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

[PATCH v2] MAINTAINERS: update Tzung-Bi's email address
by Tzung-Bi Shih 26 Oct '22

26 Oct '22

Use kernel.org account instead. Signed-off-by: Tzung-Bi Shih <tzungbi(a)kernel.org> --- Changes from v1: - Also update .mailmap. .mailmap | 1 + .../devicetree/bindings/sound/google,cros-ec-codec.yaml | 2 +- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 2 +- MAINTAINERS | 2 +- 4 files changed, 4 insertions(+), 3 deletions(-) diff --git a/.mailmap b/.mailmap index 380378e2db36..84342d781407 100644 --- a/.mailmap +++ b/.mailmap @@ -414,6 +414,7 @@ TripleX Chung <xxx.phy(a)gmail.com> <triplex(a)zh-kernel.org> TripleX Chung <xxx.phy(a)gmail.com> <zhongyu(a)18mail.cn> Tsuneo Yoshioka <Tsuneo.Yoshioka(a)f-secure.com> Tycho Andersen <tycho(a)tycho.pizza> <tycho(a)tycho.ws> +Tzung-Bi Shih <tzungbi(a)kernel.org> <tzungbi(a)google.com> Uwe Kleine-König <ukleinek(a)informatik.uni-freiburg.de> Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Uwe Kleine-König <ukleinek(a)strlen.de> diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index c3e9f3485449..dea293f403d9 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -8,7 +8,7 @@ title: Audio codec controlled by ChromeOS EC maintainers: - Cheng-Yi Chiang <cychiang(a)chromium.org> - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Google's ChromeOS EC codec is a digital mic codec provided by the diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 1d73204451b1..ea7d4900ee4a 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Realtek rt1015p codec devicetree bindings maintainers: - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Rt1015p is a rt1015 variant which does not support I2C and diff --git a/MAINTAINERS b/MAINTAINERS index cf0f18502372..f9749afc0b9d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4906,7 +4906,7 @@ F: drivers/platform/chrome/ CHROMEOS EC CODEC DRIVER M: Cheng-Yi Chiang <cychiang(a)chromium.org> -M: Tzung-Bi Shih <tzungbi(a)google.com> +M: Tzung-Bi Shih <tzungbi(a)kernel.org> R: Guenter Roeck <groeck(a)chromium.org> L: chrome-platform(a)lists.linux.dev S: Maintained -- 2.38.0.135.g90850a2211-goog

2 1

[PATCH v2 0/9] ASoC: Intel: avs: PCM power management
by Cezary Rojewski 26 Oct '22

26 Oct '22

Goal of the series is implementation of suspend/resume operations for a PCM stream along with all the collaterals connected to the subject. Start with splitting avs_dai_fe_hw_free() as ideally we would like to reuse as much of existing code as possible but snd_pcm_lib_free_pages() is not desired part of the function when speaking of suspend operation. The actual implementation of suspend/resume() for component drivers follows. For most scenarios, the PM flow is similar to standard streaming one, except for the part where the position register are being saved and the lack of PCM pages freeing. To reduce code duplication, all avs_dai_suspend_XXX() and avs_dai_resume_XXX() functions reuse their non-PM equivalents. Order of operations is affected by the fact that path binding/unbinding happens only in FE part of the stream. Above essentially unlocks SX+streaming scenarios i.e.: power transitions with an ongoing stream. As some streams are allowed to run in low power state, support is provided for S0iX state. The handlers check ACPI capabilities and the number of active low-power paths before deciding between SX and S0iX flows. The last portion of the patchset is addition of power/clock gating overrides. There is no single set of registers that ensures AudioDSP firmware loads 100% of time on every single configuration. By having them exposed, user can have the loading procedure behavior adjusted for their configuration without having to recompile the kernel. Changes in v2: - moved DRSM reg polling to separate function, snd_hdac_stream_wait_drsm() as suggested by Pierre - moved the acpi_gbl_FADT under AVS_S0IX_SUPPORTED macro and relocated the checks from pcm.c to topology.c so that the low-power streams are filtered as early as possible - fixed compilation when CONFIG_ACPI is disabled as reported by the test robot - simplified all the avs_dai_resume/suspend_xxx() by moving ->ignore_suspend check one level up, to avs_component_pm_op() (patch 0002) Amadeusz Sławiński (1): ASoC: Intel: avs: Handle SUSPEND and RESUME triggers Cezary Rojewski (7): ASoC: Intel: avs: Split pcm pages freeing operation from hw_free() ASoC: Intel: avs: Introduce PCM power management routines ALSA: hda: Introduce snd_hdac_stream_wait_drsm() ASoC: Intel: avs: Restart instead of resuming HDA capture streams ASoC: Intel: avs: Count low power streams ASoC: Intel: avs: Power and clock gating policy overriding ASoC: Intel: avs: Enact power gating policy Piotr Maziarz (1): ASoC: Intel: avs: Standby power-state support include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 5 + sound/hda/hdac_stream.c | 21 ++ sound/soc/intel/avs/avs.h | 8 + sound/soc/intel/avs/core.c | 97 +++++++-- sound/soc/intel/avs/loader.c | 10 + sound/soc/intel/avs/pcm.c | 373 +++++++++++++++++++++++++++++---- sound/soc/intel/avs/topology.c | 10 + 8 files changed, 465 insertions(+), 60 deletions(-) -- 2.25.1

