From e4746d94d00c52918461bc169e009b6784a38e21 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 16 Aug 2022 13:17:22 +0200 Subject: ASoC: Intel: Skylake: Introduce HDA codec init and exit routines Preliminary step in making snd_hda_codec_device_init() the only constructor for struct hda_codec instances. To do that, existing usage of hdac_ext equivalents has to be dropped. Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Signed-off-by: Cezary Rojewski Acked-by: Mark Brown Link: https://lore.kernel.org/r/20220816111727.3218543-2-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/intel/skylake/skl.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index aeca58246fc7..33b0ed6b0534 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -689,6 +689,35 @@ static void load_codec_module(struct hda_codec *codec) #endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ +static void skl_codec_device_exit(struct device *dev) +{ + snd_hdac_device_exit(dev_to_hdac_dev(dev)); +} + +static __maybe_unused struct hda_codec *skl_codec_device_init(struct hdac_bus *bus, int addr) +{ + struct hda_codec *codec; + int ret; + + codec = snd_hda_codec_device_init(to_hda_bus(bus), addr, "ehdaudio%dD%d", bus->idx, addr); + if (IS_ERR(codec)) { + dev_err(bus->dev, "device init failed for hdac device\n"); + return codec; + } + + codec->core.type = HDA_DEV_ASOC; + codec->core.dev.release = skl_codec_device_exit; + + ret = snd_hdac_device_register(&codec->core); + if (ret) { + dev_err(bus->dev, "failed to register hdac device\n"); + snd_hdac_device_exit(&codec->core); + return ERR_PTR(ret); + } + + return codec; +} + /* * Probe the given codec address */ -- cgit v1.2.3 From 829c67319806009abfe3b0b82b3b8b153a2c5e32 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 16 Aug 2022 13:17:23 +0200 Subject: ASoC: SOF: Intel: Introduce HDA codec init and exit routines Preliminary step in making snd_hda_codec_device_init() the only constructor for struct hda_codec instances. To do that, existing usage of hdac_ext equivalents has to be dropped. Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Signed-off-by: Cezary Rojewski Acked-by: Mark Brown Link: https://lore.kernel.org/r/20220816111727.3218543-3-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/sof/intel/hda-codec.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 2f3f4a733d9e..4c128ba02340 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -109,6 +109,36 @@ EXPORT_SYMBOL_NS(hda_codec_jack_check, SND_SOC_SOF_HDA_AUDIO_CODEC); #define is_generic_config(x) 0 #endif +static void hda_codec_device_exit(struct device *dev) +{ + snd_hdac_device_exit(dev_to_hdac_dev(dev)); +} + +static __maybe_unused struct hda_codec * +hda_codec_device_init(struct hdac_bus *bus, int addr, int type) +{ + struct hda_codec *codec; + int ret; + + codec = snd_hda_codec_device_init(to_hda_bus(bus), addr, "ehdaudio%dD%d", bus->idx, addr); + if (IS_ERR(codec)) { + dev_err(bus->dev, "device init failed for hdac device\n"); + return codec; + } + + codec->core.type = type; + codec->core.dev.release = hda_codec_device_exit; + + ret = snd_hdac_device_register(&codec->core); + if (ret) { + dev_err(bus->dev, "failed to register hdac device\n"); + snd_hdac_device_exit(&codec->core); + return ERR_PTR(ret); + } + + return codec; +} + /* probe individual codec */ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, bool hda_codec_use_common_hdmi) -- cgit v1.2.3 From 3fd63658caed9494cca1d4789a66d3d2def2a0ab Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 16 Aug 2022 13:17:24 +0200 Subject: ASoC: Intel: Drop hdac_ext usage for codec device creation To make snd_hda_codec_device_init() the only constructor for struct hda_codec instances remaining tasks are: 1) no struct may wrap struct hda_codec as its base type 2) bus drivers (skylake and sof) which are the current hdac_ext users need to be adjusted to make use of newly added codec init and exit routines instead 3) as bus drivers (skylake and sof) are to be responsible for creating codec device and assigning it to hdac_hda_priv->codec, hdac_hda_dev_probe() has to be freed of that job To keep git bisect happy, all of these in made in one-go. Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Signed-off-by: Cezary Rojewski Acked-by: Mark Brown Link: https://lore.kernel.org/r/20220816111727.3218543-4-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/codecs/hdac_hda.c | 26 ++++++++++--------------- sound/soc/codecs/hdac_hda.h | 2 +- sound/soc/intel/boards/hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/skylake/skl.c | 26 +++++++++++-------------- sound/soc/sof/intel/hda-codec.c | 29 ++++++++++++---------------- 6 files changed, 36 insertions(+), 51 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 8debcee59224..77df4c5b274a 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -246,7 +246,7 @@ static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, return -EINVAL; hda_stream = &pcm->stream[substream->stream]; - snd_hda_codec_cleanup(&hda_pvt->codec, hda_stream, substream); + snd_hda_codec_cleanup(hda_pvt->codec, hda_stream, substream); return 0; } @@ -264,7 +264,7 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, int ret = 0; hda_pvt = snd_soc_component_get_drvdata(component); - hdev = &hda_pvt->codec.core; + hdev = &hda_pvt->codec->core; pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); if (!pcm) return -EINVAL; @@ -274,7 +274,7 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream]; format_val = hda_pvt->pcm[dai->id].format_val[substream->stream]; - ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream, + ret = snd_hda_codec_prepare(hda_pvt->codec, hda_stream, stream, format_val, substream); if (ret < 0) dev_err(&hdev->dev, "codec prepare failed %d\n", ret); @@ -299,7 +299,7 @@ static int hdac_hda_dai_open(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - return hda_stream->ops.open(hda_stream, &hda_pvt->codec, substream); + return hda_stream->ops.open(hda_stream, hda_pvt->codec, substream); } static void hdac_hda_dai_close(struct snd_pcm_substream *substream, @@ -317,7 +317,7 @@ static void hdac_hda_dai_close(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - hda_stream->ops.close(hda_stream, &hda_pvt->codec, substream); + hda_stream->ops.close(hda_stream, hda_pvt->codec, substream); snd_hda_codec_pcm_put(pcm); } @@ -325,7 +325,7 @@ static void hdac_hda_dai_close(struct snd_pcm_substream *substream, static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, struct snd_soc_dai *dai) { - struct hda_codec *hcodec = &hda_pvt->codec; + struct hda_codec *hcodec = hda_pvt->codec; struct hda_pcm *cpcm; const char *pcm_name; @@ -394,8 +394,8 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - struct hdac_device *hdev = &hda_pvt->codec.core; - struct hda_codec *hcodec = &hda_pvt->codec; + struct hdac_device *hdev = &hda_pvt->codec->core; + struct hda_codec *hcodec = hda_pvt->codec; struct hdac_ext_link *hlink; hda_codec_patch_t patch; int ret; @@ -515,8 +515,8 @@ static void hdac_hda_codec_remove(struct snd_soc_component *component) { struct hdac_hda_priv *hda_pvt = snd_soc_component_get_drvdata(component); - struct hdac_device *hdev = &hda_pvt->codec.core; - struct hda_codec *codec = &hda_pvt->codec; + struct hdac_device *hdev = &hda_pvt->codec->core; + struct hda_codec *codec = hda_pvt->codec; struct hdac_ext_link *hlink = NULL; hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev)); @@ -584,7 +584,6 @@ static const struct snd_soc_component_driver hdac_hda_codec = { static int hdac_hda_dev_probe(struct hdac_device *hdev) { struct hdac_ext_link *hlink; - struct hdac_hda_priv *hda_pvt; int ret; /* hold the ref while we probe */ @@ -595,10 +594,6 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) } snd_hdac_ext_bus_link_get(hdev->bus, hlink); - hda_pvt = hdac_to_hda_priv(hdev); - if (!hda_pvt) - return -ENOMEM; - /* ASoC specific initialization */ ret = devm_snd_soc_register_component(&hdev->dev, &hdac_hda_codec, hdac_hda_dais, @@ -608,7 +603,6 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) return ret; } - dev_set_drvdata(&hdev->dev, hda_pvt); snd_hdac_ext_bus_link_put(hdev->bus, hlink); return ret; diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h index d0efc5e254ae..fc19c34ca00e 100644 --- a/sound/soc/codecs/hdac_hda.h +++ b/sound/soc/codecs/hdac_hda.h @@ -23,7 +23,7 @@ struct hdac_hda_pcm { }; struct hdac_hda_priv { - struct hda_codec codec; + struct hda_codec *codec; struct hdac_hda_pcm pcm[HDAC_LAST_DAI_ID]; bool need_display_power; }; diff --git a/sound/soc/intel/boards/hda_dsp_common.c b/sound/soc/intel/boards/hda_dsp_common.c index 83c7dfbccd9d..04b7d4f7f9e2 100644 --- a/sound/soc/intel/boards/hda_dsp_common.c +++ b/sound/soc/intel/boards/hda_dsp_common.c @@ -54,7 +54,7 @@ int hda_dsp_hdmi_build_controls(struct snd_soc_card *card, return -EINVAL; hda_pvt = snd_soc_component_get_drvdata(comp); - hcodec = &hda_pvt->codec; + hcodec = hda_pvt->codec; list_for_each_entry(hpcm, &hcodec->pcm_list_head, list) { spcm = hda_dsp_hdmi_pcm_handle(card, i); diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 81144efb4b44..879ebba52832 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -190,7 +190,7 @@ static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) * all codecs are on the same bus, so it's sufficient * to look up only the first one */ - snd_hda_set_power_save(hda_pvt->codec.bus, + snd_hda_set_power_save(hda_pvt->codec->bus, HDA_CODEC_AUTOSUSPEND_DELAY_MS); break; } diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 33b0ed6b0534..c7c1cad2a753 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -694,7 +694,7 @@ static void skl_codec_device_exit(struct device *dev) snd_hdac_device_exit(dev_to_hdac_dev(dev)); } -static __maybe_unused struct hda_codec *skl_codec_device_init(struct hdac_bus *bus, int addr) +static struct hda_codec *skl_codec_device_init(struct hdac_bus *bus, int addr) { struct hda_codec *codec; int ret; @@ -729,9 +729,8 @@ static int probe_codec(struct hdac_bus *bus, int addr) struct skl_dev *skl = bus_to_skl(bus); #if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) struct hdac_hda_priv *hda_codec; - int err; #endif - struct hdac_device *hdev; + struct hda_codec *codec; mutex_lock(&bus->cmd_mutex); snd_hdac_bus_send_cmd(bus, cmd); @@ -747,25 +746,22 @@ static int probe_codec(struct hdac_bus *bus, int addr) if (!hda_codec) return -ENOMEM; - hda_codec->codec.bus = skl_to_hbus(skl); - hdev = &hda_codec->codec.core; + codec = skl_codec_device_init(bus, addr); + if (IS_ERR(codec)) + return PTR_ERR(codec); - err = snd_hdac_ext_bus_device_init(bus, addr, hdev, HDA_DEV_ASOC); - if (err < 0) - return err; + hda_codec->codec = codec; + dev_set_drvdata(&codec->core.dev, hda_codec); /* use legacy bus only for HDA codecs, idisp uses ext bus */ if ((res & 0xFFFF0000) != IDISP_INTEL_VENDOR_ID) { - hdev->type = HDA_DEV_LEGACY; - load_codec_module(&hda_codec->codec); + codec->core.type = HDA_DEV_LEGACY; + load_codec_module(hda_codec->codec); } return 0; #else - hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL); - if (!hdev) - return -ENOMEM; - - return snd_hdac_ext_bus_device_init(bus, addr, hdev, HDA_DEV_ASOC); + codec = skl_codec_device_init(bus, addr); + return PTR_ERR_OR_ZERO(codec); #endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ } diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 4c128ba02340..73336648cd25 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -114,8 +114,7 @@ static void hda_codec_device_exit(struct device *dev) snd_hdac_device_exit(dev_to_hdac_dev(dev)); } -static __maybe_unused struct hda_codec * -hda_codec_device_init(struct hdac_bus *bus, int addr, int type) +static struct hda_codec *hda_codec_device_init(struct hdac_bus *bus, int addr, int type) { struct hda_codec *codec; int ret; @@ -145,11 +144,10 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, { #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) struct hdac_hda_priv *hda_priv; - struct hda_codec *codec; int type = HDA_DEV_LEGACY; #endif struct hda_bus *hbus = sof_to_hbus(sdev); - struct hdac_device *hdev; + struct hda_codec *codec; u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; u32 resp = -1; @@ -172,20 +170,20 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (!hda_priv) return -ENOMEM; - hda_priv->codec.bus = hbus; - hdev = &hda_priv->codec.core; - codec = &hda_priv->codec; - /* only probe ASoC codec drivers for HDAC-HDMI */ if (!hda_codec_use_common_hdmi && (resp & 0xFFFF0000) == IDISP_VID_INTEL) type = HDA_DEV_ASOC; - ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev, type); + codec = hda_codec_device_init(&hbus->core, address, type); + ret = PTR_ERR_OR_ZERO(codec); if (ret < 0) return ret; + hda_priv->codec = codec; + dev_set_drvdata(&codec->core.dev, hda_priv); + if ((resp & 0xFFFF0000) == IDISP_VID_INTEL) { - if (!hdev->bus->audio_component) { + if (!hbus->core.audio_component) { dev_dbg(sdev->dev, "iDisp hw present but no driver\n"); ret = -ENOENT; @@ -211,15 +209,12 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, out: if (ret < 0) { - snd_hdac_device_unregister(hdev); - put_device(&hdev->dev); + snd_hdac_device_unregister(&codec->core); + put_device(&codec->core.dev); } #else - hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL); - if (!hdev) - return -ENOMEM; - - ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev, HDA_DEV_ASOC); + codec = hda_codec_device_init(&hbus->core, address); + ret = PTR_ERR_OR_ZERO(codec); #endif return ret; -- cgit v1.2.3 From 0c5c29cafcea20f1f6a9943640e4a5a790e259ee Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 16 Aug 2022 13:17:25 +0200 Subject: ALSA: hda: Always free codec on the device release With all HDAudio drivers aligned to make use of the same constructor, have codec freed on the device release regardless of its type. Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220816111727.3218543-5-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 384426d7e9dd..aa7a362be290 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -883,13 +883,7 @@ static void snd_hda_codec_dev_release(struct device *dev) snd_hda_sysfs_clear(codec); kfree(codec->modelname); kfree(codec->wcaps); - - /* - * In the case of ASoC HD-audio, hda_codec is device managed. - * It will be freed when the ASoC device is removed. - */ - if (codec->core.type == HDA_DEV_LEGACY) - kfree(codec); + kfree(codec); } #define DEV_NAME_LEN 31 -- cgit v1.2.3 From fb5987844808bf9abeb23f695e94b75b439daa42 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 16 Aug 2022 13:17:26 +0200 Subject: ALSA: hda: Remove codec init and exit routines There are no users for snd_hdac_ext_bus_device_init() and snd_hdac_ext_bus_device_exit(). While at it, remove hdac_to_hda_priv() too for the exact same reason. Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220816111727.3218543-6-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- include/sound/hda_codec.h | 2 -- include/sound/hdaudio_ext.h | 3 --- sound/hda/ext/hdac_ext_bus.c | 53 -------------------------------------------- 3 files changed, 58 deletions(-) (limited to 'sound') diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 6d3c82c4b6ac..2a8fe7240f10 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -293,8 +293,6 @@ struct hda_codec { #define dev_to_hda_codec(_dev) container_of(_dev, struct hda_codec, core.dev) #define hda_codec_dev(_dev) (&(_dev)->core.dev) -#define hdac_to_hda_priv(_hdac) \ - container_of(_hdac, struct hdac_hda_priv, codec.core) #define hdac_to_hda_codec(_hdac) container_of(_hdac, struct hda_codec, core) #define list_for_each_codec(c, bus) \ diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index d26234f9ee46..88ebb64fd8a5 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -11,9 +11,6 @@ int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, const struct hdac_ext_bus_ops *ext_ops); void snd_hdac_ext_bus_exit(struct hdac_bus *bus); -int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, - struct hdac_device *hdev, int type); -void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev); void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus); #define HDA_CODEC_REV_EXT_ENTRY(_vid, _rev, _name, drv_data) \ diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 765c40a6ccba..6004ea1c373e 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -60,59 +60,6 @@ void snd_hdac_ext_bus_exit(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_exit); -static void default_release(struct device *dev) -{ - snd_hdac_ext_bus_device_exit(dev_to_hdac_dev(dev)); -} - -/** - * snd_hdac_ext_bus_device_init - initialize the HDA extended codec base device - * @bus: hdac bus to attach to - * @addr: codec address - * @hdev: hdac device to init - * @type: codec type (HDAC_DEV_*) to use for this device - * - * Returns zero for success or a negative error code. - */ -int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, - struct hdac_device *hdev, int type) -{ - char name[15]; - int ret; - - hdev->bus = bus; - - snprintf(name, sizeof(name), "ehdaudio%dD%d", bus->idx, addr); - - ret = snd_hdac_device_init(hdev, bus, name, addr); - if (ret < 0) { - dev_err(bus->dev, "device init failed for hdac device\n"); - return ret; - } - hdev->type = type; - hdev->dev.release = default_release; - - ret = snd_hdac_device_register(hdev); - if (ret) { - dev_err(bus->dev, "failed to register hdac device\n"); - snd_hdac_ext_bus_device_exit(hdev); - return ret; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_init); - -/** - * snd_hdac_ext_bus_device_exit - clean up a HD-audio extended codec base device - * @hdev: hdac device to clean up - */ -void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev) -{ - snd_hdac_device_exit(hdev); -} -EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_exit); - /** * snd_hdac_ext_bus_device_remove - remove HD-audio extended codec base devices * -- cgit v1.2.3 From f2bd1c5ae2cb0cf9525c9bffc0038c12dd7e1338 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Tue, 16 Aug 2022 13:17:27 +0200 Subject: ALSA: hda: Fix page fault in snd_hda_codec_shutdown() If early probe of HDAudio bus driver fails e.g.: due to missing firmware file, snd_hda_codec_shutdown() ends in manipulating uninitialized codec->pcm_list_head causing page fault. Initialization of HDAudio codec in ASoC is split in two: - snd_hda_codec_device_init() - snd_hda_codec_device_new() snd_hda_codec_device_init() is called during probe_codecs() by HDAudio bus driver while snd_hda_codec_device_new() is called by codec-component's ->probe(). The second call will not happen until all components required by related sound card are present within the ASoC framework. With firmware failing to load during the PCI's deferred initialization i.e.: probe_work(), no platform components are ever registered. HDAudio codec enumeration is done at that point though, so the codec components became registered to ASoC framework, calling snd_hda_codec_device_init() in the process. Now, during platform reboot snd_hda_codec_shutdown() is called for every codec found on the HDAudio bus causing oops if any of them has not completed both of their initialization steps. Relocating field initialization fixes the issue. Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220816111727.3218543-7-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 41 ++++++++++++++++++++--------------------- 1 file changed, 20 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index aa7a362be290..b4d1e658c556 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -925,8 +925,28 @@ snd_hda_codec_device_init(struct hda_bus *bus, unsigned int codec_addr, } codec->bus = bus; + codec->depop_delay = -1; + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; + codec->core.dev.release = snd_hda_codec_dev_release; + codec->core.exec_verb = codec_exec_verb; codec->core.type = HDA_DEV_LEGACY; + mutex_init(&codec->spdif_mutex); + mutex_init(&codec->control_mutex); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); + snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); + snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); + snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); + INIT_LIST_HEAD(&codec->conn_list); + INIT_LIST_HEAD(&codec->pcm_list_head); + INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); + refcount_set(&codec->pcm_ref, 1); + init_waitqueue_head(&codec->remove_sleep); + return codec; } EXPORT_SYMBOL_GPL(snd_hda_codec_device_init); @@ -979,29 +999,8 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; - codec->core.dev.release = snd_hda_codec_dev_release; - codec->core.exec_verb = codec_exec_verb; - codec->card = card; codec->addr = codec_addr; - mutex_init(&codec->spdif_mutex); - mutex_init(&codec->control_mutex); - snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); - snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); - snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); - snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); - INIT_LIST_HEAD(&codec->conn_list); - INIT_LIST_HEAD(&codec->pcm_list_head); - refcount_set(&codec->pcm_ref, 1); - init_waitqueue_head(&codec->remove_sleep); - - INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); - codec->depop_delay = -1; - codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM codec->power_jiffies = jiffies; -- cgit v1.2.3 From d91857059defe6acb443d8a25691b43a0f9390e8 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Thu, 18 Aug 2022 16:15:15 +0200 Subject: ALSA: hda: Rework snd_hdac_stream_reset() to use macros MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We can use existing macros to poll and update register values instead of open coding the functionality. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20220818141517.109280-3-amadeuszx.slawinski@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/hdac_stream.c | 26 ++++++-------------------- 1 file changed, 6 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index f3582012d22f..bdf6d4db6769 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -165,7 +165,6 @@ EXPORT_SYMBOL_GPL(snd_hdac_stop_streams_and_chip); void snd_hdac_stream_reset(struct hdac_stream *azx_dev) { unsigned char val; - int timeout; int dma_run_state; snd_hdac_stream_clear(azx_dev); @@ -173,30 +172,17 @@ void snd_hdac_stream_reset(struct hdac_stream *azx_dev) dma_run_state = snd_hdac_stream_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START; snd_hdac_stream_updateb(azx_dev, SD_CTL, 0, SD_CTL_STREAM_RESET); - udelay(3); - timeout = 300; - do { - val = snd_hdac_stream_readb(azx_dev, SD_CTL) & - SD_CTL_STREAM_RESET; - if (val) - break; - } while (--timeout); + + /* wait for hardware to report that the stream entered reset */ + snd_hdac_stream_readb_poll(azx_dev, SD_CTL, val, (val & SD_CTL_STREAM_RESET), 3, 300); if (azx_dev->bus->dma_stop_delay && dma_run_state) udelay(azx_dev->bus->dma_stop_delay); - val &= ~SD_CTL_STREAM_RESET; - snd_hdac_stream_writeb(azx_dev, SD_CTL, val); - udelay(3); + snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_CTL_STREAM_RESET, 0); - timeout = 300; - /* waiting for hardware to report that the stream is out of reset */ - do { - val = snd_hdac_stream_readb(azx_dev, SD_CTL) & - SD_CTL_STREAM_RESET; - if (!val) - break; - } while (--timeout); + /* wait for hardware to report that the stream is out of reset */ + snd_hdac_stream_readb_poll(azx_dev, SD_CTL, val, !(val & SD_CTL_STREAM_RESET), 3, 300); /* reset first position - may not be synced with hw at this time */ if (azx_dev->posbuf) -- cgit v1.2.3 From 21b3d4f58401350cc73e67717366d1127caa6f7f Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Thu, 18 Aug 2022 16:15:16 +0200 Subject: ALSA: hda: Remove unused MAX_PIN_CONFIGS constant MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since it was introduced around v2.6.30 it was never used. Also HDA specification does not mention any limitation on number of PIN configurations. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20220818141517.109280-4-amadeuszx.slawinski@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_sysfs.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index bf951c10ae61..69ebc37a4d6f 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -375,8 +375,6 @@ static ssize_t user_pin_configs_show(struct device *dev, return pin_configs_show(codec, &codec->user_pins, buf); } -#define MAX_PIN_CONFIGS 32 - static int parse_user_pin_configs(struct hda_codec *codec, const char *buf) { int nid, cfg, err; -- cgit v1.2.3 From da9d635f07f21b07ceda13a2ac815a058995f113 Mon Sep 17 00:00:00 2001 From: Amadeusz Sławiński Date: Thu, 18 Aug 2022 16:15:17 +0200 Subject: ALSA: hda: Remove unused defines MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit There is no need to keep unused defines in file. