinic AW883xx audio amplifiers. They are controlled using I2C. v2 -> v3 : i2c reads and writes are changed to regmap The monitor, calib, and spin files were deleted Deleted firmware load retry Linux version compatibility and driver version control were removed The global variable was deleted The renaming of dai was removed Platform compatibility was removed Simplified log information use module_i2c_driver Delete the custom interface and use the kernel standard interface Weidong Wang (6): ASoC: codecs: Add i2c and codec registration for aw883xx and their associated operation functions ASoC: codecs: Added configuration file parsing for aw883xx ASoC: codecs: Add aw883xx chip control logic, such as power-on and power-off ASoC: codecs: Realize aw883xx register configuration and register address file ASoC: dt-bindings: Add schema for "awinic,aw883xx" ASoC:codecs:aw883xx corresponds to the modified Makefile and Kconfig .../devicetree/bindings/sound/awinic,aw883xx.yaml | 60 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/aw883xx/aw883xx.c | 1806 +++++++++++++++ sound/soc/codecs/aw883xx/aw883xx.h | 155 ++ sound/soc/codecs/aw883xx/aw883xx_bin_parse.c | 1294 +++++++++++ sound/soc/codecs/aw883xx/aw883xx_bin_parse.h | 145 ++ sound/soc/codecs/aw883xx/aw883xx_data_type.h | 148 ++ sound/soc/codecs/aw883xx/aw883xx_device.c | 1618 +++++++++++++ sound/soc/codecs/aw883xx/aw883xx_device.h | 544 +++++ sound/soc/codecs/aw883xx/aw883xx_init.c | 549 +++++ sound/soc/codecs/aw883xx/aw883xx_pid_2049_reg.h | 2387 ++++++++++++++++++++ 12 files changed, 8723 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/awinic,aw883xx.yaml create mode 100644 sound/soc/codecs/aw883xx/aw883xx.c create mode 100644 sound/soc/codecs/aw883xx/aw883xx.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_bin_parse.c create mode 100644 sound/soc/codecs/aw883xx/aw883xx_bin_parse.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_data_type.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_device.c create mode 100644 sound/soc/codecs/aw883xx/aw883xx_device.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_init.c create mode 100644 sound/soc/codecs/aw883xx/aw883xx_pid_2049_reg.h -- 2.7.4

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[PATCH 0/2] ASoC: Add a driver for the Iron Device SMA1303 Amp
by Kiseok Jo 29 Nov '22

29 Nov '22

This adds basic audio support for the Iron Device SMA1303 amplifier. Kiseok Jo (2): ASoC: sma1303: Add driver for Iron Device SMA1303 Amp ASoC: dt-bindings: sma1303: Iron Device SMA1303 Amp .../bindings/sound/irondevice,sma1303.yaml | 40 + .../devicetree/bindings/vendor-prefixes.yaml | 2 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1303.c | 2119 +++++++++++++++++ sound/soc/codecs/sma1303.h | 609 +++++ 6 files changed, 2779 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml create mode 100644 sound/soc/codecs/sma1303.c create mode 100644 sound/soc/codecs/sma1303.h -- 2.30.2

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[PATCH 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 29 Nov '22

29 Nov '22

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Jiaxin Yu (3): drm: bridge: it6505: bridge to hdmi-codec ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drivers/gpu/drm/bridge/ite-it6505.c | 36 +++++++++++++++--- include/sound/hdmi-codec.h | 6 +++ sound/soc/codecs/hdmi-codec.c | 37 +++++++++++++++++-- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 71 insertions(+), 12 deletions(-) -- 2.18.0

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[PATCH 00/11] ASoC: Intel: avs: Refactor jack handling
by Cezary Rojewski 29 Nov '22

