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

[PATCH] ASoC: adau17x1: Remove redundant null check before clk_disable_unprepare
by Xu Wang 12 Jan '21

12 Jan '21

Because clk_disable_unprepare() already checked NULL clock parameter, so the additional check is unnecessary, just remove it. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/soc/codecs/adau17x1.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 30e072c80ac1..546ee8178038 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -1095,8 +1095,7 @@ void adau17x1_remove(struct device *dev) { struct adau *adau = dev_get_drvdata(dev); - if (adau->mclk) - clk_disable_unprepare(adau->mclk); + clk_disable_unprepare(adau->mclk); } EXPORT_SYMBOL_GPL(adau17x1_remove); -- 2.17.1

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[PATCH] ASoC: Intel: cht_bsw_nau8824: Move snd_soc_dai_set_tdm_slot call to cht_codec_fixup
by Hans de Goede 12 Jan '21

12 Jan '21

Move the snd_soc_dai_set_tdm_slot() call from cht_codec_init() to cht_codec_fixup(). There are 2 reasons for doing this: 1. This aligns the cht_bsw_nau8824 with all the other BYT/CHT machine drivers which also do this from their codec_fixup function. 2. When using the SOF driver, things like the TDM info is set from the topology file. Moving the call to the codec_fixup function, which gets skipped when using the SOF driver avoids the call interfering with the settings when using the SOF driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_nau8824.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index fd5e25ca05f7..da5a5cbc8759 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -100,13 +100,6 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_component *component = codec_dai->component; int ret, jack_type; - /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0x1, 4, 24); - if (ret < 0) { - dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); - return ret; - } - /* NAU88L24 supports 4 butons headset detection * KEY_PLAYPAUSE * KEY_VOICECOMMAND @@ -141,6 +134,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + int ret; /* The DSP will covert the FE rate to 48k, stereo, 24bits */ rate->min = rate->max = 48000; @@ -150,6 +144,13 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, snd_mask_none(fmt); params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xf, 0x1, 4, 24); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + return 0; } -- 2.28.0

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[PATCH] ASoC: SOF: add mutex to protect the dsp_power_state access
by Kai Vehmanen 12 Jan '21

12 Jan '21

From: Keyon Jie <yang.jie(a)linux.intel.com> There could be more than one thread read/write the dsp_power_state simultaneously (e.g. hda_dsp_d0i3_work and sof_ipc_tx_message), add a mutex power_state_access to make sure the access to it is mutually exclusive. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/core.c | 1 + sound/soc/sof/ops.h | 11 ++++++++--- sound/soc/sof/sof-priv.h | 2 ++ 3 files changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index adc7c37145d6..2b85ef5d6092 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -316,6 +316,7 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) INIT_LIST_HEAD(&sdev->route_list); spin_lock_init(&sdev->ipc_lock); spin_lock_init(&sdev->hw_lock); + mutex_init(&sdev->power_state_access); if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) INIT_WORK(&sdev->probe_work, sof_probe_work); diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 95e748b36903..4c1f9daaa6e8 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -208,11 +208,16 @@ static inline int snd_sof_dsp_set_power_state(struct snd_sof_dev *sdev, const struct sof_dsp_power_state *target_state) { + int ret = 0; + + mutex_lock(&sdev->power_state_access); + if (sof_ops(sdev)->set_power_state) - return sof_ops(sdev)->set_power_state(sdev, target_state); + ret = sof_ops(sdev)->set_power_state(sdev, target_state); - /* D0 substate is not supported, do nothing here. */ - return 0; + mutex_unlock(&sdev->power_state_access); + + return ret; } /* debug */ diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 68da8f797403..28d19fa30614 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -375,6 +375,8 @@ struct snd_sof_dev { /* current DSP power state */ struct sof_dsp_power_state dsp_power_state; + /* mutex to protect the dsp_power_state access */ + struct mutex power_state_access; /* Intended power target of system suspend */ enum sof_system_suspend_state system_suspend_target; -- 2.29.2

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[PATCH] ALSA: hda/hdmi - enable runtime pm for CI AMD display audio
by Alex Deucher 12 Jan '21

12 Jan '21

We are able to power down the GPU and audio via the GPU driver so flag these asics as supporting runtime pm. Reviewed-by: Evan Quan <evan.quan(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/pci/hda/hda_intel.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d539f52009a1..2ea46b5a9a23 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2597,7 +2597,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0x1308), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x157a), @@ -2659,9 +2660,11 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xaab0), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaac0), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0xaac8), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0xaad8), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, -- 2.29.2

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[PATCH/RFC 0/2] ALSA: firewire: Fix integer overflows on 32-bit
by Geert Uytterhoeven 12 Jan '21

12 Jan '21

Hi all, This patch series fixes two integer overflows on 32-bit platforms when in multiplications with the NSEC_PER_SEC constant, found by code inspection. They are marked "RFC", as I don't know the maximum transfer length of MIDI. Depending on this maximum length, a small tweak may be necessary. Thanks for your comments! Geert Uytterhoeven (2): ALSA: fireface: Fix integer overflow in transmit_midi_msg() ALSA: firewire-tascam: Fix integer overflow in midi_port_work() sound/firewire/fireface/ff-transaction.c | 2 +- sound/firewire/tascam/tascam-transaction.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 2.25.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[PATCH v2 0/2] fix tegra-hda on tegra30 devices
by Peter Geis 12 Jan '21