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20220818141517.109280-5-amadeuszx.slawinski@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a77165bd92a9..7720978dc132 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -86,9 +86,6 @@ enum { #define INTEL_SCH_HDA_DEVC 0x78 #define INTEL_SCH_HDA_DEVC_NOSNOOP (0x1<<11) -/* Define VIA HD Audio Device ID*/ -#define VIA_HDAC_DEVICE_ID 0x3288 - /* max number of SDs */ /* ICH, ATI and VIA have 4 playback and 4 capture */ #define ICH6_NUM_CAPTURE 4 @@ -102,10 +99,6 @@ enum { #define ATIHDMI_NUM_CAPTURE 0 #define ATIHDMI_NUM_PLAYBACK 8 -/* TERA has 4 playback and 3 capture */ -#define TERA_NUM_CAPTURE 3 -#define TERA_NUM_PLAYBACK 4 - static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; -- cgit v1.2.3 From 1cda83e42bf66beb06bf61c7a78951ec0c028898 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 19 Aug 2022 14:47:40 +0200 Subject: ASoC: SOF: Fix compilation when HDA_AUDIO_CODEC config is disabled hda_codec_device_init() expects three parameters, not two. Fixes: 3fd63658caed ("ASoC: Intel: Drop hdac_ext usage for codec device creation") Reported-by: kernel test robot Signed-off-by: Cezary Rojewski Acked-by: Mark Brown Link: https://lore.kernel.org/r/20220819124740.3564862-1-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/sof/intel/hda-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 73336648cd25..1e9afc48394c 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -213,7 +213,7 @@ out: put_device(&codec->core.dev); } #else - codec = hda_codec_device_init(&hbus->core, address); + codec = hda_codec_device_init(&hbus->core, address, HDA_DEV_ASOC); ret = PTR_ERR_OR_ZERO(codec); #endif -- cgit v1.2.3 From b01104fc62b6194c852124f6c6df1c0a5c031fc1 Mon Sep 17 00:00:00 2001 From: Conner Knox Date: Thu, 18 Aug 2022 17:14:33 -0300 Subject: ALSA: usb-audio: Add quirk to enable Avid Mbox 3 support Add support for Avid Mbox3 USB audio interface at 48kHz Signed-off-by: Conner Knox Link: https://lore.kernel.org/r/20220818201433.16360-1-mbarriolinares@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 76 ++++++++++++ sound/usb/quirks.c | 302 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 378 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f93201a830b5..06dfdd45cff8 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2985,6 +2985,82 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +/* DIGIDESIGN MBOX 3 */ +{ + USB_DEVICE(0x0dba, 0x5000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "Mbox 3", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x00, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x81, + .attributes = 0x00, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .maxpacksize = 0x009c, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &(const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, { /* Tascam US122 MKII - playback-only support */ USB_DEVICE_VENDOR_SPEC(0x0644, 0x8021), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 168fd802d70b..1b05b0220fad 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1020,6 +1020,304 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } +static void mbox3_setup_48_24_magic(struct usb_device *dev) +{ + /* The Mbox 3 is "little endian" */ + /* max volume is: 0x0000. */ + /* min volume is: 0x0080 (shown in little endian form) */ + + + /* Load 48000Hz rate into buffer */ + u8 com_buff[4] = {0x80, 0xbb, 0x00, 0x00}; + + /* Set 48000Hz sample rate */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x0001, &com_buff, 4); //Is this really needed? + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &com_buff, 4); + + /* Deactivate Tuner */ + /* on = 0x01*/ + /* off = 0x00*/ + com_buff[0] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0003, 0x2001, &com_buff, 1); + + /* Set clock source to Internal (as opposed to S/PDIF) */ + com_buff[0] = 0x01; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x8001, &com_buff, 1); + + /* Mute the hardware loopbacks to start the device in a known state. */ + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* Analogue input 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0110, 0x4001, &com_buff, 2); + /* Analogue input 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0111, 0x4001, &com_buff, 2); + /* Analogue input 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0114, 0x4001, &com_buff, 2); + /* Analogue input 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0115, 0x4001, &com_buff, 2); + /* Analogue input 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0118, 0x4001, &com_buff, 2); + /* Analogue input 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0119, 0x4001, &com_buff, 2); + /* Analogue input 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011c, 0x4001, &com_buff, 2); + /* Analogue input 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011d, 0x4001, &com_buff, 2); + + /* Set software sends to output */ + com_buff[0] = 0x00; + com_buff[1] = 0x00; + /* Analogue software return 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4001, &com_buff, 2); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* Analogue software return 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4001, &com_buff, 2); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* Analogue software return 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4001, &com_buff, 2); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + /* Analogue software return 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4001, &com_buff, 2); + + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* Analogue software return 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0108, 0x4001, &com_buff, 2); + /* Analogue software return 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0109, 0x4001, &com_buff, 2); + /* Analogue software return 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010c, 0x4001, &com_buff, 2); + /* Analogue software return 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010d, 0x4001, &com_buff, 2); + + /* Return to muting sends */ + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* Analogue fx return left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0120, 0x4001, &com_buff, 2); + /* Analogue fx return right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0121, 0x4001, &com_buff, 2); + + /* Analogue software input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4201, &com_buff, 2); + /* Analogue software input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4201, &com_buff, 2); + /* Analogue software input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4201, &com_buff, 2); + /* Analogue software input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4201, &com_buff, 2); + /* Analogue input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4201, &com_buff, 2); + /* Analogue input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4201, &com_buff, 2); + /* Analogue input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4201, &com_buff, 2); + /* Analogue input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4201, &com_buff, 2); + + /* Toggle allowing host control */ + com_buff[0] = 0x02; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + + /* Do not dim fx returns */ + com_buff[0] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0002, 0x2001, &com_buff, 1); + + /* Do not set fx returns to mono */ + com_buff[0] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0001, 0x2001, &com_buff, 1); + + /* Mute the S/PDIF hardware loopback + * same odd volume logic here as above + */ + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* S/PDIF hardware input 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0112, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0113, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0116, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0117, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011a, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011b, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011e, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011f, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4001, &com_buff, 2); + + com_buff[0] = 0x00; + com_buff[1] = 0x00; + /* S/PDIF software return 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010a, 0x4001, &com_buff, 2); + + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* S/PDIF software return 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010b, 0x4001, &com_buff, 2); + /* S/PDIF software return 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010e, 0x4001, &com_buff, 2); + + com_buff[0] = 0x00; + com_buff[1] = 0x00; + /* S/PDIF software return 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010f, 0x4001, &com_buff, 2); + + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* S/PDIF fx returns left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0122, 0x4001, &com_buff, 2); + /* S/PDIF fx returns right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0123, 0x4001, &com_buff, 2); + + /* Set the dropdown "Effect" to the first option */ + /* Room1 = 0x00 */ + /* Room2 = 0x01 */ + /* Room3 = 0x02 */ + /* Hall 1 = 0x03 */ + /* Hall 2 = 0x04 */ + /* Plate = 0x05 */ + /* Delay = 0x06 */ + /* Echo = 0x07 */ + com_buff[0] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0200, 0x4301, &com_buff, 1); /* max is 0xff */ + /* min is 0x00 */ + + + /* Set the effect duration to 0 */ + /* max is 0xffff */ + /* min is 0x0000 */ + com_buff[0] = 0x00; + com_buff[1] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0400, 0x4301, &com_buff, 2); + + /* Set the effect volume and feedback to 0 */ + /* max is 0xff */ + /* min is 0x00 */ + com_buff[0] = 0x00; + /* feedback: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0500, 0x4301, &com_buff, 1); + /* volume: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0300, 0x4301, &com_buff, 1); + + /* Set soft button hold duration */ + /* 0x03 = 250ms */ + /* 0x05 = 500ms DEFAULT */ + /* 0x08 = 750ms */ + /* 0x0a = 1sec */ + com_buff[0] = 0x05; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0005, 0x2001, &com_buff, 1); + + /* Use dim LEDs for button of state */ + com_buff[0] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0004, 0x2001, &com_buff, 1); +} + +#define MBOX3_DESCRIPTOR_SIZE 464 + +static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) +{ + struct usb_host_config *config = dev->actconfig; + int err; + int descriptor_size; + + descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength); + + if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) { + dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size); + return -ENODEV; + } + + dev_dbg(&dev->dev, "device initialised!\n"); + + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) + dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err); + + err = usb_reset_configuration(dev); + if (err < 0) + dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err); + dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + + mbox3_setup_48_24_magic(dev); + dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + + return 0; /* Successful boot */ +} #define MICROBOOK_BUF_SIZE 128 @@ -1324,6 +1622,10 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x0dba, 0x3000): /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); + case USB_ID(0x0dba, 0x5000): + /* Digidesign Mbox 3 */ + return snd_usb_mbox3_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ -- cgit v1.2.3 From 18afcf90d8807fef66d1fd428eeb2b407df90fa8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 22 Aug 2022 21:00:44 +0200 Subject: ALSA: hda: cleanup definitions for multi-link registers For some reason two masks are used without the AZX prefix, and the pattern MLCLT should be ML_LCTL for consistency. Pure rename, no functionality change. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220822190044.170495-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/sound/hda_register.h | 20 +++++++++++--------- sound/hda/ext/hdac_ext_controller.c | 16 ++++++++-------- sound/pci/hda/hda_intel.c | 14 +++++++------- 3 files changed, 26 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index ad8b71b1dbb6..d37cf43546eb 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -260,7 +260,18 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LCAP 0x00 #define AZX_REG_ML_LCTL 0x04 + +#define AZX_ML_LCTL_CPA BIT(23) +#define AZX_ML_LCTL_CPA_SHIFT 23 +#define AZX_ML_LCTL_SPA BIT(16) +#define AZX_ML_LCTL_SPA_SHIFT 16 +#define AZX_ML_LCTL_SCF GENMASK(3, 0) + #define AZX_REG_ML_LOSIDV 0x08 + +/* bit0 is reserved, with BIT(1) mapping to stream1 */ +#define AZX_ML_LOSIDV_STREAM_MASK 0xFFFE + #define AZX_REG_ML_LSDIID 0x0C #define AZX_REG_ML_LPSOO 0x10 #define AZX_REG_ML_LPSIO 0x12 @@ -268,15 +279,6 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LOUTPAY 0x20 #define AZX_REG_ML_LINPAY 0x30 -/* bit0 is reserved, with BIT(1) mapping to stream1 */ -#define ML_LOSIDV_STREAM_MASK 0xFFFE - -#define ML_LCTL_SCF_MASK 0xF -#define AZX_MLCTL_SPA (0x1 << 16) -#define AZX_MLCTL_CPA (0x1 << 23) -#define AZX_MLCTL_SPA_SHIFT 16 -#define AZX_MLCTL_CPA_SHIFT 23 - /* registers for DMA Resume Capability Structure */ #define AZX_DRSM_CAP_ID 0x5 #define AZX_REG_DRSM_CTL 0x4 diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index a42f66f561f5..80876b9a87f4 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -170,7 +170,7 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) { int timeout; u32 val; - int mask = (1 << AZX_MLCTL_CPA_SHIFT); + int mask = (1 << AZX_ML_LCTL_CPA_SHIFT); udelay(3); timeout = 150; @@ -178,10 +178,10 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) do { val = readl(link->ml_addr + AZX_REG_ML_LCTL); if (enable) { - if (((val & mask) >> AZX_MLCTL_CPA_SHIFT)) + if (((val & mask) >> AZX_ML_LCTL_CPA_SHIFT)) return 0; } else { - if (!((val & mask) >> AZX_MLCTL_CPA_SHIFT)) + if (!((val & mask) >> AZX_ML_LCTL_CPA_SHIFT)) return 0; } udelay(3); @@ -197,7 +197,7 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link) { snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, - AZX_MLCTL_SPA, AZX_MLCTL_SPA); + AZX_ML_LCTL_SPA, AZX_ML_LCTL_SPA); return check_hdac_link_power_active(link, true); } @@ -209,7 +209,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_up); */ int snd_hdac_ext_bus_link_power_down(struct hdac_ext_link *link) { - snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0); + snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, AZX_ML_LCTL_SPA, 0); return check_hdac_link_power_active(link, false); } @@ -226,7 +226,7 @@ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, - AZX_MLCTL_SPA, AZX_MLCTL_SPA); + AZX_ML_LCTL_SPA, AZX_ML_LCTL_SPA); ret = check_hdac_link_power_active(hlink, true); if (ret < 0) return ret; @@ -247,7 +247,7 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, - AZX_MLCTL_SPA, 0); + AZX_ML_LCTL_SPA, 0); ret = check_hdac_link_power_active(hlink, false); if (ret < 0) return ret; @@ -281,7 +281,7 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, * clear the register to invalidate all the output streams */ snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, - ML_LOSIDV_STREAM_MASK, 0); + AZX_ML_LOSIDV_STREAM_MASK, 0); /* * wait for 521usec for codec to report status * HDA spec section 4.3 - Codec Discovery diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7720978dc132..bf9df9bc8f1b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -489,14 +489,14 @@ static int intel_ml_lctl_set_power(struct azx *chip, int state) * If other links are enabled for stream, they need similar fix */ val = readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); - val &= ~AZX_MLCTL_SPA; - val |= state << AZX_MLCTL_SPA_SHIFT; + val &= ~AZX_ML_LCTL_SPA; + val |= state << AZX_ML_LCTL_SPA_SHIFT; writel(val, bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); /* wait for CPA */ timeout = 50; while (timeout) { if (((readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL)) & - AZX_MLCTL_CPA) == (state << AZX_MLCTL_CPA_SHIFT)) + AZX_ML_LCTL_CPA) == (state << AZX_ML_LCTL_CPA_SHIFT)) return 0; timeout--; udelay(10); @@ -514,15 +514,15 @@ static void intel_init_lctl(struct azx *chip) /* 0. check lctl register value is correct or not */ val = readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); /* if SCF is already set, let's use it */ - if ((val & ML_LCTL_SCF_MASK) != 0) + if ((val & AZX_ML_LCTL_SCF) != 0) return; /* * Before operating on SPA, CPA must match SPA. * Any deviation may result in undefined behavior. */ - if (((val & AZX_MLCTL_SPA) >> AZX_MLCTL_SPA_SHIFT) != - ((val & AZX_MLCTL_CPA) >> AZX_MLCTL_CPA_SHIFT)) + if (((val & AZX_ML_LCTL_SPA) >> AZX_ML_LCTL_SPA_SHIFT) != + ((val & AZX_ML_LCTL_CPA) >> AZX_ML_LCTL_CPA_SHIFT)) return; /* 1. turn link down: set SPA to 0 and wait CPA to 0 */ @@ -532,7 +532,7 @@ static void intel_init_lctl(struct azx *chip) goto set_spa; /* 2. update SCF to select a properly audio clock*/ - val &= ~ML_LCTL_SCF_MASK; + val &= ~AZX_ML_LCTL_SCF; val |= intel_get_lctl_scf(chip); writel(val, bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); -- cgit v1.2.3 From a5ed0c547d50d30a60d67b0911b4ec2c85c54310 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2022 13:57:36 +0200 Subject: ALSA: vx: Drop superfluous GFP setup The extra setup with GFP_DMA32 is superfluous for this driver. The whole operation is a simple copy loop, and there is no memory address restriction at all. Drop the useless GFP setup. Link: https://lore.kernel.org/r/20220823115740.14123-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 3924f5283745..ceaeb257003b 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -1215,8 +1215,7 @@ int snd_vx_pcm_new(struct vx_core *chip) if (ins) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &vx_pcm_capture_ops); snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, - snd_dma_continuous_data(GFP_KERNEL | GFP_DMA32), - 0, 0); + NULL, 0, 0); pcm->private_data = chip; pcm->private_free = snd_vx_pcm_free; -- cgit v1.2.3 From 63bfc84672bbdfc19e54ce181d094fc1aab09e8c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2022 13:57:37 +0200 Subject: ALSA: pdaudiocf: Drop superfluous GFP setup The extra setup with GFP_DMA32 is superfluous for this driver. The whole operation is a simple copy loop, and there is no memory address restriction at all. Drop the useless GFP setup. Link: https://lore.kernel.org/r/20220823115740.14123-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index dfc4295b69c4..aaa82ec36540 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -257,8 +257,7 @@ int snd_pdacf_pcm_new(struct snd_pdacf *chip) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pdacf_pcm_capture_ops); - snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, - snd_dma_continuous_data(GFP_KERNEL | GFP_DMA32), + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, NULL, 0, 0); pcm->private_data = chip; -- cgit v1.2.3 From 97557ec97a2473ffb9ca5d2e19d21e4807f43fb1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2022 13:57:38 +0200 Subject: ASoC: Intel: sst: Switch to standard device pages ASoC Atom SST driver is using the continuous RAM pages with GFP_DMA flag for its PCM buffer, but this should work fine with the standard DMA pages. As a part of cleanup work, this patch replaces the buffer allocation to the standard device pages with SNDRV_DMA_TYPE_DEV. Link: https://lore.kernel.org/r/20220823115740.14123-4-tiwai@suse.de Signed-off-by: Takashi Iwai Reviewed-by: Cezary Rojewski --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index a56dd48c045f..c75616a5fd0a 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -676,10 +676,9 @@ static int sst_soc_pcm_new(struct snd_soc_component *component, if (dai->driver->playback.channels_min || dai->driver->capture.channels_min) { - snd_pcm_set_managed_buffer_all(pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_DMA), - SST_MIN_BUFFER, SST_MAX_BUFFER); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + pcm->card->dev, + SST_MIN_BUFFER, SST_MAX_BUFFER); } return 0; } -- cgit v1.2.3 From dd164fbfdc20ccf17be9186b1a5a4b2bc11b6a97 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2022 13:57:39 +0200 Subject: ALSA: memalloc: Drop special handling of GFP for CONTINUOUS allocation Now that all users of snd_dma_continuous_data() is gone, let's drop this ugly (and dangerous) way. After this commit, SNDRV_DMA_TYPE_CONTINUOUS may take the standard device pointer instead of the hacked pointer by the macro above, and the memalloc core refers to the coherent_dma_mask of the given device like other SNDRV_DMA_TYPE. It's still allowed to pass NULL there, and in that case, the allocation is performed always in the normal zone. For SNDRV_DMA_TYPE_VMALLOC, the device pointer is simply ignored. Link: https://lore.kernel.org/r/20220823115740.14123-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 3 -- sound/core/memalloc.c | 113 ++++++++++++++++++++--------------------------- 2 files changed, 48 insertions(+), 68 deletions(-) (limited to 'sound') diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 8d79cebf95f3..43d524580bd2 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -26,9 +26,6 @@ struct snd_dma_device { struct device *dev; /* generic device */ }; -#define snd_dma_continuous_data(x) ((struct device *)(__force unsigned long)(x)) - - /* * buffer types */ diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index b665ac66ccbe..39561faef6e9 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -18,25 +18,18 @@ #include #include "memalloc_local.h" +#define DEFAULT_GFP \ + (GFP_KERNEL | \ + __GFP_COMP | /* compound page lets parts be mapped */ \ + __GFP_NORETRY | /* don't trigger OOM-killer */ \ + __GFP_NOWARN) /* no stack trace print - this call is non-critical */ + static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab); #ifdef CONFIG_SND_DMA_SGBUF -static void *do_alloc_fallback_pages(struct device *dev, size_t size, - dma_addr_t *addr, bool wc); -static void do_free_fallback_pages(void *p, size_t size, bool wc); static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); #endif -/* a cast to gfp flag from the dev pointer; for CONTINUOUS and VMALLOC types */ -static inline gfp_t snd_mem_get_gfp_flags(const struct snd_dma_buffer *dmab, - gfp_t default_gfp) -{ - if (!dmab->dev.dev) - return default_gfp; - else - return (__force gfp_t)(unsigned long)dmab->dev.dev; -} - static void *__snd_dma_alloc_pages(struct snd_dma_buffer *dmab, size_t size) { const struct snd_malloc_ops *ops = snd_dma_get_ops(dmab); @@ -284,24 +277,54 @@ EXPORT_SYMBOL(snd_sgbuf_get_chunk_size); /* * Continuous pages allocator */ -static void *do_alloc_pages(size_t size, dma_addr_t *addr, gfp_t gfp) +static void *do_alloc_pages(struct device *dev, size_t size, dma_addr_t *addr, + bool wc) { - void *p = alloc_pages_exact(size, gfp); + void *p; + gfp_t gfp = GFP_KERNEL | __GFP_NORETRY | __GFP_NOWARN; - if (p) - *addr = page_to_phys(virt_to_page(p)); + again: + p = alloc_pages_exact(size, gfp); + if (!p) + return NULL; + *addr = page_to_phys(virt_to_page(p)); + if (!dev) + return p; + if ((*addr + size - 1) & ~dev->coherent_dma_mask) { + if (IS_ENABLED(CONFIG_ZONE_DMA32) && !(gfp & GFP_DMA32)) { + gfp |= GFP_DMA32; + goto again; + } + if (IS_ENABLED(CONFIG_ZONE_DMA) && !(gfp & GFP_DMA)) { + gfp = (gfp & ~GFP_DMA32) | GFP_DMA; + goto again; + } + } +#ifdef CONFIG_X86 + if (wc) + set_memory_wc((unsigned long)(p), size >> PAGE_SHIFT); +#endif return p; } +static void do_free_pages(void *p, size_t size, bool wc) +{ +#ifdef CONFIG_X86 + if (wc) + set_memory_wb((unsigned long)(p), size >> PAGE_SHIFT); +#endif + free_pages_exact(p, size); +} + + static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) { - return do_alloc_pages(size, &dmab->addr, - snd_mem_get_gfp_flags(dmab, GFP_KERNEL)); + return do_alloc_pages(dmab->dev.dev, size, &dmab->addr, false); } static void snd_dma_continuous_free(struct snd_dma_buffer *dmab) { - free_pages_exact(dmab->area, dmab->bytes); + do_free_pages(dmab->area, dmab->bytes, false); } static int snd_dma_continuous_mmap(struct snd_dma_buffer *dmab, @@ -324,9 +347,7 @@ static const struct snd_malloc_ops snd_dma_continuous_ops = { */ static void *snd_dma_vmalloc_alloc(struct snd_dma_buffer *dmab, size_t size) { - gfp_t gfp = snd_mem_get_gfp_flags(dmab, GFP_KERNEL | __GFP_HIGHMEM); - - return __vmalloc(size, gfp); + return vmalloc(size); } static void snd_dma_vmalloc_free(struct snd_dma_buffer *dmab) @@ -440,12 +461,6 @@ static const struct snd_malloc_ops snd_dma_iram_ops = { }; #endif /* CONFIG_GENERIC_ALLOCATOR */ -#define DEFAULT_GFP \ - (GFP_KERNEL | \ - __GFP_COMP | /* compound page lets parts be mapped */ \ - __GFP_NORETRY | /* don't trigger OOM-killer */ \ - __GFP_NOWARN) /* no stack trace print - this call is non-critical */ - /* * Coherent device pages allocator */ @@ -479,12 +494,12 @@ static const struct snd_malloc_ops snd_dma_dev_ops = { #ifdef CONFIG_SND_DMA_SGBUF static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) { - return do_alloc_fallback_pages(dmab->dev.dev, size, &dmab->addr, true); + return do_alloc_pages(dmab->dev.dev, size, &dmab->addr, true); } static void snd_dma_wc_free(struct snd_dma_buffer *dmab) { - do_free_fallback_pages(dmab->area, dmab->bytes, true); + do_free_pages(dmab->area, dmab->bytes, true); } static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, @@ -697,37 +712,6 @@ static const struct snd_malloc_ops snd_dma_sg_wc_ops = { .get_chunk_size = snd_dma_noncontig_get_chunk_size, }; -/* manual page allocations with wc setup */ -static void *do_alloc_fallback_pages(struct device *dev, size_t size, - dma_addr_t *addr, bool wc) -{ - gfp_t gfp = GFP_KERNEL | __GFP_NORETRY | __GFP_NOWARN; - void *p; - - again: - p = do_alloc_pages(size, addr, gfp); - if (!p || (*addr + size - 1) & ~dev->coherent_dma_mask) { - if (IS_ENABLED(CONFIG_ZONE_DMA32) && !(gfp & GFP_DMA32)) { - gfp |= GFP_DMA32; - goto again; - } - if (IS_ENABLED(CONFIG_ZONE_DMA) && !