29 Nov '22

For all the boards included in this patchset, a similar set of changes is proposed: 1) Move jack unassignment from platform_device->remove() to dai_link->exit(). This is done to make jack init and deinit flows symmetric 2) Remove platform_device->remove() function 3) Simplify card->suspend_pre() and card->resume_post() by making use of snd_soc_card_get_codec_dai() helper While bdw_rt286 board - which is utilized by the catpt-driver - is definitely not part of "avs", same treatment applies. And thus decided to make it part of this series instead of sending it separately. Amadeusz Sławiński (10): ASoC: Intel: avs: da7219: Refactor jack handling ASoC: Intel: avs: nau8825: Refactor jack handling ASoC: Intel: avs: rt274: Refer to DAI name through a constant ASoC: Intel: avs: rt274: Refactor jack handling ASoC: Intel: avs: rt286: Add define for codec DAI name ASoC: Intel: avs: rt286: Refactor jack handling ASoC: Intel: avs: rt298: Add define for codec DAI name ASoC: Intel: avs: rt298: Refactor jack handling ASoC: Intel: avs: rt5682: Add define for codec DAI name ASoC: Intel: avs: rt5682: Refactor jack handling Cezary Rojewski (1): ASoC: Intel: bdw_rt286: Refactor jack handling sound/soc/intel/avs/boards/da7219.c | 34 +++++++---------- sound/soc/intel/avs/boards/nau8825.c | 35 ++++++------------ sound/soc/intel/avs/boards/rt274.c | 30 ++++++--------- sound/soc/intel/avs/boards/rt286.c | 37 ++++++++----------- sound/soc/intel/avs/boards/rt298.c | 37 ++++++++----------- sound/soc/intel/avs/boards/rt5682.c | 31 +++++++--------- sound/soc/intel/boards/bdw_rt286.c | 55 ++++++++-------------------- 7 files changed, 95 insertions(+), 164 deletions(-) -- 2.25.1

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[PATCH v2] ASoC: qcom: lpass-sc7180: Add maybe_unused tag for system PM ops
by Srinivasa Rao Mandadapu 29 Nov '22

29 Nov '22

Add __maybe_unused tag for system PM ops suspend and resume. This is required to fix allmodconfig compilation issue. Fixes: 2d68148f8f85 ("ASoC: qcom: lpass-sc7180: Add system suspend/resume PM ops") Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- Changes since v1: -- Updated Fixes tag. sound/soc/qcom/lpass-sc7180.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index b96b85a..41db661 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -163,14 +163,14 @@ static int sc7180_lpass_exit(struct platform_device *pdev) return 0; } -static int sc7180_lpass_dev_resume(struct device *dev) +static int __maybe_unused sc7180_lpass_dev_resume(struct device *dev) { struct lpass_data *drvdata = dev_get_drvdata(dev); return clk_bulk_prepare_enable(drvdata->num_clks, drvdata->clks); } -static int sc7180_lpass_dev_suspend(struct device *dev) +static int __maybe_unused sc7180_lpass_dev_suspend(struct device *dev) { struct lpass_data *drvdata = dev_get_drvdata(dev); -- 2.7.4

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[PATCH v2 1/1] ASoC: SOF: Add DAI configuration support for AMD platforms.
by V sujith kumar Reddy 29 Nov '22