12 Jan '21

The following patches fix tegra-hda on legacy tegra devices. Two issues were discovered preventing tegra-hda from functioning: -The hda clocks on tegra30 were assigned to clk_m and running at too low of a rate to function. -The tegra-hda encounters an input/output error when opening a stream. Since the only mainline device that used tegra-hda until recently was the t124, it is unknown exactly when this was broken. Fortunately a recent patch was submitted that fixed the issue only on t194 devices. We can apply it to the tegra30-hda device to resolve the issue across the board. Note that downstream devices used the spdif device instead of hda for hdmi audio. The spdif device lacks a driver on mainline. -Checkpatch seems to have issues finding [1], but git show has no issue. [1] commit 60019d8c650d ("ALSA: hda/tegra: workaround playback failure on Tegra194") Changelog: v2: -Added ack and reviewed-by from Jon -Updated fix to apply to tegra30-hda vice universally (Thanks Jon) -Updated commit to include comments from hardware team (Thanks Sameer) -Cleaned up commit messages Peter Geis (2): clk: tegra30: Add hda clock default rates to clock driver ALSA: hda/tegra: fix tegra-hda on tegra30 soc drivers/clk/tegra/clk-tegra30.c | 2 ++ sound/pci/hda/hda_tegra.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) -- 2.25.1

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SoF support for 8086:9d71?
by Kai-Heng Feng 12 Jan '21

12 Jan '21

Hi, On more recent kernels legacy HDA stops working for device 8086:9d71: [ 47.211381] snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 [ 47.211383] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SST driver However, only legacy HDA supports it: 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Sunrise Point-LP HD Audio [8086:9d71] (rev 21) .. Kernel modules: snd_hda_intel Ubuntu kernel doesn't select SND_SOC_INTEL_SKYLAKE, which builds snd-soc-skl, because based on the Kconfig description SOF should support it. I wonder if SND_SOC_INTEL_SKYLAKE should be select again? BugLink: <https://bugs.launchpad.net/bugs/1902457>

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alsa-lib: snd_pcm_plugin_status() recently broken
by Mark Hills 12 Jan '21

12 Jan '21

The following assertion: relay: pcm_plugin.c:570: snd_pcm_plugin_status: Assertion `status->appl_ptr == *pcm->appl.ptr' failed. It bisects to 28cc099, the introduction of that assertion. #4 0x00007ffff7d0dd03 in snd_pcm_plugin_status (pcm=0x633560, status=0x7fffffffe1e0) at pcm_plugin.c:570 570 assert(status->appl_ptr == *pcm->appl.ptr); (gdb) print status->appl_ptr $1 = 0 (gdb) print pcm->appl.ptr $2 = (volatile snd_pcm_uframes_t *) 0x6334f8 (gdb) print *pcm->appl.ptr $3 = 47936 The calling code is single threaded on a dmix device, with these operations in order: snd_pcm_avail_update(); snd_pcm_status(); snd_pcm_mmap_begin(); Commenting out the assertion leaves me with a credible result from the function as it did previously; snd_pcm_status_get_delay() == 64. -- Mark >From 28cc099d9ea3962b033cb1cb9c3e07db828d9ff7 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela <perex(a)perex.cz> Date: Sun, 3 Jan 2021 16:41:29 +0100 Subject: [PATCH] pcm: plugin - optimize sync in snd_pcm_plugin_status() Do hw_ptr sync only once after the status call. This avoids double update. Also, the application pointer must not change when the status is called. Add assert() call for this condition. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- src/pcm/pcm_plugin.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index 83793397..6a815145 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -553,14 +553,11 @@ static int snd_pcm_plugin_status(snd_pcm_t *pcm, snd_pcm_status_t * status) snd_pcm_plugin_t *plugin = pcm->private_data; snd_pcm_sframes_t err; - /* sync with the latest hw and appl ptrs */ - snd_pcm_plugin_avail_update(pcm); - err = snd_pcm_status(plugin->gen.slave, status); if (err < 0) return err; - status->appl_ptr = *pcm->appl.ptr; - status->hw_ptr = *pcm->hw.ptr; + assert(status->appl_ptr == *pcm->appl.ptr); + snd_pcm_plugin_sync_hw_ptr(pcm, status->hw_ptr, status->avail); return 0; } -- 2.17.5

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[PATCH][next] ASoC: soc-pcm: Fix uninitialised return value in variable ret
by Colin King 12 Jan '21

12 Jan '21

From: Colin Ian King <colin.king(a)canonical.com> Currently when attempting to start the BE fails because the FE is not started the error return variable ret is not initialized and garbage is returned. Fix this by setting it to 0 so the caller does not report the error "ASoC: failed to shutdown some BEs" and because this failure path has already reported the reason for the early return. Addresses-Coverity: ("Uninitialized scalar variable") Fixes: 2c1382840c19 ("ASoC: soc-pcm: disconnect BEs if the FE is not ready") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/soc-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 481a4a25acb0..b787ce4ceb5a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2443,6 +2443,7 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) { dev_err(fe->dev, "ASoC: FE %s is not ready %d\n", fe->dai_link->name, fe->dpcm[stream].state); + ret = 0; goto disconnect; } -- 2.29.2

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[PATCH] docs: ALSA: Fix reference to mixart.rst
by Jonathan Neuschäfer 12 Jan '21

12 Jan '21

MIXART.txt has been converted to ReST and renamed. Fix the reference in alsa-configuration.rst. Fixes: 3d8e81862ce4 ("ALSA: doc: ReSTize MIXART.txt") Signed-off-by: Jonathan Neuschäfer <j.neuschaefer(a)gmx.net> --- Documentation/sound/alsa-configuration.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index fe52c314b7639..b36af65a08edf 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -1501,7 +1501,7 @@ Module for Digigram miXart8 sound cards. This module supports multiple cards. Note: One miXart8 board will be represented as 4 alsa cards. -See MIXART.txt for details. +See Documentation/sound/cards/mixart.rst for details. When the driver is compiled as a module and the hotplug firmware is supported, the firmware data is loaded via hotplug automatically. -- 2.29.2

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