(gfp & GFP_DMA)) { - gfp = (gfp & ~GFP_DMA32) | GFP_DMA; - goto again; - } - } - if (p && wc) - set_memory_wc((unsigned long)(p), size >> PAGE_SHIFT); - return p; -} - -static void do_free_fallback_pages(void *p, size_t size, bool wc) -{ - if (wc) - set_memory_wb((unsigned long)(p), size >> PAGE_SHIFT); - free_pages_exact(p, size); -} - /* Fallback SG-buffer allocations for x86 */ struct snd_dma_sg_fallback { size_t count; @@ -742,7 +726,7 @@ static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab, size_t i; for (i = 0; i < sgbuf->count && sgbuf->pages[i]; i++) - do_free_fallback_pages(page_address(sgbuf->pages[i]), PAGE_SIZE, wc); + do_free_pages(page_address(sgbuf->pages[i]), PAGE_SIZE, wc); kvfree(sgbuf->pages); kvfree(sgbuf->addrs); kfree(sgbuf); @@ -769,8 +753,7 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) goto error; for (i = 0; i < count; sgbuf->count++, i++) { - p = do_alloc_fallback_pages(dmab->dev.dev, PAGE_SIZE, - &sgbuf->addrs[i], wc); + p = do_alloc_pages(dmab->dev.dev, PAGE_SIZE, &sgbuf->addrs[i], wc); if (!p) goto error; sgbuf->pages[i] = virt_to_page(p); -- cgit v1.2.3 From 999b95a72d90ed7a7073eae594fa35462d71854f Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Mon, 29 Aug 2022 16:52:03 -0500 Subject: ALSA: hda/hdmi: Replace zero-length array with DECLARE_FLEX_ARRAY() helper Zero-length arrays are deprecated and we are moving towards adopting C99 flexible-array members, instead. So, replace zero-length array declaration in union audio_infoframe with the new DECLARE_FLEX_ARRAY() helper macro. This helper allows for a flexible-array member in a union. Link: https://github.com/KSPP/linux/issues/193 Link: https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html Signed-off-by: Gustavo A. R. Silva Reviewed-by: Kees Cook Link: https://lore.kernel.org/r/Yw01A+TvF1FWQ588@work Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6c209cd26c0c..2191d445d74e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -229,7 +229,7 @@ struct dp_audio_infoframe { union audio_infoframe { struct hdmi_audio_infoframe hdmi; struct dp_audio_infoframe dp; - u8 bytes[0]; + DECLARE_FLEX_ARRAY(u8, bytes); }; /* -- cgit v1.2.3 From ac5e2fb425e1121ceef2b9d1b3ffccc195d55707 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 31 Aug 2022 15:00:21 +0200 Subject: ALSA: usb-audio: Drop superfluous interface setup at parsing We reset each interface that is being parsed for each stream, but this is superfluous and even can lead to spurious errors. Since the interface is set up properly at opening the endpoint for each actual stream operation, let's drop the superfluous one. Link: https://lore.kernel.org/r/20220831130021.4762-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/stream.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ceb93d798182..99578e9a8af0 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1221,12 +1221,6 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, if (err < 0) return err; } - - /* try to set the interface... */ - usb_set_interface(chip->dev, iface_no, 0); - snd_usb_init_pitch(chip, fp); - snd_usb_init_sample_rate(chip, fp, fp->rate_max); - usb_set_interface(chip->dev, iface_no, altno); } return 0; } -- cgit v1.2.3 From f51ba1148a810a16eead9f0b29bfa2a8f8ab3afb Mon Sep 17 00:00:00 2001 From: Valentina Goncharenko Date: Thu, 1 Sep 2022 13:28:14 +0300 Subject: ALSA: asihpi - Remove useless code in hpi_meter_get_peak() The hpi_meter_get_peak() function contains the expression "hm.obj_index = hm.obj_index", which does not carry any semantic load. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 719f82d3987a ("ALSA: Add support of AudioScience ASI boards") Signed-off-by: Valentina Goncharenko Link: https://lore.kernel.org/r/20220901102814.131855-1-goncharenko.vp@ispras.ru Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpifunc.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 1de05383126a..24047fafef51 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -2020,7 +2020,6 @@ u16 hpi_meter_get_peak(u32 h_control, short an_peakdB[HPI_MAX_CHANNELS] HPI_CONTROL_GET_STATE); if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.obj_index = hm.obj_index; hm.u.c.attribute = HPI_METER_PEAK; hpi_send_recv(&hm, &hr); -- cgit v1.2.3 From 32eeeed963ad4f41b422b3e314d96ded7283b201 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Sep 2022 15:08:31 +0200 Subject: ALSA: usb-audio: Clean up endpoint setups at PCM prepare This patch cleans up the superfluous checks and calls for setting up the endpoints at PCM prepare callback: - Drop stop_endpoints() and sync_pending_stops() calls; the stream is guaranteed to have been already stopped and synced at each PCM prepare call by ALSA PCM core - Call snd_usb_endpoint_prepare() unconditionally; the check for endpoint->need_setup is done in snd_pcm_hw_endpoint_prepare() itself - Apply snd_usb_set_format_quirk() only when the endpoint is actually set up (i.e. the return code from snd_usb_endpoint_prepare() > 0) - Move a few lines back into snd_usb_pcm_prepare(); it's even easier to follow than a small useless function Link: https://lore.kernel.org/r/20220901130831.6136-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 40 ++++++++++------------------------------ 1 file changed, 10 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b604f7e95e82..4ed53a3dc922 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -433,35 +433,6 @@ static void close_endpoints(struct snd_usb_audio *chip, } } -static int configure_endpoints(struct snd_usb_audio *chip, - struct snd_usb_substream *subs) -{ - int err; - - if (subs->data_endpoint->need_setup) { - /* stop any running stream beforehand */ - if (stop_endpoints(subs, false)) - sync_pending_stops(subs); - if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); - if (err < 0) - return err; - snd_usb_set_format_quirk(subs, subs->cur_audiofmt); - } else { - if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - } - - return 0; -} - /* * hw_params callback * @@ -640,9 +611,18 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (ret < 0) goto unlock; + else if (ret > 0) + snd_usb_set_format_quirk(subs, subs->cur_audiofmt); + ret = 0; /* reset the pointer */ subs->buffer_bytes = frames_to_bytes(runtime, runtime->buffer_size); -- cgit v1.2.3 From 6392dcd1d0c7034ccf630ec55fc9e5810ecadf3b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 4 Sep 2022 18:12:47 +0200 Subject: ALSA: usb-audio: Register card at the last interface The USB-audio driver matches per interface, and as default, it registers the card instance at the very first instance. This can be a problem for the devices that have multiple interfaces to be probed, as the udev rule isn't applied properly for the later appearing interfaces. Although we introduced the delayed_register option and the quirks for covering those shortcomings, it's nothing but a workaround for specific devices. This patch is an another attempt to fix the problem in a more generic way. Now the driver checks the whole USB device descriptor at the very first time when an interface is attached to a sound card. It looks at each matching interface in the descriptor and remembers the last matching one. The snd_card_register() is invoked only when this last interface is probed. After this change, the quirks for the delayed registration become superfluous, hence they are removed along with the patch. OTOH, the delayed_register option is still kept, as it might be useful for some corner cases (e.g. a special driver overtakes the interface probe from the standard driver, and the last interface probe may miss). Link: https://lore.kernel.org/r/20220904161247.16461-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.c | 32 +++++++++++++++++++++++++------- sound/usb/quirks.c | 42 ------------------------------------------ sound/usb/quirks.h | 2 -- sound/usb/usbaudio.h | 1 + 4 files changed, 26 insertions(+), 51 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 706d249a9ad6..3aea241435fb 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -690,7 +690,7 @@ static bool get_alias_id(struct usb_device *dev, unsigned int *id) return false; } -static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) +static int check_delayed_register_option(struct snd_usb_audio *chip) { int i; unsigned int id, inum; @@ -699,14 +699,31 @@ static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) if (delayed_register[i] && sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 && id == chip->usb_id) - return iface < inum; + return inum; } - return false; + return -1; } static const struct usb_device_id usb_audio_ids[]; /* defined below */ +/* look for the last interface that matches with our ids and remember it */ +static void find_last_interface(struct snd_usb_audio *chip) +{ + struct usb_host_config *config = chip->dev->actconfig; + struct usb_interface *intf; + int i; + + if (!config) + return; + for (i = 0; i < config->desc.bNumInterfaces; i++) { + intf = config->interface[i]; + if (usb_match_id(intf, usb_audio_ids)) + chip->last_iface = intf->altsetting[0].desc.bInterfaceNumber; + } + usb_audio_dbg(chip, "Found last interface = %d\n", chip->last_iface); +} + /* look for the corresponding quirk */ static const struct snd_usb_audio_quirk * get_alias_quirk(struct usb_device *dev, unsigned int id) @@ -813,6 +830,7 @@ static int usb_audio_probe(struct usb_interface *intf, err = -ENODEV; goto __error; } + find_last_interface(chip); } if (chip->num_interfaces >= MAX_CARD_INTERFACES) { @@ -862,11 +880,11 @@ static int usb_audio_probe(struct usb_interface *intf, chip->need_delayed_register = false; /* clear again */ } - /* we are allowed to call snd_card_register() many times, but first - * check to see if a device needs to skip it or do anything special + /* register card if we reach to the last interface or to the specified + * one given via option */ - if (!snd_usb_registration_quirk(chip, ifnum) && - !check_delayed_register_option(chip, ifnum)) { + if (check_delayed_register_option(chip) == ifnum || + chip->last_iface == ifnum) { err = snd_card_register(chip->card); if (err < 0) goto __error; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 194c75c45628..eadac586bcc8 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2030,48 +2030,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, } } -/* - * registration quirk: - * the registration is skipped if a device matches with the given ID, - * unless the interface reaches to the defined one. This is for delaying - * the registration until the last known interface, so that the card and - * devices appear at the same time. - */ - -struct registration_quirk { - unsigned int usb_id; /* composed via USB_ID() */ - unsigned int interface; /* the interface to trigger register */ -}; - -#define REG_QUIRK_ENTRY(vendor, product, iface) \ - { .usb_id = USB_ID(vendor, product), .interface = (iface) } - -static const struct registration_quirk registration_quirks[] = { - REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */ - REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */ - REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f47, 2), /* JBL Quantum 800 */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f4c, 2), /* JBL Quantum 400 */ - REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ - REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ - REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ - { 0 } /* terminator */ -}; - -/* return true if skipping registration */ -bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface) -{ - const struct registration_quirk *q; - - for (q = registration_quirks; q->usb_id; q++) - if (chip->usb_id == q->usb_id) - return iface < q->interface; - - /* Register as normal */ - return false; -} - /* * driver behavior quirk flags */ diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 31abb7cb01a5..f9bfd5ac7bab 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -48,8 +48,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, struct audioformat *fp, int stream); -bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface); - void snd_usb_init_quirk_flags(struct snd_usb_audio *chip); #endif /* __USBAUDIO_QUIRKS_H */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ffbb4b0d09a0..2c6575029b1c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -37,6 +37,7 @@ struct snd_usb_audio { unsigned int quirk_flags; unsigned int need_delayed_register:1; /* warn for delayed registration */ int num_interfaces; + int last_iface; int num_suspended_intf; int sample_rate_read_error; -- cgit v1.2.3 From 4c8d695cb9bc5f6fd298a586602947b2fc099a64 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Sep 2022 11:23:06 +0200 Subject: ALSA: hda: beep: Simplify keep-power-at-enable behavior The recent fix for IDT codecs to keep the power up while the beep is enabled can be better integrated into the beep helper code. This patch cleans up the code with refactoring. Fixes: 414d38ba8710 ("ALSA: hda/sigmatel: Keep power up while beep is enabled") Link: https://lore.kernel.org/r/20220906092306.26183-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 15 +++++++++++++-- sound/pci/hda/hda_beep.h | 1 + sound/pci/hda/patch_sigmatel.c | 25 ++----------------------- 3 files changed, 16 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 53a2b89f8983..e63621bcb214 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -118,6 +118,12 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, return 0; } +static void turn_on_beep(struct hda_beep *beep) +{ + if (beep->keep_power_at_enable) + snd_hda_power_up_pm(beep->codec); +} + static void turn_off_beep(struct hda_beep *beep) { cancel_work_sync(&beep->beep_work); @@ -125,6 +131,8 @@ static void turn_off_beep(struct hda_beep *beep) /* turn off beep */ generate_tone(beep, 0); } + if (beep->keep_power_at_enable) + snd_hda_power_down_pm(beep->codec); } /** @@ -140,7 +148,9 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) enable = !!enable; if (beep->enabled != enable) { beep->enabled = enable; - if (!enable) + if (enable) + turn_on_beep(beep); + else turn_off_beep(beep); return 1; } @@ -167,7 +177,8 @@ static int beep_dev_disconnect(struct snd_device *device) input_unregister_device(beep->dev); else input_free_device(beep->dev); - turn_off_beep(beep); + if (beep->enabled) + turn_off_beep(beep); return 0; } diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index a25358a4807a..db76e3ddba65 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -25,6 +25,7 @@ struct hda_beep { unsigned int enabled:1; unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */ unsigned int playing:1; + unsigned int keep_power_at_enable:1; /* set by driver */ struct work_struct beep_work; /* scheduled task for beep event */ struct mutex mutex; void (*power_hook)(struct hda_beep *beep, bool on); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7f340f18599c..a794a01a68ca 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4311,6 +4311,8 @@ static int stac_parse_auto_config(struct hda_codec *codec) if (codec->beep) { /* IDT/STAC codecs have linear beep tone parameter */ codec->beep->linear_tone = spec->linear_tone_beep; + /* keep power up while beep is enabled */ + codec->beep->keep_power_at_enable = 1; /* if no beep switch is available, make its own one */ caps = query_amp_caps(codec, nid, HDA_OUTPUT); if (!(caps & AC_AMPCAP_MUTE)) { @@ -4444,28 +4446,6 @@ static int stac_suspend(struct hda_codec *codec) return 0; } - -static int stac_check_power_status(struct hda_codec *codec, hda_nid_t nid) -{ -#ifdef CONFIG_SND_HDA_INPUT_BEEP - struct sigmatel_spec *spec = codec->spec; -#endif - int ret = snd_hda_gen_check_power_status(codec, nid); - -#ifdef CONFIG_SND_HDA_INPUT_BEEP - if (nid == spec->gen.beep_nid && codec->beep) { - if (codec->beep->enabled != spec->beep_power_on) { - spec->beep_power_on = codec->beep->enabled; - if (spec->beep_power_on) - snd_hda_power_up_pm(codec); - else - snd_hda_power_down_pm(codec); - } - ret |= spec->beep_power_on; - } -#endif - return ret; -} #else #define stac_suspend NULL #endif /* CONFIG_PM */ @@ -4478,7 +4458,6 @@ static const struct hda_codec_ops stac_patch_ops = { .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .suspend = stac_suspend, - .check_power_status = stac_check_power_status, #endif }; -- cgit v1.2.3 From aca289f7cd233b3c983b43b59cdaa0d934ea3da7 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Mon, 5 Sep 2022 19:58:25 +0300 Subject: ALSA: hda: cs35l41: Call put_device() in the scope of get_device() When put_device() is called in another function it's hard to realize that and easy to "fix" the code in a wrong way. Instead, move put_device() to be in the same scope as get_device(), so we prevent appearance of any attempts to "fix" the code. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20220905165826.35979-1-andriy.shevchenko@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 15e2a0009080..12e955931044 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1154,7 +1154,6 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd hw_cfg->gpio2.func = CS35L41_INTERRUPT; hw_cfg->gpio2.valid = true; hw_cfg->valid = true; - put_device(physdev); if (strncmp(hid, "CLSA0100", 8) == 0) { hw_cfg->bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; @@ -1204,9 +1203,10 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i property = "cirrus,dev-index"; ret = device_property_count_u32(physdev, property); - if (ret <= 0) - return cs35l41_no_acpi_dsd(cs35l41, physdev, id, hid); - + if (ret <= 0) { + ret = cs35l41_no_acpi_dsd(cs35l41, physdev, id, hid); + goto err_put_physdev; + } if (ret > ARRAY_SIZE(values)) { ret = -EINVAL; goto err; @@ -1295,8 +1295,9 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i return 0; err: - put_device(physdev); dev_err(cs35l41->dev, "Failed property %s: %d\n", property, ret); +err_put_physdev: + put_device(physdev); return ret; } -- cgit v1.2.3 From 7269734abbf5960d0be9050ba3991c0af1d9f574 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Mon, 5 Sep 2022 19:58:26 +0300 Subject: ALSA: hda: cs35l41: Utilize acpi_get_subsystem_id() Replace open coded variant of recently introduced acpi_get_subsystem_id(). Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20220905165826.35979-2-andriy.shevchenko@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 46 +++++++++------------------------------------ 1 file changed, 9 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 12e955931044..3952f2853703 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -842,8 +842,8 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas comps->dev = dev; if (!cs35l41->acpi_subsystem_id) - cs35l41->acpi_subsystem_id = devm_kasprintf(dev, GFP_KERNEL, "%.8x", - comps->codec->core.subsystem_id); + cs35l41->acpi_subsystem_id = kasprintf(GFP_KERNEL, "%.8x", + comps->codec->core.subsystem_id); cs35l41->codec = comps->codec; strscpy(comps->name, dev_name(dev), sizeof(comps->name)); @@ -1048,36 +1048,6 @@ static int cs35l41_hda_apply_properties(struct cs35l41_hda *cs35l41) return cs35l41_hda_channel_map(cs35l41->dev, 0, NULL, 1, &hw_cfg->spk_pos); } -static int cs35l41_get_acpi_sub_string(struct device *dev, struct acpi_device *adev, - const char **subsysid) -{ - struct acpi_buffer buffer = { ACPI_ALLOCATE_BUFFER, NULL }; - union acpi_object *obj; - acpi_status status; - int ret = 0; - - status = acpi_evaluate_object(adev->handle, "_SUB", NULL, &buffer); - if (ACPI_SUCCESS(status)) { - obj = buffer.pointer; - if (obj->type == ACPI_TYPE_STRING) { - *subsysid = devm_kstrdup(dev, obj->string.pointer, GFP_KERNEL); - if (*subsysid == NULL) { - dev_err(dev, "Cannot allocate Subsystem ID"); - ret = -ENOMEM; - } - } else { - dev_warn(dev, "Warning ACPI _SUB did not return a string\n"); - ret = -ENODEV; - } - acpi_os_free(buffer.pointer); - } else { - dev_dbg(dev, "Warning ACPI _SUB failed: %#x\n", status); - ret = -ENODEV; - } - - return ret; -} - static int cs35l41_get_speaker_id(struct device *dev, int amp_index, int num_amps, int fixed_gpio_id) { @@ -1182,6 +1152,7 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i u32 values[HDA_MAX_COMPONENTS]; struct acpi_device *adev; struct device *physdev; + const char *sub; char *property; size_t nval; int i, ret; @@ -1195,11 +1166,10 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i physdev = get_device(acpi_get_first_physical_node(adev)); acpi_dev_put(adev); - ret = cs35l41_get_acpi_sub_string(cs35l41->dev, adev, &cs35l41->acpi_subsystem_id); - if (ret) - dev_info(cs35l41->dev, "No Subsystem ID found in ACPI: %d", ret); - else - dev_dbg(cs35l41->dev, "Subsystem ID %s found", cs35l41->acpi_subsystem_id); + sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + if (IS_ERR(sub)) + sub = NULL; + cs35l41->acpi_subsystem_id = sub; property = "cirrus,dev-index"; ret = device_property_count_u32(physdev, property); @@ -1434,6 +1404,7 @@ err: if (cs35l41_safe_reset(cs35l41->regmap, cs35l41->hw_cfg.bst_type)) gpiod_set_value_cansleep(cs35l41->reset_gpio, 0); gpiod_put(cs35l41->reset_gpio); + kfree(cs35l41->acpi_subsystem_id); return ret; } @@ -1456,6 +1427,7 @@ void cs35l41_hda_remove(struct device *dev) if (cs35l41_safe_reset(cs35l41->regmap, cs35l41->hw_cfg.bst_type)) gpiod_set_value_cansleep(cs35l41->reset_gpio, 0); gpiod_put(cs35l41->reset_gpio); + kfree(cs35l41->acpi_subsystem_id); } EXPORT_SYMBOL_NS_GPL(cs35l41_hda_remove, SND_HDA_SCODEC_CS35L41); -- cgit v1.2.3 From a0e3a293bc001f5aba8a252b6191030ad5911c82 Mon Sep 17 00:00:00 2001 From: Gaosheng Cui Date: Fri, 9 Sep 2022 11:54:42 +0800 Subject: ALSA: line6: remove line6_set_raw declaration line6_set_raw has been removed since commit 9f673d7a6022 ("staging: line6: drop CONFIG_LINE6_USB_RAW"), so remove it. Signed-off-by: Gaosheng Cui Link: https://lore.kernel.org/r/20220909035443.1065737-2-cuigaosheng1@huawei.com Signed-off-by: Takashi Iwai --- sound/usb/line6/driver.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index ecf3a2b39c7e..dbb1d90d3647 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -193,8 +193,6 @@ extern int line6_send_raw_message_async(struct usb_line6 *line6, const char *buffer, int size); extern int line6_send_sysex_message(struct usb_line6 *line6, const char *buffer, int size); -extern ssize_t line6_set_raw(struct device *dev, struct device_attribute *attr, - const char *buf, size_t count); extern int line6_version_request_async(struct usb_line6 *line6); extern int line6_write_data(struct usb_line6 *line6, unsigned address, void *data, unsigned datalen); -- cgit v1.2.3 From 5a55b51a3dea50c8b700cdde0aeb75e0a388486b Mon Sep 17 00:00:00 2001 From: Gaosheng Cui Date: Fri, 9 Sep 2022 11:54:43 +0800 Subject: ALSA: memalloc: remove snd_dma_sg_ops declaration snd_dma_sg_ops has been removed since commit 2c95b92ecd92 ("ALSA: memalloc: Unify x86 SG-buffer handling (take#3)"), so remove it. Signed-off-by: Gaosheng Cui Link: https://lore.kernel.org/r/20220909035443.1065737-3-cuigaosheng1@huawei.com Signed-off-by: Takashi Iwai --- sound/core/memalloc_local.h | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/core/memalloc_local.h b/sound/core/memalloc_local.h index a6f3a87194da..8b19f3a68a4b 100644 --- a/sound/core/memalloc_local.h +++ b/sound/core/memalloc_local.h @@ -13,8 +13,4 @@ struct snd_malloc_ops { void (*sync)(struct snd_dma_buffer *dmab, enum snd_dma_sync_mode mode); }; -#ifdef CONFIG_SND_DMA_SGBUF -extern const struct snd_malloc_ops snd_dma_sg_ops; -#endif - #endif /* __MEMALLOC_LOCAL_H */ -- cgit v1.2.3 From 5b4fc3956bfda2da22a6f7f25b157ad24ba1cd95 Mon Sep 17 00:00:00 2001 From: Gaosheng Cui Date: Fri, 9 Sep 2022 14:11:26 +0800 Subject: sound: oss: dmasound: remove software_input_volume declaration expand_read_bal has been removed since commit fc37449f7959 ("The next round of scheduled OSS code removal"). software_input_volume has been removed since commit 0a1b42db4bf9 ("sound: sound/oss/dmasound/: cleanups"). so remove the declare for them from header file. Signed-off-by: Gaosheng Cui Link: https://lore.kernel.org/r/20220909061126.1129585-1-cuigaosheng1@huawei.com Signed-off-by: Takashi Iwai --- sound/oss/dmasound/dmasound.