29 Nov '22

From: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> Add support for configuring sp and hs DAI from topology. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> Changes since v1 -- Apply on latest broonie-git for-next --- include/sound/sof/dai-amd.h | 1 + include/sound/sof/dai.h | 2 ++ include/uapi/sound/sof/tokens.h | 5 +++++ sound/soc/sof/ipc3-pcm.c | 2 ++ sound/soc/sof/ipc3-topology.c | 36 +++++++++++++++++++++++++-------- sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 10 +++++++++ 7 files changed, 49 insertions(+), 8 deletions(-) diff --git a/include/sound/sof/dai-amd.h b/include/sound/sof/dai-amd.h index 92f45c180b7c..9df7ac824efe 100644 --- a/include/sound/sof/dai-amd.h +++ b/include/sound/sof/dai-amd.h @@ -17,6 +17,7 @@ struct sof_ipc_dai_acp_params { uint32_t fsync_rate; /* FSYNC frequency in Hz */ uint32_t tdm_slots; + uint32_t tdm_mode; } __packed; /* ACPDMIC Configuration Request - SOF_IPC_DAI_AMD_CONFIG */ diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index 9fbd3832bcdc..3041f5805b7b 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -86,6 +86,8 @@ enum sof_ipc_dai_type { SOF_DAI_AMD_DMIC, /**< AMD ACP DMIC */ SOF_DAI_MEDIATEK_AFE, /**< Mediatek AFE */ SOF_DAI_AMD_HS, /**< Amd HS */ + SOF_DAI_AMD_SP_VIRTUAL, /**< AMD ACP SP VIRTUAL */ + SOF_DAI_AMD_HS_VIRTUAL, /**< AMD ACP HS VIRTUAL */ }; /* general purpose DAI configuration */ diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index f187dfbd9325..bacaf8a6317e 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -198,4 +198,9 @@ /* COPIER */ #define SOF_TKN_INTEL_COPIER_NODE_TYPE 1980 +/* ACP I2S */ +#define SOF_TKN_AMD_ACPI2S_RATE 1700 +#define SOF_TKN_AMD_ACPI2S_CH 1701 +#define SOF_TKN_AMD_ACPI2S_TDM_MODE 1702 + #endif diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c index dad57bef38f6..f10bfc9bd5cb 100644 --- a/sound/soc/sof/ipc3-pcm.c +++ b/sound/soc/sof/ipc3-pcm.c @@ -336,6 +336,7 @@ static int sof_ipc3_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, channels->min, channels->max); break; case SOF_DAI_AMD_SP: + case SOF_DAI_AMD_SP_VIRTUAL: rate->min = private->dai_config->acpsp.fsync_rate; rate->max = private->dai_config->acpsp.fsync_rate; channels->min = private->dai_config->acpsp.tdm_slots; @@ -347,6 +348,7 @@ static int sof_ipc3_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, channels->min, channels->max); break; case SOF_DAI_AMD_HS: + case SOF_DAI_AMD_HS_VIRTUAL: rate->min = private->dai_config->acphs.fsync_rate; rate->max = private->dai_config->acphs.fsync_rate; channels->min = private->dai_config->acphs.tdm_slots; diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 0720e1eae084..b94cc40485ed 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -276,6 +276,16 @@ static const struct sof_topology_token acpdmic_tokens[] = { offsetof(struct sof_ipc_dai_acpdmic_params, pdm_ch)}, }; +/* ACPI2S */ +static const struct sof_topology_token acpi2s_tokens[] = { + {SOF_TKN_AMD_ACPI2S_RATE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acp_params, fsync_rate)}, + {SOF_TKN_AMD_ACPI2S_CH, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acp_params, tdm_slots)}, + {SOF_TKN_AMD_ACPI2S_TDM_MODE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acp_params, tdm_mode)}, +}; + /* Core tokens */ static const struct sof_topology_token core_tokens[] = { {SOF_TKN_COMP_CORE_ID, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, @@ -311,6 +321,7 @@ static const struct sof_token_info ipc3_token_list[SOF_TOKEN_COUNT] = { [SOF_SAI_TOKENS] = {"SAI tokens", sai_tokens, ARRAY_SIZE(sai_tokens)}, [SOF_AFE_TOKENS] = {"AFE tokens", afe_tokens, ARRAY_SIZE(afe_tokens)}, [SOF_ACPDMIC_TOKENS] = {"ACPDMIC tokens", acpdmic_tokens, ARRAY_SIZE(acpdmic_tokens)}, + [SOF_ACPI2S_TOKENS] = {"ACPI2S tokens", acpi2s_tokens, ARRAY_SIZE(acpi2s_tokens)}, }; /** @@ -1193,6 +1204,7 @@ static int