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h index ad8ce6a1c25c..f065840c0efb 100644 --- a/sound/oss/dmasound/dmasound.h +++ b/sound/oss/dmasound/dmasound.h @@ -250,7 +250,4 @@ extern int dmasound_catchRadius; #define SW_INPUT_VOLUME_SCALE 4 #define SW_INPUT_VOLUME_DEFAULT (128 / SW_INPUT_VOLUME_SCALE) -extern int expand_read_bal; /* Balance factor for reading */ -extern uint software_input_volume; /* software implemented recording volume! */ - #endif /* _dmasound_h_ */ -- cgit v1.2.3 From 7ae22bdf49d513b0555d25df4d361379fc8ad166 Mon Sep 17 00:00:00 2001 From: YJ Lee Date: Mon, 12 Sep 2022 15:28:54 +0800 Subject: ALSA: dummy: Fix trailing whitespaces. Fix checkpatch.pl ERROR: trailing whitespaces. Signed-off-by: YJ Lee Link: https://lore.kernel.org/r/20220912072854.760824-1-yunjunlee@chromium.org Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 2a7fc49c1a7c..fcf1ee00bd21 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -296,7 +296,7 @@ static void dummy_systimer_callback(struct timer_list *t) struct dummy_systimer_pcm *dpcm = from_timer(dpcm, t, timer); unsigned long flags; int elapsed = 0; - + spin_lock_irqsave(&dpcm->lock, flags); dummy_systimer_update(dpcm); dummy_systimer_rearm(dpcm); @@ -717,7 +717,7 @@ static int snd_dummy_volume_info(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 100; return 0; } - + static int snd_dummy_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -766,7 +766,7 @@ static const DECLARE_TLV_DB_SCALE(db_scale_dummy, -4500, 30, 0); .private_value = addr } #define snd_dummy_capsrc_info snd_ctl_boolean_stereo_info - + static int snd_dummy_capsrc_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1100,7 +1100,7 @@ static int snd_dummy_suspend(struct device *pdev) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); return 0; } - + static int snd_dummy_resume(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); -- cgit v1.2.3 From 446bc11f8614449782feac1d5ff270b3f98bcdf3 Mon Sep 17 00:00:00 2001 From: YJ Lee Date: Mon, 12 Sep 2022 15:29:45 +0800 Subject: ALSA: dummy: Add customizable volume min/max. Add module parameters to support customized min/max volume leveling, which will be useful to test devices with different volume granularity. Signed-off-by: YJ Lee Link: https://lore.kernel.org/r/20220912072945.760949-1-yunjunlee@chromium.org Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 34 ++++++++++++++++++++++++---------- 1 file changed, 24 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index fcf1ee00bd21..9c17b49a2ae1 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -42,6 +42,8 @@ MODULE_LICENSE("GPL"); #define USE_CHANNELS_MAX 2 #define USE_PERIODS_MIN 1 #define USE_PERIODS_MAX 1024 +#define USE_MIXER_VOLUME_LEVEL_MIN -50 +#define USE_MIXER_VOLUME_LEVEL_MAX 100 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -50,6 +52,8 @@ static char *model[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = NULL}; static int pcm_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; //static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; +static int mixer_volume_level_min = USE_MIXER_VOLUME_LEVEL_MIN; +static int mixer_volume_level_max = USE_MIXER_VOLUME_LEVEL_MAX; #ifdef CONFIG_HIGH_RES_TIMERS static bool hrtimer = 1; #endif @@ -69,6 +73,10 @@ module_param_array(pcm_substreams, int, NULL, 0444); MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) for dummy driver."); //module_param_array(midi_devs, int, NULL, 0444); //MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver."); +module_param(mixer_volume_level_min, int, 0444); +MODULE_PARM_DESC(mixer_volume_level_min, "Minimum mixer volume level for dummy driver. Default: -50"); +module_param(mixer_volume_level_max, int, 0444); +MODULE_PARM_DESC(mixer_volume_level_max, "Maximum mixer volume level for dummy driver. Default: 100"); module_param(fake_buffer, bool, 0444); MODULE_PARM_DESC(fake_buffer, "Fake buffer allocations."); #ifdef CONFIG_HIGH_RES_TIMERS @@ -713,8 +721,8 @@ static int snd_dummy_volume_info(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; - uinfo->value.integer.min = -50; - uinfo->value.integer.max = 100; + uinfo->value.integer.min = mixer_volume_level_min; + uinfo->value.integer.max = mixer_volume_level_max; return 0; } @@ -739,15 +747,15 @@ static int snd_dummy_volume_put(struct snd_kcontrol *kcontrol, int left, right; left = ucontrol->value.integer.value[0]; - if (left < -50) - left = -50; - if (left > 100) - left = 100; + if (left < mixer_volume_level_min) + left = mixer_volume_level_min; + if (left > mixer_volume_level_max) + left = mixer_volume_level_max; right = ucontrol->value.integer.value[1]; - if (right < -50) - right = -50; - if (right > 100) - right = 100; + if (right < mixer_volume_level_min) + right = mixer_volume_level_min; + if (right > mixer_volume_level_max) + right = mixer_volume_level_max; spin_lock_irq(&dummy->mixer_lock); change = dummy->mixer_volume[addr][0] != left || dummy->mixer_volume[addr][1] != right; @@ -1076,6 +1084,12 @@ static int snd_dummy_probe(struct platform_device *devptr) dummy->pcm_hw.channels_max = m->channels_max; } + if (mixer_volume_level_min > mixer_volume_level_max) { + pr_warn("snd-dummy: Invalid mixer volume level: min=%d, max=%d. Fall back to default value.\n", + mixer_volume_level_min, mixer_volume_level_max); + mixer_volume_level_min = USE_MIXER_VOLUME_LEVEL_MIN; + mixer_volume_level_max = USE_MIXER_VOLUME_LEVEL_MAX; + } err = snd_card_dummy_new_mixer(dummy); if (err < 0) return err; -- cgit v1.2.3 From 4053a41282f8aae290d3fe7b8daef4c8c53a4ab8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 13 Sep 2022 09:03:07 +0200 Subject: ALSA: hda/hdmi: change type for the 'assigned' variable This change converts the assigned value from int type to the bool type to retain consistency with other structure members like 'setup', 'non_pcm' etc. Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220913070307.3234038-1-perex@perex.cz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 429cb4b23a1c..1ab7541a63db 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -53,7 +53,7 @@ MODULE_PARM_DESC(enable_all_pins, "Forcibly enable all pins"); struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; - int assigned; + bool assigned; /* the stream has been assigned */ unsigned int channels_min; unsigned int channels_max; u32 rates; @@ -1193,7 +1193,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, return err; per_cvt = get_cvt(spec, cvt_idx); - per_cvt->assigned = 1; + per_cvt->assigned = true; hinfo->nid = per_cvt->cvt_nid; pin_cvt_fixup(codec, NULL, per_cvt->cvt_nid); @@ -1262,7 +1262,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ - per_cvt->assigned = 1; + per_cvt->assigned = true; set_bit(pcm_idx, &spec->pcm_in_use); per_pin = get_pin(spec, pin_idx); @@ -1296,7 +1296,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hdmi_eld_update_pcm_info(&eld->info, hinfo); if (hinfo->channels_min > hinfo->channels_max || !hinfo->rates || !hinfo->formats) { - per_cvt->assigned = 0; + per_cvt->assigned = false; hinfo->nid = 0; snd_hda_spdif_ctls_unassign(codec, pcm_idx); err = -ENODEV; @@ -1755,7 +1755,7 @@ static void silent_stream_enable(struct hda_codec *codec, } per_cvt = get_cvt(spec, cvt_idx); - per_cvt->assigned = 1; + per_cvt->assigned = true; per_pin->cvt_nid = per_cvt->cvt_nid; per_pin->silent_stream = true; @@ -1815,7 +1815,7 @@ static void silent_stream_disable(struct hda_codec *codec, cvt_idx = cvt_nid_to_cvt_index(codec, per_pin->cvt_nid); if (cvt_idx >= 0 && cvt_idx < spec->num_cvts) { per_cvt = get_cvt(spec, cvt_idx); - per_cvt->assigned = 0; + per_cvt->assigned = false; } if (spec->silent_stream_type == SILENT_STREAM_I915) { @@ -2211,7 +2211,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, goto unlock; } per_cvt = get_cvt(spec, cvt_idx); - per_cvt->assigned = 0; + per_cvt->assigned = false; hinfo->nid = 0; azx_stream(get_azx_dev(substream))->stripe = 0; -- cgit v1.2.3 From a74bfc9eaa497951effddefbcb18f1c7ab56fb35 Mon Sep 17 00:00:00 2001 From: Gaosheng Cui Date: Mon, 22 Aug 2022 11:51:33 +0800 Subject: ASoC: Intel: fix unused-variable warning in probe_codec MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In configurations with CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC=n, gcc warns about an unused variable: sound/soc/intel/skylake/skl.c: In function ‘probe_codec’: sound/soc/intel/skylake/skl.c:729:18: error: unused variable ‘skl’ [-Werror=unused-variable] struct skl_dev *skl = bus_to_skl(bus); ^~~ cc1: all warnings being treated as errors Fixes: 3fd63658caed9 ("ASoC: Intel: Drop hdac_ext usage for codec device creation") Signed-off-by: Gaosheng Cui Acked-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220822035133.2147381-1-cuigaosheng1@huawei.com Signed-off-by: Takashi Iwai --- sound/soc/intel/skylake/skl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index c7c1cad2a753..52a041d6144c 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -726,8 +726,8 @@ static int probe_codec(struct hdac_bus *bus, int addr) unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res = -1; - struct skl_dev *skl = bus_to_skl(bus); #if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) + struct skl_dev *skl = bus_to_skl(bus); struct hdac_hda_priv *hda_codec; #endif struct hda_codec *codec; -- cgit v1.2.3 From 39efc9c8a973ddff5918191525d1679d0fb368ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Sep 2022 10:59:47 +0200 Subject: ALSA: usb-audio: Fix last interface check for registration The recent fix in commit 6392dcd1d0c7 ("ALSA: usb-audio: Register card at the last interface") tried to delay the card registration until the last found interface is probed. It assumed that the probe callback gets called for those later interfaces, but it's not always true; as the driver loops over the descriptor and probes the matching ones, it's not separately called via multiple probe calls. This results in the missing card registration, i.e. no sound device. For addressing this problem, replace the check whether the last interface is processed with usb_interface_claimed() instead of the comparison with the probe interface number. Fixes: 6392dcd1d0c7 ("ALSA: usb-audio: Register card at the last interface") Link: https://lore.kernel.org/r/20220915085947.7922-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 3aea241435fb..a5ed11ea1145 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -884,7 +884,7 @@ static int usb_audio_probe(struct usb_interface *intf, * one given via option */ if (check_delayed_register_option(chip) == ifnum || - chip->last_iface == ifnum) { + usb_interface_claimed(usb_ifnum_to_if(dev, chip->last_iface))) { err = snd_card_register(chip->card); if (err < 0) goto __error; -- cgit v1.2.3 From 7883017bbcc55fcb1888add3dc825e112d7ae336 Mon Sep 17 00:00:00 2001 From: Yang Yingliang Date: Fri, 16 Sep 2022 22:11:08 +0800 Subject: ALSA: ppc: Switch to use for_each_child_of_node() macro Use for_each_child_of_node() macro instead of open coding it. No functional change. Signed-off-by: Yang Yingliang Link: https://lore.kernel.org/r/20220916141108.683080-1-yangyingliang@huawei.com Signed-off-by: Takashi Iwai --- sound/ppc/tumbler.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index c65e74d7cd0a..f3f8ad7c3df8 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1060,8 +1060,7 @@ static struct device_node *find_audio_device(const char *name) if (! gpiop) return NULL; - for (np = of_get_next_child(gpiop, NULL); np; - np = of_get_next_child(gpiop, np)) { + for_each_child_of_node(gpiop, np) { const char *property = of_get_property(np, "audio-gpio", NULL); if (property && strcmp(property, name) == 0) break; @@ -1080,8 +1079,7 @@ static struct device_node *find_compatible_audio_device(const char *name) if (!gpiop) return NULL; - for (np = of_get_next_child(gpiop, NULL); np; - np = of_get_next_child(gpiop, np)) { + for_each_child_of_node(gpiop, np) { if (of_device_is_compatible(np, name)) break; } -- cgit v1.2.3 From fc6f923ecfa2fafd0600f1b7e2de09baf29865e2 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 19 Sep 2022 15:54:44 +0200 Subject: ALSA: hda/hdmi: Fix the converter allocation for the silent stream Track the converters handling the silent stream using a new variable to avoid mixing of the open/close and silent stream use. This change ensures the proper allocation of the converters. Fixes: 5f80d6bd2b01 ("ALSA: hda/hdmi: Fix the converter reuse for the silent stream") Signed-off-by: Jaroslav Kysela Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220919135444.3554982-1-perex@perex.cz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 17f08bf4be14..1eb894e6cdf1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -54,6 +54,7 @@ MODULE_PARM_DESC(enable_all_pins, "Forcibly enable all pins"); struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; bool assigned; /* the stream has been assigned */ + bool silent_stream; /* silent stream activated */ unsigned int channels_min; unsigned int channels_max; u32 rates; @@ -988,7 +989,8 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, * of the pin. */ static int hdmi_choose_cvt(struct hda_codec *codec, - int pin_idx, int *cvt_id) + int pin_idx, int *cvt_id, + bool silent) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin; @@ -1003,6 +1005,9 @@ static int hdmi_choose_cvt(struct hda_codec *codec, if (per_pin && per_pin->silent_stream) { cvt_idx = cvt_nid_to_cvt_index(codec, per_pin->cvt_nid); + per_cvt = get_cvt(spec, cvt_idx); + if (per_cvt->assigned && !silent) + return -EBUSY; if (cvt_id) *cvt_id = cvt_idx; return 0; @@ -1013,7 +1018,7 @@ static int hdmi_choose_cvt(struct hda_codec *codec, per_cvt = get_cvt(spec, cvt_idx); /* Must not already be assigned */ - if (per_cvt->assigned) + if (per_cvt->assigned || per_cvt->silent_stream) continue; if (per_pin == NULL) break; @@ -1199,7 +1204,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, if (pcm_idx < 0) return -EINVAL; - err = hdmi_choose_cvt(codec, -1, &cvt_idx); + err = hdmi_choose_cvt(codec, -1, &cvt_idx, false); if (err) return err; @@ -1267,7 +1272,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, } } - err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx); + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, false); if (err < 0) goto unlock; @@ -1278,7 +1283,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, set_bit(pcm_idx, &spec->pcm_in_use); per_pin = get_pin(spec, pin_idx); per_pin->cvt_nid = per_cvt->cvt_nid; - per_pin->silent_stream = false; hinfo->nid = per_cvt->cvt_nid; /* flip stripe flag for the assigned stream if supported */ @@ -1760,14 +1764,14 @@ static void silent_stream_enable(struct hda_codec *codec, } pin_idx = pin_id_to_pin_index(codec, per_pin->pin_nid, per_pin->dev_id); - err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx); + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, true); if (err) { codec_err(codec, "hdmi: no free converter to enable silent mode\n"); goto unlock_out; } per_cvt = get_cvt(spec, cvt_idx); - per_cvt->assigned = true; + per_cvt->silent_stream = true; per_pin->cvt_nid = per_cvt->cvt_nid; per_pin->silent_stream = true; @@ -1827,7 +1831,7 @@ static void silent_stream_disable(struct hda_codec *codec, cvt_idx = cvt_nid_to_cvt_index(codec, per_pin->cvt_nid); if (cvt_idx >= 0 && cvt_idx < spec->num_cvts) { per_cvt = get_cvt(spec, cvt_idx); - per_cvt->assigned = false; + per_cvt->silent_stream = false; } if (spec->silent_stream_type == SILENT_STREAM_I915) { -- cgit v1.2.3 From 2ea13c83bf7bb3471e33b2d902b101af977ef2d4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 19 Sep 2022 14:10:34 +0200 Subject: ALSA: hda: make snd_hdac_stream_clear() static MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This helper has no users outside of hdac_stream.c. External users should only use snd_hdac_stream_start() and snd_hdac_stream_stop(). No functional change beyond making the function static and removing the symbol export. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220919121041.43463-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 - sound/hda/hdac_stream.c | 5 ++--- 2 files changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 6e74aeafeda4..24c731e53ccb 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -561,7 +561,6 @@ int snd_hdac_stream_setup_periods(struct hdac_stream *azx_dev); int snd_hdac_stream_set_params(struct hdac_stream *azx_dev, unsigned int format_val); void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start); -void snd_hdac_stream_clear(struct hdac_stream *azx_dev); void snd_hdac_stream_stop(struct hdac_stream *azx_dev); void snd_hdac_stop_streams_and_chip(struct hdac_bus *bus); void snd_hdac_stream_reset(struct hdac_stream *azx_dev); diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index bdf6d4db6769..2dbde3d1cf68 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -112,10 +112,10 @@ void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start) EXPORT_SYMBOL_GPL(snd_hdac_stream_start); /** - * snd_hdac_stream_clear - stop a stream DMA + * snd_hdac_stream_clear - helper to clear stream registers and stop DMA transfers * @azx_dev: HD-audio core stream to stop */ -void snd_hdac_stream_clear(struct hdac_stream *azx_dev) +static void snd_hdac_stream_clear(struct hdac_stream *azx_dev) { snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_CTL_DMA_START | SD_INT_MASK, 0); @@ -124,7 +124,6 @@ void snd_hdac_stream_clear(struct hdac_stream *azx_dev) snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0); azx_dev->running = false; } -EXPORT_SYMBOL_GPL(snd_hdac_stream_clear); /** * snd_hdac_stream_stop - stop a stream -- cgit v1.2.3 From ea2ddd2559dc6d1c4b66ccd49314add35ece9062 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 19 Sep 2022 14:10:35 +0200 Subject: ALSA: hda: document state machine for hdac_streams MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The code in this library is far from self-explanatory, hopefully this state diagram reverse-engineered from the code will help others understand the expected transitions. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220919121041.43463-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/hdac_stream.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) (limited to 'sound') diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 2dbde3d1cf68..2e98f5fd50e5 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -13,6 +13,39 @@ #include #include "trace.h" +/* + * the hdac_stream library is intended to be used with the following + * transitions. The states are not formally defined in the code but loosely + * inspired by boolean variables. Note that the 'prepared' field is not used + * in this library but by the callers during the hw_params/prepare transitions + * + * | + * stream_init() | + * v + * +--+-------+ + * | unused | + * +--+----+--+ + * | ^ + * stream_assign() | | stream_release() + * v | + * +--+----+--+ + * | opened | + * +--+----+--+ + * | ^ + * stream_reset() | | + * stream_setup() | | stream_cleanup() + * v | + * +--+----+--+ + * | prepared | + * +--+----+--+ + * | ^ + * stream_start() | | stream_stop() + * v | + * +--+----+--+ + * | running | + * +----------+ + */ + /** * snd_hdac_get_stream_stripe_ctl - get stripe control value * @bus: HD-audio core bus -- cgit v1.2.3 From 791d132a070a8358227b008c0ddda5f4d6f32cc2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 19 Sep 2022 14:10:36 +0200 Subject: ALSA: hda: ext: make snd_hdac_ext_stream_init() static MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit There are no external users of this helper, move to static and remove sympol export. No functionality change. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220919121041.43463-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/sound/hdaudio_ext.h | 3 --- sound/hda/ext/hdac_ext_stream.c | 7 +++---- 2 files changed, 3 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 07231f0b93b5..4a4bd1d88612 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -77,9 +77,6 @@ struct hdac_ext_stream { #define stream_to_hdac_ext_stream(s) \ container_of(s, struct hdac_ext_stream, hstream) -void snd_hdac_ext_stream_init(struct hdac_bus *bus, - struct hdac_ext_stream *hext_stream, int idx, - int direction, int tag); int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx, int num_stream, int dir); void snd_hdac_stream_free_all(struct hdac_bus *bus); diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index d2b5724b463f..5c665b26f853 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -26,9 +26,9 @@ * initialize the stream, if ppcap is enabled then init those and then * invoke hdac stream initialization routine */ -void snd_hdac_ext_stream_init(struct hdac_bus *bus, - struct hdac_ext_stream *hext_stream, - int idx, int direction, int tag) +static void snd_hdac_ext_stream_init(struct hdac_bus *bus, + struct hdac_ext_stream *hext_stream, + int idx, int direction, int tag) { if (bus->ppcap) { hext_stream->pphc_addr = bus->ppcap + AZX_PPHC_BASE + @@ -56,7 +56,6 @@ void snd_hdac_ext_stream_init(struct hdac_bus *bus, hext_stream->decoupled = false; snd_hdac_stream_init(bus, &hext_stream->hstream, idx, direction, tag); } -EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init); /** * snd_hdac_ext_stream_init_all - create and initialize the stream objects -- cgit v1.2.3 From 0839a04eff9778c3dc37d3a1bb8014a3386dece7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 19 Sep 2022 14:10:37 +0200 Subject: ALSA: hda: Use hdac_ext prefix in snd_hdac_stream_free_all() for clarity MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Make sure there's no ambiguity on layering with the appropriate prefix added. Pure rename, no functionality changed. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220919121041.43463-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/sound/hdaudio_ext.h | 2 +- sound/hda/ext/hdac_ext_stream.c | 6 +++--- sound/soc/intel/avs/core.c | 4 ++-- sound/soc/intel/skylake/skl.c | 2 +- 4 files changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 4a4bd1d88612..83aed26ab143 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -79,7 +79,7 @@ struct hdac_ext_stream { int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx, int num_stream, int dir); -void snd_hdac_stream_free_all(struct hdac_bus *bus); +void snd_hdac_ext_stream_free_all(struct hdac_bus *bus); void snd_hdac_link_free_all(struct hdac_bus *bus); struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream, diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 5c665b26f853..9419abd7fc03 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -87,11 +87,11 @@ int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx, EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init_all); /** - * snd_hdac_stream_free_all - free hdac extended stream objects + * snd_hdac_ext_stream_free_all - free hdac extended stream objects * * @bus: HD-audio core bus */ -void snd_hdac_stream_free_all(struct hdac_bus *bus) +void snd_hdac_ext_stream_free_all(struct hdac_bus *bus) { struct hdac_stream *s, *_s; struct hdac_ext_stream *hext_stream; @@ -103,7 +103,7 @@ void snd_hdac_stream_free_all(struct hdac_bus *bus) kfree(hext_stream); } } -EXPORT_SYMBOL_GPL(snd_hdac_stream_free_all); +EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_free_all); void snd_hdac_ext_stream_decouple_locked(struct hdac_bus *bus, struct hdac_ext_stream *hext_stream, diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index c50c20fd681a..bb0719c58ca4 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -466,7 +466,7 @@ static int avs_pci_probe(struct pci_dev *pci, const struct pci_device_id *id) err_acquire_irq: snd_hdac_bus_free_stream_pages(bus); - snd_hdac_stream_free_all(bus); + snd_hdac_ext_stream_free_all(bus); err_init_streams: iounmap(adev->dsp_ba); err_remap_bar4: @@ -502,7 +502,7 @@ static void avs_pci_remove(struct pci_dev *pci) snd_hda_codec_unregister(hdac_to_hda_codec(hdev)); snd_hdac_bus_free_stream_pages(bus); - snd_hdac_stream_free_all(bus); + snd_hdac_ext_stream_free_all(bus); /* reverse ml_capabilities */ snd_hdac_link_free_all(bus); snd_hdac_ext_bus_exit(bus); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 52a041d6144c..0122926f9c58 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -444,7 +444,7 @@ static int skl_free(struct hdac_bus *bus) if (bus->irq >= 0) free_irq(bus->irq, (void *)bus); snd_hdac_bus_free_stream_pages(bus); - snd_hdac_stream_free_all(bus); + snd_hdac_ext_stream_free_all(bus); snd_hdac_link_free_all(bus); if (bus->remap_addr) -- cgit v1.2.3 From 24ad3835a6db4f8857975effa6bf47730371a5ff Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 19 Sep 2022 14:10:38 +0200 Subject: ALSA: hda: add snd_hdac_stop_streams() helper MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Minor code reuse, no functionality change. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220919121041.43463-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 + sound/hda/hdac_stream.c | 17 ++++++++++++++--- sound/pci/hda/hda_controller.c | 4 +--- 3 files changed, 16 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 24c731e53ccb..35459d740f00 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -562,6 +562,7 @@ int snd_hdac_stream_set_params(struct hdac_stream *azx_dev, unsigned int format_val); void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start); void snd_hdac_stream_stop(struct hdac_stream *azx_dev); +void snd_hdac_stop_streams(struct hdac_bus *bus); void snd_hdac_stop_streams_and_chip(struct hdac_bus *bus); void snd_hdac_stream_reset(struct hdac_stream *azx_dev); void snd_hdac_stream_sync_trigger(struct hdac_stream *azx_dev, bool set, diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 2e98f5fd50e5..c056bcc5543d 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -174,17 +174,28 @@ void snd_hdac_stream_stop(struct hdac_stream *azx_dev) } EXPORT_SYMBOL_GPL(snd_hdac_stream_stop); +/** + * snd_hdac_stop_streams - stop all streams + * @bus: HD-audio core bus + */ +void snd_hdac_stop_streams(struct hdac_bus *bus) +{ + struct hdac_stream *stream; + + list_for_each_entry(stream, &bus->stream_list, list) + snd_hdac_stream_stop(stream); +} +EXPORT_SYMBOL_GPL(snd_hdac_stop_streams); + /** * snd_hdac_stop_streams_and_chip - stop all streams and chip if running * @bus: HD-audio core bus */ void snd_hdac_stop_streams_and_chip(struct hdac_bus *bus) { - struct hdac_stream *stream; if (bus->chip_init) { - list_for_each_entry(stream, &bus->stream_list, list) - snd_hdac_stream_stop(stream); + snd_hdac_stop_streams(bus); snd_hdac_bus_stop_chip(bus); } } diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 75dcb14ff20a..0ff286b7b66b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1033,10 +1033,8 @@ EXPORT_SYMBOL_GPL(azx_init_chip); void azx_stop_all_streams(struct azx *chip) { struct hdac_bus *bus = azx_bus(chip); - struct hdac_stream *s; - list_for_each_entry(s, &bus->stream_list, list) - snd_hdac_stream_stop(s); + snd_hdac_stop_streams(bus); } EXPORT_SYMBOL_GPL(azx_stop_all_streams); -- cgit v1.2.3 From 53f4f6b4e56d5fb6ef95a7e14c10ec244a79b996 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 19 Sep 2022 14:10:39 +0200 Subject: ALSA: hda: ext: simplify logic for stream assignment MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The logic is needlessly complicated, the basic rule is: The host streams can be found by checking the 'opened' boolean. The link streams can be found by checking the 'link_locked' boolean. Once a stream is found, it can be unconditionally decoupled. The snd_hdac_ext_stream_decouple_locked() routine will make sure the register status is modified as needed and the 'decoupled' boolean set. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220919121041.43463-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/ext/hdac_ext_stream.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 9419abd7fc03..254df9a67bd2 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -267,19 +267,15 @@ hdac_ext_link_stream_assign(struct hdac_bus *bus, if (hstream->direction != substream->stream) continue; - /* check if decoupled stream and not in use is available */ - if (hext_stream->decoupled && !hext_stream->link_locked) { - res = hext_stream; - break; - } - + /* check if link stream is available */ if (!hext_stream->link_locked) { - snd_hdac_ext_stream_decouple_locked(bus, hext_stream, true); res = hext_stream; break; } + } if (res) { + snd_hdac_ext_stream_decouple_locked(bus, res, true); res->link_locked = 1; res->link_substream = substream; } @@ -308,13 +304,12 @@ hdac_ext_host_stream_assign(struct hdac_bus *bus, continue; if (!hstream->opened) { - if (!hext_stream->decoupled) - snd_hdac_ext_stream_decouple_locked(bus, hext_stream, true); res = hext_stream; break; } } if (res) { + snd_hdac_ext_stream_decouple_locked(bus, res, true); res->hstream.opened = 1; res->hstream.running = 0; res->hstream.substream = substream; -- cgit v1.2.3 From ac3467ad7f8734a21b65fa1852316a9b1b8c1fad Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 19 Sep 2022 14:10:40 +0200 Subject: ALSA: hda: ext: fix locking in stream_release MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The snd_hdac_ext_stream_release() routine uses the bus reg_lock, but releases it before calling snd_hdac_stream_release() where the bus reg_lock is taken again. This creates a timing window where the link stream release could test an invalid 'opened' boolean status and fail to recouple the host and link parts. Fix by exposing a locked version of snd_hdac_stream_release() and use it without releasing the spinlock. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220919121041.43463-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 + sound/hda/ext/hdac_ext_stream.c | 2 +- sound/hda/hdac_stream.c | 19 ++++++++++++++++--- 3 files changed, 18 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 35459d740f00..ddff03e546e9 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -551,6 +551,7 @@ void snd_hdac_stream_init(struct hdac_bus *bus, struct hdac_stream *azx_dev, int idx, int direction, int tag); struct hdac_stream *snd_hdac_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream); +void snd_hdac_stream_release_locked(struct hdac_stream *azx_dev); void snd_hdac_stream_release(struct hdac_stream *azx_dev); struct hdac_stream *snd_hdac_get_stream(struct hdac_bus *bus, int dir, int stream_tag); diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 254df9a67bd2..9a2bc7e803dd 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -384,8 +384,8 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *hext_stream, int type) spin_lock_irq(&bus->reg_lock); if (hext_stream->decoupled && !hext_stream->link_locked) snd_hdac_ext_stream_decouple_locked(bus, hext_stream, false); + snd_hdac_stream_release_locked(&hext_stream->hstream); spin_unlock_irq(&bus->reg_lock); - snd_hdac_stream_release(&hext_stream->hstream); break; case HDAC_EXT_STREAM_TYPE_LINK: diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index c056bcc5543d..1b8be39c38a9 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -365,6 +365,21 @@ struct hdac_stream *snd_hdac_stream_assign(struct hdac_bus *bus, } EXPORT_SYMBOL_GPL(snd_hdac_stream_assign); +/** + * snd_hdac_stream_release_locked - release the assigned stream + * @azx_dev: HD-audio core stream to release + * + * Release the stream that has been assigned by snd_hdac_stream_assign(). + * The bus->reg_lock needs to be taken at a higher level + */ +void snd_hdac_stream_release_locked(struct hdac_stream *azx_dev) +{ + azx_dev->opened = 0; + azx_dev->running = 0; + azx_dev->substream = NULL; +} +EXPORT_SYMBOL_GPL(snd_hdac_stream_release_locked); + /** * snd_hdac_stream_release - release the assigned stream * @azx_dev: HD-audio core stream to release @@ -376,9 +391,7 @@ void snd_hdac_stream_release(struct hdac_stream *azx_dev) struct hdac_bus *bus = azx_dev->bus; spin_lock_irq(&bus->reg_lock); - azx_dev->opened = 0; - azx_dev->running = 0; - azx_dev->substream = NULL; + snd_hdac_stream_release_locked(azx_dev); spin_unlock_irq(&bus->reg_lock); } EXPORT_SYMBOL_GPL(snd_hdac_stream_release); -- cgit v1.2.3 From c6fe6be65aeaa03c7cdfc807b47c1e59b9c9ea71 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 19 Sep 2022 14:10:41 +0200 Subject: ALSA: hda: ext: remove always-true conditions on host and link release MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit By construction a host and link DMA are always decoupled. This decoupling happens in the assign() phase. There's no point in checking if the two parts are decoupled, this is by-design always-true. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220919121041.43463-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/ext/hdac_ext_stream.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 9a2bc7e803dd..70f3ad71aaf0 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -382,7 +382,8 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *hext_stream, int type) case HDAC_EXT_STREAM_TYPE_HOST: spin_lock_irq(&bus->reg_lock); - if (hext_stream->decoupled && !hext_stream->link_locked) + /* couple link only if not in use */ + if (!hext_stream->link_locked) snd_hdac_ext_stream_decouple_locked(bus, hext_stream, false); snd_hdac_stream_release_locked(&hext_stream->hstream); spin_unlock_irq(&bus->reg_lock); @@ -390,7 +391,8 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *hext_stream, int type) case HDAC_EXT_STREAM_TYPE_LINK: spin_lock_irq(&bus->reg_lock); - if (hext_stream->decoupled && !hext_stream->hstream.opened) + /* couple host only if not in use */ + if (!hext_stream->hstream.opened) snd_hdac_ext_stream_decouple_locked(bus, hext_stream, false); hext_stream->link_locked = 0; hext_stream->link_substream = NULL; -- cgit v1.2.3 From 96ecdc718649fe01940e7f5dc4fc15dacd18cada Mon Sep 17 00:00:00 2001 From: ye xingchen Date: Tue, 20 Sep 2022 06:46:05 +0000 Subject: ALSA: es18xx: Remove the unneeded result variable Return the value inb() directly instead of storing it in another redundant variable. Reported-by: Zeal Robot Signed-off-by: ye xingchen Link: https://lore.kernel.org/r/20220920064605.215318-1-ye.xingchen@zte.com.cn Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 3fcd168480b6..0a32845b1017 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1344,11 +1344,8 @@ ES18XX_SINGLE("GPO1 Switch", 0, ES18XX_PM, 1, 1, ES18XX_FL_PMPORT), static int snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { - int data; - outb(reg, chip->ctrl_port); - data = inb(chip->ctrl_port + 1); - return data; + return inb(chip->ctrl_port + 1); } static void snd_es18xx_config_write(struct snd_es18xx *chip, -- cgit v1.2.3 From 01a72aefbacca4d6e169caa776c87d3c1f6faf4a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Sep 2022 13:42:43 +0200 Subject: Revert "ALSA: usb-audio: Clean up endpoint setups at PCM prepare" This reverts commit 32eeeed963ad4f41b422b3e314d96ded7283b201. As the fix for endpoint configuration split is reverted at next, do another revert here for a clean patch application. Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 40 ++++++++++++++++++++++++++++++---------- 1 file changed, 30 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 4ed53a3dc922..b604f7e95e82 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -433,6 +433,35 @@ static void close_endpoints(struct snd_usb_audio *chip, } } +static int configure_endpoints(struct snd_usb_audio *chip, + struct snd_usb_substream *subs) +{ + int err; + + if (subs->data_endpoint->need_setup) { + /* stop any running stream beforehand */ + if (stop_endpoints(subs, false)) + sync_pending_stops(subs); + if (subs->sync_endpoint) { + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + if (err < 0) + return err; + } + err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); + if (err < 0) + return err; + snd_usb_set_format_quirk(subs, subs->cur_audiofmt); + } else { + if (subs->sync_endpoint) { + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + if (err < 0) + return err; + } + } + + return 0; +} + /* * hw_params callback * @@ -611,18 +640,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } - if (subs->sync_endpoint) { - ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); - if (ret < 0) - goto unlock; - } - - ret = snd_usb_endpoint_prepare(chip, subs->data_endpoint); + ret = configure_endpoints(chip, subs); if (ret < 0) goto unlock; - else if (ret > 0) - snd_usb_set_format_quirk(subs, subs->cur_audiofmt); - ret = 0; /* reset the pointer */ subs->buffer_bytes = frames_to_bytes(runtime, runtime->buffer_size); -- cgit v1.2.3 From 2be79d58645465351af5320eb14c70a94724c5ef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Sep 2022 20:11:06 +0200 Subject: ALSA: usb-audio: Split endpoint setups for hw_params and prepare (take#2) This is a second attempt to fix the bug appearing on Android with the recent kernel; the first try was ff878b408a03 and reverted at commit 79764ec772bc. The details taken from the v1 patch: One of the former changes for the endpoint management was the more consistent setup of endpoints at hw_params. snd_usb_endpoint_configure() is a single function that does the full setup, and it's called from both PCM hw_params and prepare callbacks. Although the EP setup at the prepare phase is usually skipped (by checking need_setup flag), it may be still effective in some cases like suspend/resume that requires the interface setup again. As it's a full and single setup, the invocation of snd_usb_endpoint_configure() includes not only the USB interface setup but also the buffer release and allocation. OTOH, doing the buffer release and re-allocation at PCM prepare phase is rather superfluous, and better to be done only in the hw_params phase. For those optimizations, this patch splits the endpoint setup to two phases: snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare(), to be called from hw_params and from prepare, respectively. Note that this patch changes the driver operation slightly, effectively moving the USB interface setup again to PCM prepare stage instead of hw_params stage, while the buffer allocation and such initializations are still done at hw_params stage. And, the change of the USB interface setup timing (moving to prepare) gave an interesting "fix", too: it was reported that the recent kernels caused silent output at the beginning on playbacks on some devices on Android, and this change casually fixed the regression. It seems that those devices are picky about the sample rate change (or the interface change?), and don't follow the too immediate rate changes. Meanwhile, Android operates the PCM in the following order: - open, then hw_params with the possibly highest sample rate - close without prepare - re-open, hw_params with the normal sample rate - prepare, and start streaming This procedure ended up the hw_params twice with different rates, and because the recent kernel did set up the sample rate twice one and after, it screwed up the device. OTOH, the earlier kernels didn't set up the USB interface at hw_params, hence this problem didn't appear. Now, with this patch, the USB interface setup is again back to the prepare phase, and it works around the problem automagically. Although we should address the sample rate problem in a more solid way in future, let's keep things working as before for now. *** What's new in the take#2 patch: - The regression caused by the v1 patch (bko#216500) was due to the missing check of need_setup flag at hw_params. Now the check is added, and the snd_usb_endpoint_set_params() call is skipped when the running EP is re-opened. - There was another bug in v1 where the clock reference rate wasn't updated at hw_params phase, which may lead to a lack of the proper hw constraints when an application doesn't issue the prepare but only the hw_params call. This patch fixes it as well by tracking the clock rate change in the prepare callback with a new flag "need_update" for the clock reference object, just like others. - The configure_endpoints() are simplified and folded back into snd_usb_pcm_prepare(). Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Fixes: ff878b408a03 ("ALSA: usb-audio: Split endpoint setups for hw_params and prepare") Reported-by: chihhao chen Link: https://lore.kernel.org/r/87e6d6ae69d68dc588ac9acc8c0f24d6188375c3.camel@mediatek.com Link: https://lore.kernel.org/r/20220901124136.4984-1-tiwai@suse.de Link: https://bugzilla.kernel.org/show_bug.cgi?id=216500 Link: https://lore.kernel.org/r/20220920181106.4894-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 76 +++++++++++++++++++++++++++++++--------------------- sound/usb/endpoint.h | 6 +++-- sound/usb/pcm.c | 51 ++++++++++++++--------------------- 3 files changed, 70 insertions(+), 63 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index eb71df9da831..0c94ebc98e90 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -40,6 +40,7 @@ struct snd_usb_clock_ref { unsigned char clock; atomic_t locked; int rate; + bool need_setup; struct list_head list; }; @@ -758,7 +759,8 @@ bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, * The endpoint needs to be closed via snd_usb_endpoint_close() later. * * Note that this function doesn't configure the endpoint. The substream - * needs to set it up later via snd_usb_endpoint_configure(). + * needs to set it up later via snd_usb_endpoint_set_params() and + * snd_usb_endpoint_prepare(). */ struct snd_usb_endpoint * snd_usb_endpoint_open(struct snd_usb_audio *chip, @@ -1289,15 +1291,39 @@ out_of_memory: return -ENOMEM; } +/* update the rate of the referred clock; return the actual rate */ +static int update_clock_ref_rate(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) +{ + struct snd_usb_clock_ref *clock = ep->clock_ref; + int rate = ep->cur_rate; + + if (!clock || clock->rate == rate) + return rate; + if (clock->rate) { + if (atomic_read(&clock->locked)) + return clock->rate; + if (clock->rate != rate) { + usb_audio_err(chip, "Mismatched sample rate %d vs %d for EP 0x%x\n", + clock->rate, rate, ep->ep_num); + return clock->rate; + } + } + clock->rate = rate; + clock->need_setup = true; + return rate; +} + /* * snd_usb_endpoint_set_params: configure an snd_usb_endpoint * + * It's called either from hw_params callback. * Determine the number of URBs to be used on this endpoint. * An endpoint must be configured before it can be started. * An endpoint that is already running can not be reconfigured. */ -static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { const struct audioformat *fmt = ep->cur_audiofmt; int err; @@ -1349,49 +1375,46 @@ static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, ep->maxframesize = ep->maxpacksize / ep->cur_frame_bytes; ep->curframesize = ep->curpacksize / ep->cur_frame_bytes; - return 0; + return update_clock_ref_rate(chip, ep); } static int init_sample_rate(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep) { struct snd_usb_clock_ref *clock = ep->clock_ref; - int err; + int rate, err; - if (clock) { - if (atomic_read(&clock->locked)) - return 0; - if (clock->rate == ep->cur_rate) - return 0; - if (clock->rate && clock->rate != ep->cur_rate) { - usb_audio_dbg(chip, "Mismatched sample rate %d vs %d for EP 0x%x\n", - clock->rate, ep->cur_rate, ep->ep_num); - return -EINVAL; - } - } + rate = update_clock_ref_rate(chip, ep); + if (rate < 0) + return rate; + if (clock && !clock->need_setup) + return 0; - err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); - if (err < 0) + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, rate); + if (err < 0) { + if (clock) + clock->rate = 0; /* reset rate */ return err; + } if (clock) - clock->rate = ep->cur_rate; + clock->need_setup = false; return 0; } /* - * snd_usb_endpoint_configure: Configure the endpoint + * snd_usb_endpoint_prepare: Prepare the endpoint * * This function sets up the EP to be fully usable state. - * It's called either from hw_params or prepare callback. + * It's called either from prepare callback. * The function checks need_setup flag, and performs nothing unless needed, * so it's safe to call this multiple times. * * This returns zero if unchanged, 1 if the configuration has changed, * or a negative error code. */ -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { bool iface_first; int err = 0; @@ -1412,9 +1435,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; } - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; goto done; } @@ -1442,10 +1462,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; - err = snd_usb_select_mode_quirk(chip, ep->cur_audiofmt); if (err < 0) goto unlock; diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6a9af04cf175..e67ea28faa54 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -17,8 +17,10 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, bool is_sync_ep); void snd_usb_endpoint_close(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep); -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep); +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock); bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d45d1d7e6664..e721fc12acde 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -433,35 +433,6 @@ static void close_endpoints(struct snd_usb_audio *chip, } } -static int configure_endpoints(struct snd_usb_audio *chip, - struct snd_usb_substream *subs) -{ - int err; - - if (subs->data_endpoint->need_setup) { - /* stop any running stream beforehand */ - if (stop_endpoints(subs, false)) - sync_pending_stops(subs); - if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - err = snd_usb_endpoint_configure(chip, subs->data_endpoint); - if (err < 0) - return err; - snd_usb_set_format_quirk(subs, subs->cur_audiofmt); - } else { - if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - } - - return 0; -} - /* * hw_params callback * @@ -551,7 +522,16 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, subs->cur_audiofmt = fmt; mutex_unlock(&chip->mutex); - ret = configure_endpoints(chip, subs); + if (!subs->data_endpoint->need_setup) + goto unlock; + + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_set_params(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_set_params(chip, subs->data_endpoint); unlock: if (ret < 0) @@ -634,9 +614,18 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (ret < 0) goto unlock; + else if (ret > 0) + snd_usb_set_format_quirk(subs, subs->cur_audiofmt); + ret = 0; /* reset the pointer */ subs->buffer_bytes = frames_to_bytes(runtime, runtime->buffer_size); -- cgit v1.2.3 From 9a737e7f8b371e97eb649904276407cee2c9cf30 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Sep 2022 20:11:26 +0200 Subject: ALSA: usb-audio: Properly refcounting clock rate We fixed the bug introduced by the patch for managing the shared clocks at the commit 809f44a0cc5a ("ALSA: usb-audio: Clear fixed clock rate at closing EP"), but it was merely a workaround. By this change, the clock reference rate is cleared at each EP close, hence the still remaining EP may need a re-setup of rate unnecessarily. This patch introduces the proper refcounting for the clock reference object so that the clock setup is done only when needed. Fixes: 809f44a0cc5a ("ALSA: usb-audio: Clear fixed clock rate at closing EP") Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Link: https://lore.kernel.org/r/20220920181126.4912-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0c94ebc98e90..b2d0b42b581f 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -39,6 +39,7 @@ struct snd_usb_iface_ref { struct snd_usb_clock_ref { unsigned char clock; atomic_t locked; + int opened; int rate; bool need_setup; struct list_head list; @@ -803,6 +804,7 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, ep = NULL; goto unlock; } + ep->clock_ref->opened++; } ep->cur_audiofmt = fp; @@ -926,8 +928,10 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, endpoint_set_interface(chip, ep, false); if (!--ep->opened) { - if (ep->clock_ref && !atomic_read(&ep->clock_ref->locked)) - ep->clock_ref->rate = 0; + if (ep->clock_ref) { + if (!--ep->clock_ref->opened) + ep->clock_ref->rate = 0; + } ep->iface = 0; ep->altsetting = 0; ep->cur_audiofmt = NULL; @@ -1649,8 +1653,7 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, bool keep_pending) WRITE_ONCE(ep->sync_source->sync_sink, NULL); stop_urbs(ep, false, keep_pending); if (ep->clock_ref) - if (!atomic_dec_return(&ep->clock_ref->locked)) - ep->clock_ref->rate = 0; + atomic_dec(&ep->clock_ref->locked); } } -- cgit v1.2.3 From 9bf320f0cf872bf23d9f03abefeff2130acbd6c5 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 21 Sep 2022 11:33:22 +0200 Subject: ALSA: hda/hdmi: Simplify the pcm_idx condition in hdmi_pcm_setup_pin() Make the code more readable. Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220921093322.82609-1-perex@perex.cz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1eb894e6cdf1..11c22dfced06 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1472,10 +1472,9 @@ static void hdmi_pcm_setup_pin(struct hdmi_spec *spec, int mux_idx; bool non_pcm; - if (per_pin->pcm_idx >= 0 && per_pin->pcm_idx < spec->pcm_used) - pcm = get_pcm_rec(spec, per_pin->pcm_idx); - else + if (per_pin->pcm_idx < 0 || per_pin->pcm_idx >= spec->pcm_used) return; + pcm = get_pcm_rec(spec, per_pin->pcm_idx); if (!pcm->pcm) return; if (!test_bit(per_pin->pcm_idx, &spec->pcm_in_use)) -- cgit v1.2.3 From 2fa22c3c755fb06a0c4507320c929616bbae1ec3 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 21 Sep 2022 11:33:49 +0200 Subject: ALSA: hda/hdmi: ELD procfs - print the codec NIDs It is useful for the debugging to print also the used HDA codec NIDs used for the given HDMI device. With the dynamic converter assignment the converter NID is changed dynamically. Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220921093349.82680-1-perex@perex.cz Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 6 +++++- sound/pci/hda/hda_local.h | 3 ++- sound/pci/hda/patch_hdmi.c | 3 ++- 3 files changed, 9 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 9e97443795f8..1d108ed5c6f2 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -440,7 +440,8 @@ static void hdmi_print_sad_info(int i, struct cea_sad *a, } void snd_hdmi_print_eld_info(struct hdmi_eld *eld, - struct snd_info_buffer *buffer) + struct snd_info_buffer *buffer, + hda_nid_t pin_nid, int dev_id, hda_nid_t cvt_nid) { struct parsed_hdmi_eld *e = &eld->info; char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; @@ -462,6 +463,9 @@ void snd_hdmi_print_eld_info(struct hdmi_eld *eld, snd_iprintf(buffer, "monitor_present\t\t%d\n", eld->monitor_present); snd_iprintf(buffer, "eld_valid\t\t%d\n", eld->eld_valid); + snd_iprintf(buffer, "codec_pin_nid\t\t0x%x\n", pin_nid); + snd_iprintf(buffer, "codec_dev_id\t\t0x%x\n", dev_id); + snd_iprintf(buffer, "codec_cvt_nid\t\t0x%x\n", cvt_nid); if (!eld->eld_valid) return; snd_iprintf(buffer, "monitor_name\t\t%s\n", e->monitor_name); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 682dca2057db..53a5a62b78fa 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -712,7 +712,8 @@ int snd_hdmi_get_eld_ati(struct hda_codec *codec, hda_nid_t nid, #ifdef CONFIG_SND_PROC_FS void snd_hdmi_print_eld_info(struct hdmi_eld *eld, - struct snd_info_buffer *buffer); + struct snd_info_buffer *buffer, + hda_nid_t pin_nid, int dev_id, hda_nid_t cvt_nid); void snd_hdmi_write_eld_info(struct hdmi_eld *eld, struct snd_info_buffer *buffer); #endif diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 11c22dfced06..d2c6ba2634f1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -496,7 +496,8 @@ static void print_eld_info(struct snd_info_entry *entry, struct hdmi_spec_per_pin *per_pin = entry->private_data; mutex_lock(&per_pin->lock); - snd_hdmi_print_eld_info(&per_pin->sink_eld, buffer); + snd_hdmi_print_eld_info(&per_pin->sink_eld, buffer, per_pin->pin_nid, + per_pin->dev_id, per_pin->cvt_nid); mutex_unlock(&per_pin->lock); } -- cgit v1.2.3 From b5eee17cf7ddaf7b29a031b2c48277038e7a171a Mon Sep 17 00:00:00 2001 From: ye xingchen Date: Thu, 22 Sep 2022 11:28:46 +0000 Subject: ALSA: hda/ca0132 - remove the unneeded result variable Return the value dsp_allocate_ports() directly instead of storing it in another redundant variable. Reported-by: Zeal Robot Signed-off-by: ye xingchen Link: https://lore.kernel.org/r/20220922112846.236987-1-ye.xingchen@zte.com.cn Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 208933792787..9580fe00cbd9 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2962,7 +2962,6 @@ static int dsp_allocate_ports_format(struct hda_codec *codec, const unsigned short fmt, unsigned int *port_map) { - int status; unsigned int num_chans; unsigned int sample_rate_div = ((get_hdafmt_rate(fmt) >> 0) & 3) + 1; @@ -2976,9 +2975,7 @@ static int dsp_allocate_ports_format(struct hda_codec *codec, num_chans = get_hdafmt_chs(fmt) + 1; - status = dsp_allocate_ports(codec, num_chans, rate_multi, port_map); - - return status; + return dsp_allocate_ports(codec, num_chans, rate_multi, port_map); } /* -- cgit v1.2.3 From ef6f5494faf6a37c74990689a3bb3cee76d2544c Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 22 Sep 2022 10:40:17 +0200 Subject: ALSA: hda/hdmi: Use only dynamic PCM device allocation Per discussion on the alsa-devel mailing list [1], the legacy PIN to PCM device mapping is obsolete nowadays. The maximum number of the simultaneously usable PCM devices is equal to the HDMI codec converters. Remove the extra PCM devices (beyond the detected converters) and force the use of the dynamic PCM device allocation. The legacy code is removed. I believe that all HDMI codecs have the jack sensing feature. Move the check to the codec probe function and print a warning, if a codec without this feature is detected. [1] https://lore.kernel.org/alsa-devel/2f37e0b2-1e82-8c0b-2bbd-1e5038d6ecc6@perex.cz/ Cc: Kai Vehmanen Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220922084017.25925-1-perex@perex.cz Signed-off-by: Takashi Iwai --- include/sound/hda_codec.h | 1 - sound/pci/hda/patch_hdmi.c | 153 ++++++++------------------------------------ sound/soc/codecs/hda.c | 3 - sound/soc/codecs/hdac_hda.c | 3 - 4 files changed, 28 insertions(+), 132 deletions(-) (limited to 'sound') diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 2a8fe7240f10..25ec8c181688 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -258,7 +258,6 @@ struct hda_codec { unsigned int link_down_at_suspend:1; /* link down at runtime suspend */ unsigned int relaxed_resume:1; /* don't resume forcibly for jack */ unsigned int forced_resume:1; /* forced resume for jack */ - unsigned int mst_no_extra_pcms:1; /* no backup PCMs for DP-MST */ #ifdef CONFIG_PM unsigned long power_on_acct; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d2c6ba2634f1..1863836b2685 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -167,8 +167,6 @@ struct hdmi_spec { struct hdmi_ops ops; bool dyn_pin_out; - bool dyn_pcm_assign; - bool dyn_pcm_no_legacy; /* hdmi interrupt trigger control flag for Nvidia codec */ bool hdmi_intr_trig_ctrl; bool nv_dp_workaround; /* workaround DP audio infoframe for Nvidia */ @@ -1188,9 +1186,7 @@ static void pin_cvt_fixup(struct hda_codec *codec, spec->ops.pin_cvt_fixup(codec, per_pin, cvt_nid); } -/* called in hdmi_pcm_open when no pin is assigned to the PCM - * in dyn_pcm_assign mode. - */ +/* called in hdmi_pcm_open when no pin is assigned to the PCM */ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1258,19 +1254,12 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mutex_lock(&spec->pcm_lock); pin_idx = hinfo_to_pin_index(codec, hinfo); - if (!spec->dyn_pcm_assign) { - if (snd_BUG_ON(pin_idx < 0)) { - err = -EINVAL; - goto unlock; - } - } else { - /* no pin is assigned to the PCM - * PA need pcm open successfully when probe - */ - if (pin_idx < 0) { - err = hdmi_pcm_open_no_pin(hinfo, codec, substream); - goto unlock; - } + /* no pin is assigned to the PCM + * PA need pcm open successfully when probe + */ + if (pin_idx < 0) { + err = hdmi_pcm_open_no_pin(hinfo, codec, substream); + goto unlock; } err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, false); @@ -1375,43 +1364,6 @@ static int hdmi_find_pcm_slot(struct hdmi_spec *spec, { int i; - /* on the new machines, try to assign the pcm slot dynamically, - * not use the preferred fixed map (legacy way) anymore. - */ - if (spec->dyn_pcm_no_legacy) - goto last_try; - - /* - * generic_hdmi_build_pcms() may allocate extra PCMs on some - * platforms (with maximum of 'num_nids + dev_num - 1') - * - * The per_pin of pin_nid_idx=n and dev_id=m prefers to get pcm-n - * if m==0. This guarantees that dynamic pcm assignments are compatible - * with the legacy static per_pin-pcm assignment that existed in the - * days before DP-MST. - * - * Intel DP-MST prefers this legacy behavior for compatibility, too. - * - * per_pin of m!