sof_link_acp_sp_load(struct snd_soc_component *scomp, struct snd_sof_ struct snd_soc_tplg_hw_config *hw_config = slink->hw_configs; struct sof_dai_private_data *private = dai->private; u32 size = sizeof(*config); + int ret; /* handle master/slave and inverted clocks */ sof_dai_set_format(hw_config, config); @@ -1201,12 +1213,15 @@ static int sof_link_acp_sp_load(struct snd_soc_component *scomp, struct snd_sof_ memset(&config->acpsp, 0, sizeof(config->acpsp)); config->hdr.size = size; - config->acpsp.fsync_rate = le32_to_cpu(hw_config->fsync_rate); - config->acpsp.tdm_slots = le32_to_cpu(hw_config->tdm_slots); + ret = sof_update_ipc_object(scomp, &config->acpsp, SOF_ACPI2S_TOKENS, slink->tuples, + slink->num_tuples, size, slink->num_hw_configs); + if (ret < 0) + return ret; - dev_info(scomp->dev, "ACP_SP config ACP%d channel %d rate %d\n", + + dev_info(scomp->dev, "ACP_SP config ACP%d channel %d rate %d tdm_mode %d\n", config->dai_index, config->acpsp.tdm_slots, - config->acpsp.fsync_rate); + config->acpsp.fsync_rate, config->acpsp.tdm_mode); dai->number_configs = 1; dai->current_config = 0; @@ -1223,6 +1238,7 @@ static int sof_link_acp_hs_load(struct snd_soc_component *scomp, struct snd_sof_ struct snd_soc_tplg_hw_config *hw_config = slink->hw_configs; struct sof_dai_private_data *private = dai->private; u32 size = sizeof(*config); + int ret; /* Configures the DAI hardware format and inverted clocks */ sof_dai_set_format(hw_config, config); @@ -1231,12 +1247,14 @@ static int sof_link_acp_hs_load(struct snd_soc_component *scomp, struct snd_sof_ memset(&config->acphs, 0, sizeof(config->acphs)); config->hdr.size = size; - config->acphs.fsync_rate = le32_to_cpu(hw_config->fsync_rate); - config->acphs.tdm_slots = le32_to_cpu(hw_config->tdm_slots); + ret = sof_update_ipc_object(scomp, &config->acphs, SOF_ACPI2S_TOKENS, slink->tuples, + slink->num_tuples, size, slink->num_hw_configs); + if (ret < 0) + return ret; - dev_info(scomp->dev, "ACP_HS config ACP%d channel %d rate %d\n", + dev_info(scomp->dev, "ACP_HS config ACP%d channel %d rate %d tdm_mode %d\n", config->dai_index, config->acphs.tdm_slots, - config->acphs.fsync_rate); + config->acphs.fsync_rate, config->acphs.tdm_mode); dai->number_configs = 1; dai->current_config = 0; @@ -1545,9 +1563,11 @@ static int sof_ipc3_widget_setup_comp_dai(struct snd_sof_widget *swidget) ret = sof_link_acp_bt_load(scomp, slink, config, dai); break; case SOF_DAI_AMD_SP: + case SOF_DAI_AMD_SP_VIRTUAL: ret = sof_link_acp_sp_load(scomp, slink, config, dai); break; case SOF_DAI_AMD_HS: + case SOF_DAI_AMD_HS_VIRTUAL: ret = sof_link_acp_hs_load(scomp, slink, config, dai); break; case SOF_DAI_AMD_DMIC: diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 1b5b3ea53a6e..29cf951e3526 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -248,6 +248,7 @@ enum sof_tokens { SOF_COPIER_FORMAT_TOKENS, SOF_GAIN_TOKENS, SOF_ACPDMIC_TOKENS, + SOF_ACPI2S_TOKENS, /* this should be the last */ SOF_TOKEN_COUNT, diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 9d9fcaa2a948..c668bd9d21ec 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -289,6 +289,9 @@ static const struct sof_dai_types sof_dais[] = { {"ACPDMIC", SOF_DAI_AMD_DMIC}, {"ACPHS", SOF_DAI_AMD_HS}, {"AFE", SOF_DAI_MEDIATEK_AFE}, + {"ACPSP_VIRTUAL", SOF_DAI_AMD_SP_VIRTUAL}, + {"ACPHS_VIRTUAL", SOF_DAI_AMD_HS_VIRTUAL}, + }; static enum sof_ipc_dai_type find_dai(const char *name) @@ -1895,6 +1898,13 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_ token_id = SOF_ACPDMIC_TOKENS; num_tuples += token_list[SOF_ACPDMIC_TOKENS].count; break; + case SOF_DAI_AMD_SP: + case SOF_DAI_AMD_HS: + case SOF_DAI_AMD_SP_VIRTUAL: + case SOF_DAI_AMD_HS_VIRTUAL: + token_id = SOF_ACPI2S_TOKENS; + num_tuples += token_list[SOF_ACPI2S_TOKENS].count; + break; default: break; } -- 2.25.1

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