=0 prefers to get pcm=(num_nids + (m - 1)). - */ - - if (per_pin->dev_id == 0 || spec->intel_hsw_fixup) { - if (!test_bit(per_pin->pin_nid_idx, &spec->pcm_bitmap)) - return per_pin->pin_nid_idx; - } else { - i = spec->num_nids + (per_pin->dev_id - 1); - if (i < spec->pcm_used && !(test_bit(i, &spec->pcm_bitmap))) - return i; - } - - /* have a second try; check the area over num_nids */ - for (i = spec->num_nids; i < spec->pcm_used; i++) { - if (!test_bit(i, &spec->pcm_bitmap)) - return i; - } - - last_try: - /* the last try; check the empty slots in pins */ for (i = 0; i < spec->pcm_used; i++) { if (!test_bit(i, &spec->pcm_bitmap)) return i; @@ -1573,14 +1525,12 @@ static void update_eld(struct hda_codec *codec, */ pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); - if (spec->dyn_pcm_assign) { - if (eld->eld_valid) { - hdmi_attach_hda_pcm(spec, per_pin); - hdmi_pcm_setup_pin(spec, per_pin); - } else { - hdmi_pcm_reset_pin(spec, per_pin); - hdmi_detach_hda_pcm(spec, per_pin); - } + if (eld->eld_valid) { + hdmi_attach_hda_pcm(spec, per_pin); + hdmi_pcm_setup_pin(spec, per_pin); + } else { + hdmi_pcm_reset_pin(spec, per_pin); + hdmi_detach_hda_pcm(spec, per_pin); } /* if pcm_idx == -1, it means this is in monitor connection event * we can get the correct pcm_idx now. @@ -1942,7 +1892,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * structures based on worst case. */ dev_num = spec->dev_num; - } else if (spec->dyn_pcm_assign && codec->dp_mst) { + } else if (codec->dp_mst) { dev_num = snd_hda_get_num_devices(codec, pin_nid) + 1; /* * spec->dev_num is the maxinum number of device entries @@ -1967,13 +1917,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (!per_pin) return -ENOMEM; - if (spec->dyn_pcm_assign) { - per_pin->pcm = NULL; - per_pin->pcm_idx = -1; - } else { - per_pin->pcm = get_hdmi_pcm(spec, pin_idx); - per_pin->pcm_idx = pin_idx; - } + per_pin->pcm = NULL; + per_pin->pcm_idx = -1; per_pin->pin_nid = pin_nid; per_pin->pin_nid_idx = spec->num_nids; per_pin->dev_id = i; @@ -1982,6 +1927,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) err = hdmi_read_pin_conn(codec, pin_idx); if (err < 0) return err; + if (!is_jack_detectable(codec, pin_nid)) + codec_warn(codec, "HDMI: pin NID 0x%x - jack not detectable\n", pin_nid); spec->num_pins++; } spec->num_nids++; @@ -2129,10 +2076,9 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, mutex_lock(&spec->pcm_lock); pin_idx = hinfo_to_pin_index(codec, hinfo); - if (spec->dyn_pcm_assign && pin_idx < 0) { - /* when dyn_pcm_assign and pcm is not bound to a pin - * skip pin setup and return 0 to make audio playback - * be ongoing + if (pin_idx < 0) { + /* when pcm is not bound to a pin skip pin setup and return 0 + * to make audio playback be ongoing */ pin_cvt_fixup(codec, NULL, cvt_nid); snd_hda_codec_setup_stream(codec, cvt_nid, @@ -2235,7 +2181,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, snd_hda_spdif_ctls_unassign(codec, pcm_idx); clear_bit(pcm_idx, &spec->pcm_in_use); pin_idx = hinfo_to_pin_index(codec, hinfo); - if (spec->dyn_pcm_assign && pin_idx < 0) + if (pin_idx < 0) goto unlock; if (snd_BUG_ON(pin_idx < 0)) { @@ -2333,21 +2279,8 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) struct hdmi_spec *spec = codec->spec; int idx, pcm_num; - /* - * for non-mst mode, pcm number is the same as before - * for DP MST mode without extra PCM, pcm number is same - * for DP MST mode with extra PCMs, pcm number is - * (nid number + dev_num - 1) - * dev_num is the device entry number in a pin - */ - - if (spec->dyn_pcm_no_legacy && codec->mst_no_extra_pcms) - pcm_num = spec->num_cvts; - else if (codec->mst_no_extra_pcms) - pcm_num = spec->num_nids; - else - pcm_num = spec->num_nids + spec->dev_num - 1; - + /* limit the PCM devices to the codec converters */ + pcm_num = spec->num_cvts; codec_dbg(codec, "hdmi: pcm_num set to %d\n", pcm_num); for (idx = 0; idx < pcm_num; idx++) { @@ -2386,17 +2319,12 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx) { char hdmi_str[32] = "HDMI/DP"; struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pcm_idx); struct snd_jack *jack; int pcmdev = get_pcm_rec(spec, pcm_idx)->device; int err; if (pcmdev > 0) sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); - if (!spec->dyn_pcm_assign && - !is_jack_detectable(codec, per_pin->pin_nid)) - strncat(hdmi_str, " Phantom", - sizeof(hdmi_str) - strlen(hdmi_str) - 1); err = snd_jack_new(codec->card, hdmi_str, SND_JACK_AVOUT, &jack, true, false); @@ -2429,18 +2357,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) /* create the spdif for each pcm * pin will be bound when monitor is connected */ - if (spec->dyn_pcm_assign) - err = snd_hda_create_dig_out_ctls(codec, + err = snd_hda_create_dig_out_ctls(codec, 0, spec->cvt_nids[0], HDA_PCM_TYPE_HDMI); - else { - struct hdmi_spec_per_pin *per_pin = - get_pin(spec, pcm_idx); - err = snd_hda_create_dig_out_ctls(codec, - per_pin->pin_nid, - per_pin->mux_nids[0], - HDA_PCM_TYPE_HDMI); - } if (err < 0) return err; snd_hda_spdif_ctls_unassign(codec, pcm_idx); @@ -2560,11 +2479,7 @@ static void generic_hdmi_free(struct hda_codec *codec) for (pcm_idx = 0; pcm_idx < spec->pcm_used; pcm_idx++) { if (spec->pcm_rec[pcm_idx].jack == NULL) continue; - if (spec->dyn_pcm_assign) - snd_device_free(codec->card, - spec->pcm_rec[pcm_idx].jack); - else - spec->pcm_rec[pcm_idx].jack = NULL; + snd_device_free(codec->card, spec->pcm_rec[pcm_idx].jack); } generic_spec_free(codec); @@ -3044,7 +2959,6 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, return err; spec = codec->spec; codec->dp_mst = true; - spec->dyn_pcm_assign = true; spec->vendor_nid = vendor_nid; spec->port_map = port_map; spec->port_num = port_num; @@ -3108,17 +3022,9 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) * the index indicate the port number. */ static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; - int ret; - - ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 4, - enable_silent_stream); - if (!ret) { - struct hdmi_spec *spec = codec->spec; - spec->dyn_pcm_no_legacy = true; - } - - return ret; + return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 4, + enable_silent_stream); } static int patch_i915_adlp_hdmi(struct hda_codec *codec) @@ -3758,7 +3664,6 @@ static int patch_nvhdmi(struct hda_codec *codec) codec->dp_mst = true; spec = codec->spec; - spec->dyn_pcm_assign = true; err = hdmi_parse_codec(codec); if (err < 0) { @@ -4038,10 +3943,8 @@ static int patch_tegra234_hdmi(struct hda_codec *codec) return err; codec->dp_mst = true; - codec->mst_no_extra_pcms = true; spec = codec->spec; spec->dyn_pin_out = true; - spec->dyn_pcm_assign = true; spec->hdmi_intr_trig_ctrl = true; return tegra_hdmi_init(codec); diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c index ad20a3dff9b7..61e8e9be6b8d 100644 --- a/sound/soc/codecs/hda.c +++ b/sound/soc/codecs/hda.c @@ -224,9 +224,6 @@ static int hda_codec_probe(struct snd_soc_component *component) goto err; } - /* configure codec for 1:1 PCM:DAI mapping */ - codec->mst_no_extra_pcms = 1; - ret = snd_hda_codec_parse_pcms(codec); if (ret < 0) { dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 77df4c5b274a..8af434e14bfb 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -461,9 +461,6 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) dev_dbg(&hdev->dev, "no patch file found\n"); } - /* configure codec for 1:1 PCM:DAI mapping */ - hcodec->mst_no_extra_pcms = 1; - ret = snd_hda_codec_parse_pcms(hcodec); if (ret < 0) { dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); -- cgit v1.2.3 From b23975e60a944e1a3ef419a01838fca51a29baf3 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 23 Sep 2022 10:22:36 +0200 Subject: ALSA: hda/hdmi: Limit the maximal count of PCM devices to 8 The current hardware has up to 4 converters. Save little space. The limit 8 is enough even for a more improved hardware. Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220923082236.61024-1-perex@perex.cz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1863836b2685..c172640c8a41 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -151,7 +151,7 @@ struct hdmi_spec { */ int dev_num; struct snd_array pins; /* struct hdmi_spec_per_pin */ - struct hdmi_pcm pcm_rec[16]; + struct hdmi_pcm pcm_rec[8]; struct mutex pcm_lock; struct mutex bind_lock; /* for audio component binding */ /* pcm_bitmap means which pcms have been assigned to pins*/ @@ -2299,8 +2299,8 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; pstr->substreams = 1; pstr->ops = generic_ops; - /* pcm number is less than 16 */ - if (spec->pcm_used >= 16) + /* pcm number is less than pcm_rec array size */ + if (spec->pcm_used >= ARRAY_SIZE(spec->pcm_rec)) break; /* other pstr fields are set in open */ } -- cgit v1.2.3 From a61c7d88d38cf3b9c88cf667c4f8a389a57744d4 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Fri, 23 Sep 2022 18:35:01 +0300 Subject: ALSA: memalloc: use __GFP_RETRY_MAYFAIL for DMA mem allocs Use __GFP_RETRY_MAYFAIL instead of __GFP__NORETRY in snd_dma_dev_alloc(), snd_dma_wc_alloc() and friends, to allocate pages for device memory. The MAYFAIL flag retains the semantics of not triggering the OOM killer, but lowers the risk of alloc failure. MAYFAIL flag was added in commit dcda9b04713c3 ("mm, tree wide: replace __GFP_REPEAT by __GFP_RETRY_MAYFAIL with more useful semantic"). This change addresses recurring failures with SOF audio driver in test cases where a system suspend-resume stress test is run, combined with an active high memory-load use-case. The failure typically shows up as: [ 379.480229] sof-audio-pci-intel-tgl 0000:00:1f.3: booting DSP firmware [ 379.484803] sof-audio-pci-intel-tgl 0000:00:1f.3: error: memory alloc failed: -12 [ 379.484810] sof-audio-pci-intel-tgl 0000:00:1f.3: error: dma prepare for ICCMAX stream failed Multiple fixes to reduce the memory usage of DSP boot have been identified in SOF driver, but even with those fixes, debug on affected systems has shown that even a single page alloc may fail with __GFP_NORETRY. When this occurs, system is under significant load on physical memory, but a lot of reclaimable pages are available, so the system has not run out of memory. With __GFP_RETRY_MAYFAIL, the errors are not hit in these stress tests. The alloc failure is severe as audio capability is completely lost if alloc failure is hit at system resume. An alternative solution was considered where the resources for DSP boot would be kept allocated until driver is unbound. This would avoid the allocation failure, but consume memory that is only needed temporarily at probe and resume time. It seems better to not hang on to the memory, but rather work a bit harder for allocating the pages at resume. BugLink: https://github.com/thesofproject/linux/issues/3844 Signed-off-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220923153501.3326041-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 2c11413bea61..03cffe771366 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -21,7 +21,7 @@ #define DEFAULT_GFP \ (GFP_KERNEL | \ __GFP_COMP | /* compound page lets parts be mapped */ \ - __GFP_NORETRY | /* don't trigger OOM-killer */ \ + __GFP_RETRY_MAYFAIL | /* don't trigger OOM-killer */ \ __GFP_NOWARN) /* no stack trace print - this call is non-critical */ static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab); -- cgit v1.2.3 From 1dd0dd0b1fefd1e51cfaddf62316f759fde7de7d Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Sun, 25 Sep 2022 14:57:51 +0200 Subject: ALSA: firewire: Remove some left-over license text in sound/firewire There is already a SPDX-License-Identifier tag, so the corresponding license text can be removed. While at it, be more consistent and: - add a missing .c (ff-protocol-latter) - remove an empty line (motu-protocol-v1) Signed-off-by: Christophe JAILLET Link: https://lore.kernel.org/r/2bfe76c7eeb0f5205a1427e280bf8d9da0354a62.1664110649.git.christophe.jaillet@wanadoo.fr Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-harman.c | 2 -- sound/firewire/dice/dice-presonus.c | 2 -- sound/firewire/fireface/ff-protocol-former.c | 2 -- sound/firewire/fireface/ff-protocol-latter.c | 4 +--- sound/firewire/motu/motu-protocol-v1.c | 3 --- 5 files changed, 1 insertion(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/firewire/dice/dice-harman.c b/sound/firewire/dice/dice-harman.c index a8ca00c397e8..212ae77dfca2 100644 --- a/sound/firewire/dice/dice-harman.c +++ b/sound/firewire/dice/dice-harman.c @@ -2,8 +2,6 @@ // dice-harman.c - a part of driver for DICE based devices // // Copyright (c) 2021 Takashi Sakamoto -// -// Licensed under the terms of the GNU General Public License, version 2. #include "dice.h" diff --git a/sound/firewire/dice/dice-presonus.c b/sound/firewire/dice/dice-presonus.c index 503f462a83f4..967cc3119a64 100644 --- a/sound/firewire/dice/dice-presonus.c +++ b/sound/firewire/dice/dice-presonus.c @@ -2,8 +2,6 @@ // dice-presonus.c - a part of driver for DICE based devices // // Copyright (c) 2019 Takashi Sakamoto -// -// Licensed under the terms of the GNU General Public License, version 2. #include "dice.h" diff --git a/sound/firewire/fireface/ff-protocol-former.c b/sound/firewire/fireface/ff-protocol-former.c index bf44cad7985e..8900ffe517ed 100644 --- a/sound/firewire/fireface/ff-protocol-former.c +++ b/sound/firewire/fireface/ff-protocol-former.c @@ -2,8 +2,6 @@ // ff-protocol-former.c - a part of driver for RME Fireface series // // Copyright (c) 2019 Takashi Sakamoto -// -// Licensed under the terms of the GNU General Public License, version 2. #include diff --git a/sound/firewire/fireface/ff-protocol-latter.c b/sound/firewire/fireface/ff-protocol-latter.c index 7ddb7b97f02d..76c3eab36d4e 100644 --- a/sound/firewire/fireface/ff-protocol-latter.c +++ b/sound/firewire/fireface/ff-protocol-latter.c @@ -1,9 +1,7 @@ // SPDX-License-Identifier: GPL-2.0 -// ff-protocol-latter - a part of driver for RME Fireface series +// ff-protocol-latter.c - a part of driver for RME Fireface series // // Copyright (c) 2019 Takashi Sakamoto -// -// Licensed under the terms of the GNU General Public License, version 2. #include diff --git a/sound/firewire/motu/motu-protocol-v1.c b/sound/firewire/motu/motu-protocol-v1.c index f1d6a326dc07..e811629f167b 100644 --- a/sound/firewire/motu/motu-protocol-v1.c +++ b/sound/firewire/motu/motu-protocol-v1.c @@ -1,10 +1,7 @@ // SPDX-License-Identifier: GPL-2.0-only - // motu-protocol-v1.c - a part of driver for MOTU FireWire series // // Copyright (c) 2021 Takashi Sakamoto -// -// Licensed under the terms of the GNU General Public License, version 2. #include "motu.h" -- cgit v1.2.3 From f0061c18c169f0c32d96b59485c3edee85e343ed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Sep 2022 15:55:48 +0200 Subject: ALSA: pcm: Avoid reference to status->state In the PCM core and driver code, there are lots place referring to the current PCM state via runtime->status->state. This patch introduced a local PCM state in runtime itself and replaces those references with runtime->state. It has improvements in two aspects: - The reduction of a indirect access leads to more code optimization - It avoids a possible (unexpected) modification of the state via mmap of the status record The status->state is updated together with runtime->state, so that user-space can still read the current state via mmap like before, too. This patch touches only the ALSA core code. The changes in each driver will follow in later patches. Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220926135558.26580-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 20 +++++++- sound/core/oss/pcm_oss.c | 42 ++++++++-------- sound/core/pcm.c | 9 ++-- sound/core/pcm_compat.c | 4 +- sound/core/pcm_lib.c | 16 +++--- sound/core/pcm_native.c | 127 ++++++++++++++++++++++++----------------------- 6 files changed, 118 insertions(+), 100 deletions(-) (limited to 'sound') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 8c48a5bce88c..7b1a022910e8 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -346,6 +346,8 @@ static inline void snd_pcm_pack_audio_tstamp_report(__u32 *data, __u32 *accuracy struct snd_pcm_runtime { /* -- Status -- */ + snd_pcm_state_t state; /* stream state */ + snd_pcm_state_t suspended_state; /* suspended stream state */ struct snd_pcm_substream *trigger_master; struct timespec64 trigger_tstamp; /* trigger timestamp */ bool trigger_tstamp_latched; /* trigger timestamp latched in low-level driver/hardware */ @@ -678,11 +680,25 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, */ static inline int snd_pcm_running(struct snd_pcm_substream *substream) { - return (substream->runtime->status->state == SNDRV_PCM_STATE_RUNNING || - (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING && + return (substream->runtime->state == SNDRV_PCM_STATE_RUNNING || + (substream->runtime->state == SNDRV_PCM_STATE_DRAINING && substream->stream == SNDRV_PCM_STREAM_PLAYBACK)); } +/** + * __snd_pcm_set_state - Change the current PCM state + * @runtime: PCM runtime to set + * @state: the current state to set + * + * Call within the stream lock + */ +static inline void __snd_pcm_set_state(struct snd_pcm_runtime *runtime, + snd_pcm_state_t state) +{ + runtime->state = state; + runtime->status->state = state; /* copy for mmap */ +} + /** * bytes_to_samples - Unit conversion of the size from bytes to samples * @runtime: PCM runtime instance diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 02df915eb3c6..ac2efeb63a39 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1237,12 +1237,12 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const struct snd_pcm_runtime *runtime = substream->runtime; int ret; while (1) { - if (runtime->status->state == SNDRV_PCM_STATE_XRUN || - runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { + if (runtime->state == SNDRV_PCM_STATE_XRUN || + runtime->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "pcm_oss: write: recovering from %s\n", - runtime->status->state == SNDRV_PCM_STATE_XRUN ? + runtime->state == SNDRV_PCM_STATE_XRUN ? "XRUN" : "SUSPEND"); #endif ret = snd_pcm_oss_prepare(substream); @@ -1257,7 +1257,7 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const break; /* test, if we can't store new data, because the stream */ /* has not been started */ - if (runtime->status->state == SNDRV_PCM_STATE_PREPARED) + if (runtime->state == SNDRV_PCM_STATE_PREPARED) return -EAGAIN; } return ret; @@ -1269,18 +1269,18 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p snd_pcm_sframes_t delay; int ret; while (1) { - if (runtime->status->state == SNDRV_PCM_STATE_XRUN || - runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { + if (runtime->state == SNDRV_PCM_STATE_XRUN || + runtime->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "pcm_oss: read: recovering from %s\n", - runtime->status->state == SNDRV_PCM_STATE_XRUN ? + runtime->state == SNDRV_PCM_STATE_XRUN ? "XRUN" : "SUSPEND"); #endif ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL); if (ret < 0) break; - } else if (runtime->status->state == SNDRV_PCM_STATE_SETUP) { + } else if (runtime->state == SNDRV_PCM_STATE_SETUP) { ret = snd_pcm_oss_prepare(substream); if (ret < 0) break; @@ -1293,7 +1293,7 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p frames, in_kernel); mutex_lock(&runtime->oss.params_lock); if (ret == -EPIPE) { - if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { + if (runtime->state == SNDRV_PCM_STATE_DRAINING) { ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); if (ret < 0) break; @@ -1312,12 +1312,12 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, void struct snd_pcm_runtime *runtime = substream->runtime; int ret; while (1) { - if (runtime->status->state == SNDRV_PCM_STATE_XRUN || - runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { + if (runtime->state == SNDRV_PCM_STATE_XRUN || + runtime->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "pcm_oss: writev: recovering from %s\n", - runtime->status->state == SNDRV_PCM_STATE_XRUN ? + runtime->state == SNDRV_PCM_STATE_XRUN ? "XRUN" : "SUSPEND"); #endif ret = snd_pcm_oss_prepare(substream); @@ -1330,7 +1330,7 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, void /* test, if we can't store new data, because the stream */ /* has not been started */ - if (runtime->status->state == SNDRV_PCM_STATE_PREPARED) + if (runtime->state == SNDRV_PCM_STATE_PREPARED) return -EAGAIN; } return ret; @@ -1341,18 +1341,18 @@ snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void * struct snd_pcm_runtime *runtime = substream->runtime; int ret; while (1) { - if (runtime->status->state == SNDRV_PCM_STATE_XRUN || - runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { + if (runtime->state == SNDRV_PCM_STATE_XRUN || + runtime->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "pcm_oss: readv: recovering from %s\n", - runtime->status->state == SNDRV_PCM_STATE_XRUN ? + runtime->state == SNDRV_PCM_STATE_XRUN ? "XRUN" : "SUSPEND"); #endif ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL); if (ret < 0) break; - } else if (runtime->status->state == SNDRV_PCM_STATE_SETUP) { + } else if (runtime->state == SNDRV_PCM_STATE_SETUP) { ret = snd_pcm_oss_prepare(substream); if (ret < 0) break; @@ -1635,7 +1635,7 @@ static int snd_pcm_oss_sync1(struct snd_pcm_substream *substream, size_t size) result = 0; set_current_state(TASK_INTERRUPTIBLE); snd_pcm_stream_lock_irq(substream); - state = runtime->status->state; + state = runtime->state; snd_pcm_stream_unlock_irq(substream); if (state != SNDRV_PCM_STATE_RUNNING) { set_current_state(TASK_RUNNING); @@ -2854,8 +2854,8 @@ static __poll_t snd_pcm_oss_poll(struct file *file, poll_table * wait) struct snd_pcm_runtime *runtime = psubstream->runtime; poll_wait(file, &runtime->sleep, wait); snd_pcm_stream_lock_irq(psubstream); - if (runtime->status->state != SNDRV_PCM_STATE_DRAINING && - (runtime->status->state != SNDRV_PCM_STATE_RUNNING || + if (runtime->state != SNDRV_PCM_STATE_DRAINING && + (runtime->state != SNDRV_PCM_STATE_RUNNING || snd_pcm_oss_playback_ready(psubstream))) mask |= EPOLLOUT | EPOLLWRNORM; snd_pcm_stream_unlock_irq(psubstream); @@ -2865,7 +2865,7 @@ static __poll_t snd_pcm_oss_poll(struct file *file, poll_table * wait) snd_pcm_state_t ostate; poll_wait(file, &runtime->sleep, wait); snd_pcm_stream_lock_irq(csubstream); - ostate = runtime->status->state; + ostate = runtime->state; if (ostate != SNDRV_PCM_STATE_RUNNING || snd_pcm_oss_capture_ready(csubstream)) mask |= EPOLLIN | EPOLLRDNORM; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 82925709fa12..9d95e3731123 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -387,7 +387,7 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "closed\n"); goto unlock; } - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (runtime->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); goto unlock; } @@ -424,7 +424,7 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "closed\n"); goto unlock; } - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (runtime->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); goto unlock; } @@ -970,7 +970,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, init_waitqueue_head(&runtime->sleep); init_waitqueue_head(&runtime->tsleep); - runtime->status->state = SNDRV_PCM_STATE_OPEN; + __snd_pcm_set_state(runtime, SNDRV_PCM_STATE_OPEN); mutex_init(&runtime->buffer_mutex); atomic_set(&runtime->buffer_accessing, 0); @@ -1112,7 +1112,8 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) if (snd_pcm_running(substream)) snd_pcm_stop(substream, SNDRV_PCM_STATE_DISCONNECTED); /* to be sure, set the state unconditionally */ - substream->runtime->status->state = SNDRV_PCM_STATE_DISCONNECTED; + __snd_pcm_set_state(substream->runtime, + SNDRV_PCM_STATE_DISCONNECTED); wake_up(&substream->runtime->sleep); wake_up(&substream->runtime->tsleep); } diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 917c5b4f19d7..42c2ada8e888 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -295,7 +295,7 @@ static int snd_pcm_ioctl_xferi_compat(struct snd_pcm_substream *substream, return -ENOTTY; if (substream->stream != dir) return -EINVAL; - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; if (get_user(buf, &data32->buf) || @@ -341,7 +341,7 @@ static int snd_pcm_ioctl_xfern_compat(struct snd_pcm_substream *substream, return -ENOTTY; if (substream->stream != dir) return -EINVAL; - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; ch = substream->runtime->channels; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 40751e5aff09..8b6aeb8a78f7 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -186,7 +186,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, avail = snd_pcm_avail(substream); if (avail > runtime->avail_max) runtime->avail_max = avail; - if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { + if (runtime->state == SNDRV_PCM_STATE_DRAINING) { if (avail >= runtime->buffer_size) { snd_pcm_drain_done(substream); return -EPIPE; @@ -1911,7 +1911,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_stream_lock_irq(substream); set_current_state(TASK_INTERRUPTIBLE); - switch (runtime->status->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_SUSPENDED: err = -ESTRPIPE; goto _endloop; @@ -2099,14 +2099,14 @@ static int pcm_sanity_check(struct snd_pcm_substream *substream) runtime = substream->runtime; if (snd_BUG_ON(!substream->ops->copy_user && !runtime->dma_area)) return -EINVAL; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (runtime->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; return 0; } static int pcm_accessible_state(struct snd_pcm_runtime *runtime) { - switch (runtime->status->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_PREPARED: case SNDRV_PCM_STATE_RUNNING: case SNDRV_PCM_STATE_PAUSED: @@ -2225,7 +2225,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, goto _end_unlock; runtime->twake = runtime->control->avail_min ? : 1; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) + if (runtime->state == SNDRV_PCM_STATE_RUNNING) snd_pcm_update_hw_ptr(substream); /* @@ -2233,7 +2233,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, * thread may start capture */ if (!is_playback && - runtime->status->state == SNDRV_PCM_STATE_PREPARED && + runtime->state == SNDRV_PCM_STATE_PREPARED && size >= runtime->start_threshold) { err = snd_pcm_start(substream); if (err < 0) @@ -2247,7 +2247,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, snd_pcm_uframes_t cont; if (!avail) { if (!is_playback && - runtime->status->state == SNDRV_PCM_STATE_DRAINING) { + runtime->state == SNDRV_PCM_STATE_DRAINING) { snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); goto _end_unlock; } @@ -2303,7 +2303,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, xfer += frames; avail -= frames; if (is_playback && - runtime->status->state == SNDRV_PCM_STATE_PREPARED && + runtime->state == SNDRV_PCM_STATE_PREPARED && snd_pcm_playback_hw_avail(runtime) >= (snd_pcm_sframes_t)runtime->start_threshold) { err = snd_pcm_start(substream); if (err < 0) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ad0541e9e888..d9485b1ab719 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -595,8 +595,8 @@ static void snd_pcm_set_state(struct snd_pcm_substream *substream, snd_pcm_state_t state) { snd_pcm_stream_lock_irq(substream); - if (substream->runtime->status->state != SNDRV_PCM_STATE_DISCONNECTED) - substream->runtime->status->state = state; + if (substream->runtime->state != SNDRV_PCM_STATE_DISCONNECTED) + __snd_pcm_set_state(substream->runtime, state); snd_pcm_stream_unlock_irq(substream); } @@ -724,7 +724,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; snd_pcm_stream_lock_irq(substream); - switch (runtime->status->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_OPEN: case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: @@ -889,7 +889,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) if (result < 0) return result; snd_pcm_stream_lock_irq(substream); - switch (runtime->status->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: if (atomic_read(&substream->mmap_count)) @@ -920,7 +920,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, return -ENXIO; runtime = substream->runtime; snd_pcm_stream_lock_irq(substream); - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (runtime->state == SNDRV_PCM_STATE_OPEN) { snd_pcm_stream_unlock_irq(substream); return -EBADFD; } @@ -1013,8 +1013,8 @@ int snd_pcm_status64(struct snd_pcm_substream *substream, } else runtime->audio_tstamp_report.valid = 1; - status->state = runtime->status->state; - status->suspended_state = runtime->status->suspended_state; + status->state = runtime->state; + status->suspended_state = runtime->suspended_state; if (status->state == SNDRV_PCM_STATE_OPEN) goto _end; status->trigger_tstamp_sec = runtime->trigger_tstamp.tv_sec; @@ -1148,7 +1148,7 @@ static int snd_pcm_channel_info(struct snd_pcm_substream *substream, channel = info->channel; runtime = substream->runtime; snd_pcm_stream_lock_irq(substream); - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (runtime->state == SNDRV_PCM_STATE_OPEN) { snd_pcm_stream_unlock_irq(substream); return -EBADFD; } @@ -1411,7 +1411,7 @@ static int snd_pcm_pre_start(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; - if (runtime->status->state != SNDRV_PCM_STATE_PREPARED) + if (runtime->state != SNDRV_PCM_STATE_PREPARED) return -EBADFD; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && !snd_pcm_playback_data(substream)) @@ -1444,7 +1444,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, runtime->hw_ptr_jiffies = jiffies; runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / runtime->rate; - runtime->status->state = state; + __snd_pcm_set_state(runtime, state); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); @@ -1485,7 +1485,7 @@ static int snd_pcm_pre_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (runtime->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; runtime->trigger_master = substream; return 0; @@ -1506,9 +1506,9 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; - if (runtime->status->state != state) { + if (runtime->state != state) { snd_pcm_trigger_tstamp(substream); - runtime->status->state = state; + __snd_pcm_set_state(runtime, state); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTOP); } wake_up(&runtime->sleep); @@ -1584,9 +1584,9 @@ static int snd_pcm_pre_pause(struct snd_pcm_substream *substream, if (!(runtime->info & SNDRV_PCM_INFO_PAUSE)) return -ENOSYS; if (pause_pushed(state)) { - if (runtime->status->state != SNDRV_PCM_STATE_RUNNING) + if (runtime->state != SNDRV_PCM_STATE_RUNNING) return -EBADFD; - } else if (runtime->status->state != SNDRV_PCM_STATE_PAUSED) + } else if (runtime->state != SNDRV_PCM_STATE_PAUSED) return -EBADFD; runtime->trigger_master = substream; return 0; @@ -1628,12 +1628,12 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); if (pause_pushed(state)) { - runtime->status->state = SNDRV_PCM_STATE_PAUSED; + __snd_pcm_set_state(runtime, SNDRV_PCM_STATE_PAUSED); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MPAUSE); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } else { - runtime->status->state = SNDRV_PCM_STATE_RUNNING; + __snd_pcm_set_state(runtime, SNDRV_PCM_STATE_RUNNING); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MCONTINUE); } } @@ -1668,7 +1668,7 @@ static int snd_pcm_pre_suspend(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; - switch (runtime->status->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_SUSPENDED: return -EBUSY; /* unresumable PCM state; return -EBUSY for skipping suspend */ @@ -1699,8 +1699,9 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); - runtime->status->suspended_state = runtime->status->state; - runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; + runtime->suspended_state = runtime->state; + runtime->status->suspended_state = runtime->suspended_state; + __snd_pcm_set_state(runtime, SNDRV_PCM_STATE_SUSPENDED); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSUSPEND); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); @@ -1791,8 +1792,8 @@ static int snd_pcm_do_resume(struct snd_pcm_substream *substream, if (runtime->trigger_master != substream) return 0; /* DMA not running previously? */ - if (runtime->status->suspended_state != SNDRV_PCM_STATE_RUNNING && - (runtime->status->suspended_state != SNDRV_PCM_STATE_DRAINING || + if (runtime->suspended_state != SNDRV_PCM_STATE_RUNNING && + (runtime->suspended_state != SNDRV_PCM_STATE_DRAINING || substream->stream != SNDRV_PCM_STREAM_PLAYBACK)) return 0; return substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_RESUME); @@ -1811,7 +1812,7 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); - runtime->status->state = runtime->status->suspended_state; + __snd_pcm_set_state(runtime, runtime->suspended_state); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME); } @@ -1848,7 +1849,7 @@ static int snd_pcm_xrun(struct snd_pcm_substream *substream) int result; snd_pcm_stream_lock_irq(substream); - switch (runtime->status->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_XRUN: result = 0; /* already there */ break; @@ -1871,7 +1872,7 @@ static int snd_pcm_pre_reset(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; - switch (runtime->status->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_RUNNING: case SNDRV_PCM_STATE_PREPARED: case SNDRV_PCM_STATE_PAUSED: @@ -1933,8 +1934,8 @@ static int snd_pcm_pre_prepare(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; int f_flags = (__force int)state; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN || - runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + if (runtime->state == SNDRV_PCM_STATE_OPEN || + runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; if (snd_pcm_running(substream)) return -EBUSY; @@ -1985,7 +1986,7 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, f_flags = substream->f_flags; snd_pcm_stream_lock_irq(substream); - switch (substream->runtime->status->state) { + switch (substream->runtime->state) { case SNDRV_PCM_STATE_PAUSED: snd_pcm_pause(substream, false); fallthrough; @@ -2009,7 +2010,7 @@ static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; - switch (runtime->status->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_OPEN: case SNDRV_PCM_STATE_DISCONNECTED: case SNDRV_PCM_STATE_SUSPENDED: @@ -2024,28 +2025,28 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - switch (runtime->status->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_PREPARED: /* start playback stream if possible */ if (! snd_pcm_playback_empty(substream)) { snd_pcm_do_start(substream, SNDRV_PCM_STATE_DRAINING); snd_pcm_post_start(substream, SNDRV_PCM_STATE_DRAINING); } else { - runtime->status->state = SNDRV_PCM_STATE_SETUP; + __snd_pcm_set_state(runtime, SNDRV_PCM_STATE_SETUP); } break; case SNDRV_PCM_STATE_RUNNING: - runtime->status->state = SNDRV_PCM_STATE_DRAINING; + __snd_pcm_set_state(runtime, SNDRV_PCM_STATE_DRAINING); break; case SNDRV_PCM_STATE_XRUN: - runtime->status->state = SNDRV_PCM_STATE_SETUP; + __snd_pcm_set_state(runtime, SNDRV_PCM_STATE_SETUP); break; default: break; } } else { /* stop running stream */ - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) { + if (runtime->state == SNDRV_PCM_STATE_RUNNING) { snd_pcm_state_t new_state; new_state = snd_pcm_capture_avail(runtime) > 0 ? @@ -2055,7 +2056,7 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, } } - if (runtime->status->state == SNDRV_PCM_STATE_DRAINING && + if (runtime->state == SNDRV_PCM_STATE_DRAINING && runtime->trigger_master == substream && (runtime->hw.info & SNDRV_PCM_INFO_DRAIN_TRIGGER)) return substream->ops->trigger(substream, @@ -2096,7 +2097,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, card = substream->pcm->card; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (runtime->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; if (file) { @@ -2107,7 +2108,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, snd_pcm_stream_lock_irq(substream); /* resume pause */ - if (runtime->status->state == SNDRV_PCM_STATE_PAUSED) + if (runtime->state == SNDRV_PCM_STATE_PAUSED) snd_pcm_pause(substream, false); /* pre-start/stop - all running streams are changed to DRAINING state */ @@ -2135,7 +2136,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, if (s->stream != SNDRV_PCM_STREAM_PLAYBACK) continue; runtime = s->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { + if (runtime->state == SNDRV_PCM_STATE_DRAINING) { to_check = runtime; break; } @@ -2174,7 +2175,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, break; } if (tout == 0) { - if (substream->runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) + if (substream->runtime->state == SNDRV_PCM_STATE_SUSPENDED) result = -ESTRPIPE; else { dev_dbg(substream->pcm->card->dev, @@ -2206,13 +2207,13 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream) return -ENXIO; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN || - runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + if (runtime->state == SNDRV_PCM_STATE_OPEN || + runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; snd_pcm_stream_lock_irq(substream); /* resume pause */ - if (runtime->status->state == SNDRV_PCM_STATE_PAUSED) + if (runtime->state == SNDRV_PCM_STATE_PAUSED) snd_pcm_pause(substream, false); snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); @@ -2275,8 +2276,8 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) snd_pcm_group_init(group); down_write(&snd_pcm_link_rwsem); - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN || - substream->runtime->status->state != substream1->runtime->status->state || + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN || + substream->runtime->state != substream1->runtime->state || substream->pcm->nonatomic != substream1->pcm->nonatomic) { res = -EBADFD; goto _end; @@ -2700,7 +2701,7 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state != SNDRV_PCM_STATE_OPEN) do_hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; @@ -2904,7 +2905,7 @@ static int snd_pcm_release(struct inode *inode, struct file *file) */ static int do_pcm_hwsync(struct snd_pcm_substream *substream) { - switch (substream->runtime->status->state) { + switch (substream->runtime->state) { case SNDRV_PCM_STATE_DRAINING: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) return -EBADFD; @@ -3203,7 +3204,7 @@ static int snd_pcm_xferi_frames_ioctl(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_sframes_t result; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (runtime->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; if (put_user(0, &_xferi->result)) return -EFAULT; @@ -3226,7 +3227,7 @@ static int snd_pcm_xfern_frames_ioctl(struct snd_pcm_substream *substream, void *bufs; snd_pcm_sframes_t result; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (runtime->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; if (runtime->channels > 128) return -EINVAL; @@ -3290,7 +3291,7 @@ static int snd_pcm_common_ioctl(struct file *file, if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; - if (substream->runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + if (substream->runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; res = snd_power_wait(substream->pcm->card); @@ -3421,7 +3422,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, snd_pcm_uframes_t *frames = arg; snd_pcm_sframes_t result; - if (substream->runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + if (substream->runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; switch (cmd) { @@ -3466,8 +3467,8 @@ static ssize_t snd_pcm_read(struct file *file, char __user *buf, size_t count, if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN || - runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + if (runtime->state == SNDRV_PCM_STATE_OPEN || + runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; if (!frame_aligned(runtime, count)) return -EINVAL; @@ -3491,8 +3492,8 @@ static ssize_t snd_pcm_write(struct file *file, const char __user *buf, if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN || - runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + if (runtime->state == SNDRV_PCM_STATE_OPEN || + runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; if (!frame_aligned(runtime, count)) return -EINVAL; @@ -3518,8 +3519,8 @@ static ssize_t snd_pcm_readv(struct kiocb *iocb, struct iov_iter *to) if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN || - runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + if (runtime->state == SNDRV_PCM_STATE_OPEN || + runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; if (!iter_is_iovec(to)) return -EINVAL; @@ -3555,8 +3556,8 @@ static ssize_t snd_pcm_writev(struct kiocb *iocb, struct iov_iter *from) if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN || - runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + if (runtime->state == SNDRV_PCM_STATE_OPEN || + runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; if (!iter_is_iovec(from)) return -EINVAL; @@ -3595,7 +3596,7 @@ static __poll_t snd_pcm_poll(struct file *file, poll_table *wait) return ok | EPOLLERR; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + if (runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return ok | EPOLLERR; poll_wait(file, &runtime->sleep, wait); @@ -3603,7 +3604,7 @@ static __poll_t snd_pcm_poll(struct file *file, poll_table *wait) mask = 0; snd_pcm_stream_lock_irq(substream); avail = snd_pcm_avail(substream); - switch (runtime->status->state) { + switch (runtime->state) { case SNDRV_PCM_STATE_RUNNING: case SNDRV_PCM_STATE_PREPARED: case SNDRV_PCM_STATE_PAUSED: @@ -3874,7 +3875,7 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, return -EINVAL; } runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (runtime->state == SNDRV_PCM_STATE_OPEN) return -EBADFD; if (!(runtime->info & SNDRV_PCM_INFO_MMAP)) return -ENXIO; @@ -3911,7 +3912,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; - if (substream->runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + if (substream->runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; offset = area->vm_pgoff << PAGE_SHIFT; @@ -3949,7 +3950,7 @@ static int snd_pcm_fasync(int fd, struct file * file, int on) if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + if (runtime->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; return snd_fasync_helper(fd, file, on, &runtime->fasync); } -- cgit v1.2.3 From 1be2143fb7b19e247f7c4aa1097f85fe92c132bf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Sep 2022 15:55:49 +0200 Subject: ALSA: pcm: Make mmap status read-only The mmap status record should be read-only. Modifying it from user-space may screw up things unexpectedly, so let's clear the write bits at exposing it. Note that alsa-lib and other known user-space apps access the mmapped status only as read-only, hence this change shouldn't break the existing applications. Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220926135558.26580-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d9485b1ab719..33769ca78cc8 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3668,6 +3668,7 @@ static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file area->vm_ops = &snd_pcm_vm_ops_status; area->vm_private_data = substream; area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + area->vm_flags &= ~(VM_WRITE | VM_MAYWRITE); return 0; } -- cgit v1.2.3 From f7efa9b8a7d959813c63275b2e980de996b8e626 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Sep 2022 15:55:50 +0200 Subject: ALSA: aloop: Replace runtime->status->state reference to runtime->state The recent change in ALSA core allows drivers to get the current PCM state directly from runtime object. Replace the calls accordingly. Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220926135558.26580-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 12f12a294df5..a38e602b4fc6 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -535,7 +535,7 @@ static void copy_play_buf(struct loopback_pcm *play, /* check if playback is draining, trim the capture copy size * when our pointer is at the end of playback ring buffer */ - if (runtime->status->state == SNDRV_PCM_STATE_DRAINING && + if (runtime->state == SNDRV_PCM_STATE_DRAINING && snd_pcm_playback_hw_avail(runtime) < runtime->buffer_size) { snd_pcm_uframes_t appl_ptr, appl_ptr1, diff; appl_ptr = appl_ptr1 = runtime->control->appl_ptr; @@ -730,7 +730,7 @@ static void loopback_snd_timer_period_elapsed(struct loopback_cable *cable, if (event == SNDRV_TIMER_EVENT_MSTOP) { if (!dpcm_play || - dpcm_play->substream->runtime->status->state != + dpcm_play->substream->runtime->state != SNDRV_PCM_STATE_DRAINING) { spin_unlock_irqrestore(&cable->lock, flags); return; -- cgit v1.2.3 From 23cb0767f0544858169c02cec445d066d4e02e2b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Sep 2022 15:55:51 +0200 Subject: ALSA: firewire: Replace runtime->status->state reference to runtime->state The recent change in ALSA core allows drivers to get the current PCM state directly from runtime object. Replace the calls accordingly. Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220926135558.26580-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob_pcm.c | 4 ++-- sound/firewire/dice/dice-pcm.c | 4 ++-- sound/firewire/digi00x/digi00x-pcm.c | 4 ++-- sound/firewire/fireface/ff-pcm.c | 4 ++-- sound/firewire/fireworks/fireworks_pcm.c | 4 ++-- sound/firewire/motu/motu-pcm.c | 4 ++-- sound/firewire/oxfw/oxfw-pcm.c | 8 ++++---- sound/firewire/tascam/tascam-pcm.c | 4 ++-- 8 files changed, 18 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index f8d9a2041264..ce49eef0fcba 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -214,7 +214,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_bebob *bebob = substream->private_data; int err = 0; - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); unsigned int frames_per_buffer = params_buffer_size(hw_params); @@ -236,7 +236,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&bebob->mutex); - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state != SNDRV_PCM_STATE_OPEN) bebob->substreams_counter--; snd_bebob_stream_stop_duplex(bebob); diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index a69ca1111b03..d64366217d57 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -266,7 +266,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_dice *dice = substream->private_data; int err = 0; - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int events_per_period = params_period_size(hw_params); unsigned int events_per_buffer = params_buffer_size(hw_params); @@ -293,7 +293,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&dice->mutex); - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state != SNDRV_PCM_STATE_OPEN) --dice->substreams_counter; snd_dice_stream_stop_duplex(dice); diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index b7f6eda09f9f..3bd1575c9d9c 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -190,7 +190,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_dg00x *dg00x = substream->private_data; int err = 0; - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); unsigned int frames_per_buffer = params_buffer_size(hw_params); @@ -212,7 +212,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&dg00x->mutex); - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state != SNDRV_PCM_STATE_OPEN) --dg00x->substreams_counter; snd_dg00x_stream_stop_duplex(dg00x); diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index f978cc2fed7d..ec915671a79b 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -230,7 +230,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_ff *ff = substream->private_data; int err = 0; - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); unsigned int frames_per_buffer = params_buffer_size(hw_params); @@ -252,7 +252,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&ff->mutex); - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state != SNDRV_PCM_STATE_OPEN) --ff->substreams_counter; snd_ff_stream_stop_duplex(ff); diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index a0d5db1d8eb2..c3c21860b245 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -250,7 +250,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_efw *efw = substream->private_data; int err = 0; - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); unsigned int frames_per_buffer = params_buffer_size(hw_params); @@ -272,7 +272,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&efw->mutex); - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state != SNDRV_PCM_STATE_OPEN) --efw->substreams_counter; snd_efw_stream_stop_duplex(efw); diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 8e1437371263..d410c2efbde5 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -210,7 +210,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_motu *motu = substream->private_data; int err = 0; - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); unsigned int frames_per_buffer = params_buffer_size(hw_params); @@ -232,7 +232,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&motu->mutex); - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state != SNDRV_PCM_STATE_OPEN) --motu->substreams_counter; snd_motu_stream_stop_duplex(motu); diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 2dfa7e179cb6..5f43a0b826d2 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -239,7 +239,7 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_oxfw *oxfw = substream->private_data; int err = 0; - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int channels = params_channels(hw_params); unsigned int frames_per_period = params_period_size(hw_params); @@ -262,7 +262,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_oxfw *oxfw = substream->private_data; int err = 0; - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int channels = params_channels(hw_params); unsigned int frames_per_period = params_period_size(hw_params); @@ -286,7 +286,7 @@ static int pcm_capture_hw_free(struct snd_pcm_substream *substream) mutex_lock(&oxfw->mutex); - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state != SNDRV_PCM_STATE_OPEN) --oxfw->substreams_count; snd_oxfw_stream_stop_duplex(oxfw); @@ -301,7 +301,7 @@ static int pcm_playback_hw_free(struct snd_pcm_substream *substream) mutex_lock(&oxfw->mutex); - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state != SNDRV_PCM_STATE_OPEN) --oxfw->substreams_count; snd_oxfw_stream_stop_duplex(oxfw); diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 36c1353f2494..f6da571707ac 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -119,7 +119,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_tscm *tscm = substream->private_data; int err = 0; - if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + if (substream->runtime->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); unsigned int frames_per_buffer = params_buffer_size(hw_params); @@ -141,7 +141,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&tscm->mutex); - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->state != SNDRV_PCM_STATE_OPEN) --tscm->substreams_counter; snd_tscm_stream_stop_duplex(tscm); -- cgit v1.2.3 From 38d8be5df88539dc4a024250ab5988028e21826e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Sep 2022 15:55:52 +0200 Subject: ALSA: hda: Replace runtime->status->state reference to runtime->state The recent change in ALSA core allows drivers to get the current PCM state directly from runtime object. Replace the calls accordingly. Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220926135558.26580-6-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/hdmi_chmap.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index aad5c4bf4d34..5d8e1d944b0a 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -774,7 +774,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, substream = snd_pcm_chmap_substream(info, ctl_idx); if (!substream || !substream->runtime) return 0; /* just for avoiding error from alsactl restore */ - switch (substream->runtime->status->state) { + switch (substream->runtime->state) { case SNDRV_PCM_STATE_OPEN: case SNDRV_PCM_STATE_SETUP: break; -- cgit v1.2.3 From 7246e5c80630bb4dfdd50c7de2c38c4a91fd36fc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Sep 2022 15:55:53 +0200 Subject: ALSA: asihpi: Replace runtime->status->state reference to runtime->state The recent change in ALSA core allows drivers to get the current PCM state directly from runtime object. Replace the calls accordingly. Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220926135558.26580-7-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 5e1f9f10051b..8de43aaa10aa 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -632,7 +632,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, /*? workaround linked streams don't transition to SETUP 20070706*/ - s->runtime->status->state = SNDRV_PCM_STATE_SETUP; + __snd_pcm_set_state(s->runtime, SNDRV_PCM_STATE_SETUP); if (card->support_grouping) { snd_printdd("%d group\n", s->number); -- cgit v1.2.3 From d8b4efeeb37ae5e221be6b265a5b93f54c242e82 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Sep 2022 15:55:54 +0200 Subject: ALSA: usb-audio: Replace runtime->status->state reference to runtime->state The recent change in ALSA core allows drivers to get the current PCM state directly from runtime object. Replace the calls accordingly. Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220926135558.26580-8-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e721fc12acde..8ed165f036a0 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1395,7 +1395,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, transfer_done = subs->transfer_done; if (subs->lowlatency_playback && - runtime->status->state != SNDRV_PCM_STATE_DRAINING) { + runtime->state != SNDRV_PCM_STATE_DRAINING) { unsigned int hwptr = subs->hwptr_done / stride; /* calculate the byte offset-in-buffer of the appl_ptr */ @@ -1583,7 +1583,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea return 0; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: - stop_endpoints(subs, substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING); + stop_endpoints(subs, substream->runtime->state == SNDRV_PCM_STATE_DRAINING); snd_usb_endpoint_set_callback(subs->data_endpoint, NULL, NULL, NULL); subs->running = 0; -- cgit v1.2.3 From ca4833c5a21bf419fe505e9798bbf49cbd482e8f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Sep 2022 15:55:55 +0200 Subject: ALSA: usx2y: Replace runtime->status->state reference to runtime->state The recent change in ALSA core allows drivers to get the current PCM state directly from runtime object. Replace the calls accordingly. Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20220926135558.26580-9-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usbusx2yaudio.c | 3 +-- sound/usb/usx2y/usx2yhwdeppcm.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 9cd5e3aae4f7..5197599e7aa6 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -822,8 +822,7 @@ static int snd_usx2y_pcm_hw_free(struct snd_pcm_substream *substream) usx2y_urbs_release(subs); if (!cap_subs->pcm_substream || !cap_subs->pcm_substream->runtime || - !cap_subs->pcm_substream->runtime->status || - cap_subs->pcm_substream->runtime->status->state < SNDRV_PCM_STATE_PREPARED) { + cap_subs->pcm_substream->runtime->state < SNDRV_PCM_STATE_PREPARED) { atomic_set(&cap_subs->state, STATE_STOPPED); usx2y_urbs_release(cap_subs); } diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 240349b644f3..767a227d54da 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -374,8 +374,7 @@ static int snd_usx2y_usbpcm_hw_free(struct snd_pcm_substream *substream) usx2y_usbpcm_urbs_release(subs); if (!cap_subs->pcm_substream || !cap_subs->pcm_substream->runtime || - !cap_subs->pcm_substream->runtime->status || - cap_subs->pcm_substream->runtime->status->state < SNDRV_PCM_STATE_PREPARED) { + cap_subs->pcm_substream->runtime->state < SNDRV_PCM_STATE_PREPARED) { atomic_set(&cap_subs->state, STATE_STOPPED); if (cap_subs2) atomic_set(&cap_subs2->state, STATE_STOPPED); -- cgit v1.2.3 From 2bd2dc2672b2d0d45be371430970084330879a46 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Sep 2022 15:55:56 +0200 Subject: ASoC: intel: Replace runtime->status->state reference to runtime->state The recent change in ALSA core allows drivers to get the current PCM state directly from runtime object. Replace the calls accordingly. Reviewed-by: Jaroslav Kysela Acked-by: Mark Brown Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20220926135558.26580-10-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/intel/skylake/skl-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 9d72ebd812af..1015716f9336 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -275,7 +275,7 @@ static int skl_pcm_prepare(struct snd_pcm_substream *substream, * calls prepare another time, reset the FW pipe to clean state */ if (mconfig && - (substream->runtime->status->state == SNDRV_PCM_STATE_XRUN || + (substream->runtime->state == SNDRV_PCM_STATE_XRUN || mconfig->pipe->state == SKL_PIPE_CREATED || mconfig->pipe->state == SKL_PIPE_PAUSED)) { @@ -593,7 +593,7 @@ static int skl_link_pcm_prepare(struct snd_pcm_substream *substream, /* In case of XRUN recovery, reset the FW pipe to clean state */ mconfig = skl_tplg_be_get_cpr_module(dai, substream->stream); if (mconfig && !mconfig->pipe->passthru && - (substream->runtime->status->state == SNDRV_PCM_STATE_XRUN)) + (substream->runtime->state == SNDRV_PCM_STATE_XRUN)) skl_reset_pipe(skl, mconfig->pipe); return 0; -- cgit v1.2.3 From a267fdd0a6ce3edd6419b10ee7bcde61aa15eb43 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 26 Sep 2022 15:55:57 +0200 Subject: ASoC: sh: Replace runtime->status->state reference to runtime->state The recent change in ALSA core allows drivers to get the current PCM state directly from runtime object. Replace the calls accordingly. Reviewed-by: Jaroslav Kysela Acked-by: Mark Brown Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20220926135558.26580-11-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/sh/rz-ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 7ace0c0db5b1..5d6bae33ae34 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -598,7 +598,7 @@ static int rz_ssi_dma_transfer(struct rz_ssi_priv *ssi, return -EINVAL; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) + if (runtime->state == SNDRV_PCM_STATE_DRAINING) /* * Stream is ending, so do not queue up any more DMA * transfers otherwise we play partial sound clips -- cgit v1.2.3 From d1c442019594692c64a70a86ad88eb5b6db92216 Mon Sep 17 00:00:00 2001 From: Andreas Pape Date: Mon, 26 Sep 2022 18:58:13 +0200 Subject: ALSA: dmaengine: increment buffer pointer atomically Setting pointer and afterwards checking for wraparound leads to the possibility of returning the inconsistent pointer position. This patch increments buffer pointer atomically to avoid this issue. Fixes: e7f73a1613567a ("ASoC: Add dmaengine PCM helper functions") Signed-off-by: Andreas Pape Signed-off-by: Eugeniu Rosca Link: https://lore.kernel.org/r/1664211493-11789-1-git-send-email-erosca@de.adit-jv.com Signed-off-by: Takashi Iwai --- sound/core/pcm_dmaengine.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 5b2ca028f5aa..494ec0c207fa 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -133,12 +133,14 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); static void dmaengine_pcm_dma_complete(void *arg) { + unsigned int new_pos; struct snd_pcm_substream *substream = arg; struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - prtd->pos += snd_pcm_lib_period_bytes(substream); - if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) - prtd->pos = 0; + new_pos = prtd->pos + snd_pcm_lib_period_bytes(substream); + if (new_pos >= snd_pcm_lib_buffer_bytes(substream)) + new_pos = 0; + prtd->pos = new_pos; snd_pcm_period_elapsed(substream); } -- cgit v1.2.3 From eefe77fdc0de86480f5dbb6bc721396d82d095d3 Mon Sep 17 00:00:00 2001 From: Shang XiaoJing Date: Tue, 27 Sep 2022 22:11:10 +0800 Subject: ALSA: sb: Use DIV_ROUND_UP() instead of open-coding it Use DIV_ROUND_UP() instead of open-coding it, which intents and makes it more clear what is going on for the casual reviewer. The Coccinelle references Commit e4d8aef21403 ("ALSA: usb: Use DIV_ROUND_UP() instead of open-coding it"). Signed-off-by: Shang XiaoJing Link: https://lore.kernel.org/r/20220927141110.18033-1-shangxiaojing@huawei.com Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index f8d90a1e989b..c8afc4347c54 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -236,7 +236,7 @@ static int emu8k_pcm_open(struct snd_pcm_substream *subs) /* use timer to update periods.. (specified in msec) */ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - (1000000 + HZ - 1) / HZ, UINT_MAX); + DIV_ROUND_UP(1000000, HZ), UINT_MAX); return 0; } -- cgit v1.2.3 From 2d6bd853cabc40f9fa7bf48edaa728e19f335e48 Mon Sep 17 00:00:00 2001 From: Yuan Can Date: Wed, 28 Sep 2022 08:48:33 +0000 Subject: ALSA: asihpi - Remove unused struct hpi_subsys_response After commit 3285ea10e9b0("ALSA: asihpi - Interrelated HPI tidy up."), struct hpi_subsys_response is not used any more and can be removed as well. Signed-off-by: Yuan Can Link: https://lore.kernel.org/r/20220928084833.61131-1-yuancan@huawei.com Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpimsgx.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c index f7427f8eb630..d0caef299481 100644 --- a/sound/pci/asihpi/hpimsgx.c +++ b/sound/pci/asihpi/hpimsgx.c @@ -93,11 +93,6 @@ static void HPIMSGX__cleanup(u16 adapter_index, void *h_owner); #pragma pack(push, 1) #endif -struct hpi_subsys_response { - struct hpi_response_header h; - struct hpi_subsys_res s; -}; - struct hpi_adapter_response { struct hpi_response_header h; struct hpi_adapter_res a; -- cgit v1.2.3 From 225f6e1bc151978041595c7d2acaded3aac41f54 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 29 Sep 2022 08:14:55 +0200 Subject: ALSA: hda/realtek: Add quirk for HP Zbook Firefly 14 G9 model HP Zbook Firefly 14 G9 model (103c:8abb) requires yet another binding with CS35L41 codec, but with a slightly different configuration. It's over spi1 instead of spi0. Create a new fixup entry for that. Cc: Link: https://lore.kernel.org/r/20220929061455.13355-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f9d46ae4c7b7..3dc19174670e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6741,6 +6741,11 @@ static void cs35l41_fixup_spi_two(struct hda_codec *codec, const struct hda_fixu cs35l41_generic_fixup(codec, action, "spi0", "CSC3551", 2); } +static void cs35l41_fixup_spi1_two(struct hda_codec *codec, const struct hda_fixup *fix, int action) +{ + cs35l41_generic_fixup(codec, action, "spi1", "CSC3551", 2); +} + static void cs35l41_fixup_spi_four(struct hda_codec *codec, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(codec, action, "spi0", "CSC3551", 4); @@ -7132,6 +7137,8 @@ enum { ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED, ALC245_FIXUP_CS35L41_SPI_2, ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED, + ALC245_FIXUP_CS35L41_SPI1_2, + ALC245_FIXUP_CS35L41_SPI1_2_HP_GPIO_LED, ALC245_FIXUP_CS35L41_SPI_4, ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED, ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED, @@ -8979,6 +8986,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_GPIO_LED, }, + [ALC245_FIXUP_CS35L41_SPI1_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_spi1_two, + }, + [ALC245_FIXUP_CS35L41_SPI1_2_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_spi1_two, + .chained = true, + .chain_id = ALC285_FIXUP_HP_GPIO_LED, + }, [ALC245_FIXUP_CS35L41_SPI_4] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_spi_four, @@ -9341,6 +9358,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8abb, "HP ZBook Firefly 14 G9", ALC245_FIXUP_CS35L41_SPI1_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), -- cgit v1.2.3 From 35a1744423743247026668e2323d1b932583fc2a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 Sep 2022 10:48:10 +0200 Subject: ALSA: hda/realtek: More robust component matching for CS35L41 As the previous commit implies, a system may have a different SPI bus number that is embedded in the device string. And, assuming the fixed bus number is rather fragile; it may be assigned differently depending on the configuration or on the boot environment. Once when a bus number change happens, the binding fails, resulting in the silence. This patch tries to make the matching a bit more relaxed, allowing to bind with a different bus number (or without it). So the previous fix, the introduction of ALC245_FIXUP_CS35L41_SPI1_2 fixup became superfluous, and this is unified to ALC245_FIXUP_CS35L41_SPI_2. Fixes: 225f6e1bc151 ("ALSA: hda/realtek: Add quirk for HP Zbook Firefly 14 G9 model") Link: https://lore.kernel.org/r/20220930084810.10435-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 62 ++++++++++++++++++++++++++----------------- 1 file changed, 37 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3dc19174670e..bce82b834cec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -6704,23 +6705,51 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ } } +struct cs35l41_dev_name { + const char *bus; + const char *hid; + int index; +}; + +/* match the device name in a slightly relaxed manner */ +static int comp_match_cs35l41_dev_name(struct device *dev, void *data) +{ + struct cs35l41_dev_name *p = data; + const char *d = dev_name(dev); + int n = strlen(p->bus); + char tmp[32]; + + /* check the bus name */ + if (strncmp(d, p->bus, n)) + return 0; + /* skip the bus number */ + if (isdigit(d[n])) + n++; + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), "-%s:00-cs35l41-hda.%d", p->hid, p->index); + return !strcmp(d + n, tmp); +} + static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, const char *hid, int count) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - char *name; + struct cs35l41_dev_name *rec; int ret, i; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: for (i = 0; i < count; i++) { - name = devm_kasprintf(dev, GFP_KERNEL, - "%s-%s:00-cs35l41-hda.%d", bus, hid, i); - if (!name) + rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); + if (!rec) return; + rec->bus = bus; + rec->hid = hid; + rec->index = i; spec->comps[i].codec = cdc; - component_match_add(dev, &spec->match, component_compare_dev_name, name); + component_match_add(dev, &spec->match, + comp_match_cs35l41_dev_name, rec); } ret = component_master_add_with_match(dev, &comp_master_ops, spec->match); if (ret) @@ -6738,17 +6767,12 @@ static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup static void cs35l41_fixup_spi_two(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - cs35l41_generic_fixup(codec, action, "spi0", "CSC3551", 2); -} - -static void cs35l41_fixup_spi1_two(struct hda_codec *codec, const struct hda_fixup *fix, int action) -{ - cs35l41_generic_fixup(codec, action, "spi1", "CSC3551", 2); + cs35l41_generic_fixup(codec, action, "spi", "CSC3551", 2); } static void cs35l41_fixup_spi_four(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - cs35l41_generic_fixup(codec, action, "spi0", "CSC3551", 4); + cs35l41_generic_fixup(codec, action, "spi", "CSC3551", 4); } static void alc287_fixup_legion_16achg6_speakers(struct hda_codec *cdc, const struct hda_fixup *fix, @@ -7137,8 +7161,6 @@ enum { ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED, ALC245_FIXUP_CS35L41_SPI_2, ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED, - ALC245_FIXUP_CS35L41_SPI1_2, - ALC245_FIXUP_CS35L41_SPI1_2_HP_GPIO_LED, ALC245_FIXUP_CS35L41_SPI_4, ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED, ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED, @@ -8986,16 +9008,6 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_GPIO_LED, }, - [ALC245_FIXUP_CS35L41_SPI1_2] = { - .type = HDA_FIXUP_FUNC, - .v.func = cs35l41_fixup_spi1_two, - }, - [ALC245_FIXUP_CS35L41_SPI1_2_HP_GPIO_LED] = { - .type = HDA_FIXUP_FUNC, - .v.func = cs35l41_fixup_spi1_two, - .chained = true, - .chain_id = ALC285_FIXUP_HP_GPIO_LED, - }, [ALC245_FIXUP_CS35L41_SPI_4] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_spi_four, @@ -9358,7 +9370,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8abb, "HP ZBook Firefly 14 G9", ALC245_FIXUP_CS35L41_SPI1_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8abb, "HP ZBook Firefly 14 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), -- cgit v1.2.3 From 568be8aaf8a535f79c4db76cabe17b035aa2584d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 Sep 2022 12:01:29 +0200 Subject: ALSA: usb-audio: Fix NULL dererence at error path At an error path to release URB buffers and contexts, the driver might hit a NULL dererence for u->urb pointer, when u->buffer_size has been already set but the actual URB allocation failed. Fix it by adding the NULL check of urb. Also, make sure that buffer_size is cleared after the error path or the close. Cc: Reported-by: Sabri N. Ferreiro Link: https://lore.kernel.org/r/CAKG+3NRjTey+fFfUEGwuxL-pi_=T4cUskYG9OzpzHytF+tzYng@mail.gmail.com Link: https://lore.kernel.org/r/20220930100129.19445-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index b2d0b42b581f..36f753a28341 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -95,12 +95,13 @@ static inline unsigned get_usb_high_speed_rate(unsigned int rate) */ static void release_urb_ctx(struct snd_urb_ctx *u) { - if (u->buffer_size) + if (u->urb && u->buffer_size) usb_free_coherent(u->ep->chip->dev, u->buffer_size, u->urb->transfer_buffer, u->urb->transfer_dma); usb_free_urb(u->urb); u->urb = NULL; + u->buffer_size = 0; } static const char *usb_error_string(int err) -- cgit v1.2.3 From 6382da0828995af87aa8b8bef28cc61aceb4aff3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 Sep 2022 12:01:51 +0200 Subject: ALSA: usb-audio: Fix potential memory leaks When the driver hits -ENOMEM at allocating a URB or a buffer, it aborts and goes to the error path that releases the all previously allocated resources. However, when -ENOMEM hits at the middle of the sync EP URB allocation loop, the partially allocated URBs might be left without released, because ep->nurbs is still zero at that point. Fix it by setting ep->nurbs at first, so that the error handler loops over the full URB list. Cc: Link: https://lore.kernel.org/r/20220930100151.19461-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 36f753a28341..48a3843a08f1 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1268,6 +1268,7 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep) if (!ep->syncbuf) return -ENOMEM; + ep->nurbs = SYNC_URBS; for (i = 0; i < SYNC_URBS; i++) { struct snd_urb_ctx *u = &ep->urb[i]; u->index = i; @@ -1287,8 +1288,6 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep) u->urb->complete = snd_complete_urb; } - ep->nurbs = SYNC_URBS; - return 0; out_of_memory: -- cgit v1.2.3 From 5226c7b9784eee215e3914f440b3c2e1764f67a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 1 Oct 2022 09:48:10 +0200 Subject: ALSA: hda/hdmi: Don't skip notification handling during PM operation The HDMI driver skips the notification handling from the graphics driver when the codec driver is being in the PM operation. This behavior was introduced by the commit eb399d3c99d8 ("ALSA: hda - Skip ELD notification during PM process"). This skip may cause a problem, as we may miss the ELD update when the connection/disconnection happens right at the runtime-PM operation of the audio codec. Although this workaround was valid at that time, it's no longer true; the fix was required just because the ELD update procedure needed to wake up the audio codec, which had lead to a runtime-resume during a runtime-suspend. Meanwhile, the ELD update procedure doesn't need a codec wake up any longer since the commit 788d441a164c ("ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling"); i.e. there is no much reason for skipping the notification. Let's drop those checks for addressing the missing notification. Fixes: 788d441a164c ("ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling") Reported-by: Brent Lu Link: https://lore.kernel.org/r/20220927135807.4097052-1-brent.lu@intel.com Link: https://lore.kernel.org/r/20221001074809.7461-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c172640c8a41..21edf7a619f0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2666,9 +2666,6 @@ static void generic_acomp_pin_eld_notify(void *audio_ptr, int port, int dev_id) */ if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; - /* ditto during suspend/resume process itself */ - if (snd_hdac_is_in_pm(&codec->core)) - return; check_presence_and_report(codec, pin_nid, dev_id); } @@ -2852,9 +2849,6 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) */ if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; - /* ditto during suspend/resume process itself */ - if (snd_hdac_is_in_pm(&codec->core)) - return; snd_hdac_i915_set_bclk(&codec->bus->core); check_presence_and_report(codec, pin_nid, dev_id); -- cgit v1.2.3 From 56e696c0f0c71b77fff921fc94b58a02f0445b2c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 1 Oct 2022 16:21:24 +0200 Subject: ALSA: hda: Fix position reporting on Poulsbo Hans reported that his Sony VAIO VPX11S1E showed the broken sound behavior at the start of the stream for a couple of seconds, and it turned out that the position_fix=1 option fixes the issue. It implies that the position reporting is inaccurate, and very likely hitting on all Poulsbo devices. The patch applies the workaround for Poulsbo generically to switch to LPIB mode instead of the default position buffer. Reported-and-tested-by: Hans de Goede Cc: Link: https://lore.kernel.org/r/3e8697e1-87c6-7a7b-d2e8-b21f1d2f181b@redhat.com Link: https://lore.kernel.org/r/20221001142124.7241-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2a93bc64c2d8..6ff19dd0d10c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2547,7 +2547,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Poulsbo */ { PCI_DEVICE(0x8086, 0x811b), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_BASE }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_BASE | + AZX_DCAPS_POSFIX_LPIB }, /* Oaktrail */ { PCI_DEVICE(0x8086, 0x080a), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_BASE }, -- cgit v1.2.3