2 10

[PATCH] ALSA/ASoC: replace ternary operator with min()
by wangkailong＠jari.cn 26 Oct '22

26 Oct '22

Fix the following coccicheck warning: sound/soc/soc-ops.c:817: WARNING opportunity for min() sound/core/vmaster.c:73: WARNING opportunity for min() sound/pci/hda/hda_codec.c:337: WARNING opportunity for min() sound/pci/ctxfi/ctatc.c:448: WARNING opportunity for min() sound/pci/ctxfi/ctatc.c:387: WARNING opportunity for min() Signed-off-by: KaiLong Wang <wangkailong(a)jari.cn> --- sound/core/vmaster.c | 2 +- sound/pci/ctxfi/ctatc.c | 4 ++-- sound/pci/hda/hda_codec.c | 2 +- sound/soc/soc-ops.c | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index d0f11f37889b..704a09f4bfd6 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -70,7 +70,7 @@ static int follower_update(struct link_follower *follower) follower->vals[ch] = uctl->value.integer.value[ch]; error: kfree(uctl); - return err < 0 ? err : 0; + return min(err, 0); } /* get the follower ctl info and save the initial values */ diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index fbdb8a3d5b8e..9fea50b72cfb 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -384,7 +384,7 @@ static int atc_pcm_playback_start(struct ct_atc *atc, struct ct_atc_pcm *apcm) apcm->started = 1; max_cisz = src->multi * src->rsc.msr; - max_cisz = 0x80 * (max_cisz < 8 ? max_cisz : 8); + max_cisz = 0x80 * min(max_cisz, 8); src->ops->set_sa(src, apcm->vm_block->addr); src->ops->set_la(src, apcm->vm_block->addr + apcm->vm_block->size); @@ -445,7 +445,7 @@ atc_pcm_playback_position(struct ct_atc *atc, struct ct_atc_pcm *apcm) size = apcm->vm_block->size; max_cisz = src->multi * src->rsc.msr; - max_cisz = 128 * (max_cisz < 8 ? max_cisz : 8); + max_cisz = 128 * min(max_cisz, 8); return (position + size - max_cisz - apcm->vm_block->addr) % size; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b4d1e658c556..c195f99bd8d5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -334,7 +334,7 @@ int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, return 0; dev_len = parm + 1; - dev_len = dev_len < max_devices ? dev_len : max_devices; + dev_len = min(dev_len, max_devices); devices = 0; while (devices < dev_len) { diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index bd88de056358..d71d10055ed7 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -814,7 +814,7 @@ int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { struct soc_bytes_ext *params = (void *)kcontrol->private_value; - unsigned int count = size < params->max ? size : params->max; + unsigned int count = min_t(unsigned int, size, params->max); int ret = -ENXIO; switch (op_flag) { -- 2.25.1

3 3

[PATCH AUTOSEL 6.0 07/44] ALSA: hda: Fix page fault in snd_hda_codec_shutdown()
by Sasha Levin 25 Oct '22

25 Oct '22

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit f2bd1c5ae2cb0cf9525c9bffc0038c12dd7e1338 ] 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 <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220816111727.3218543-7-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_codec.c | 41 +++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 384426d7e9dd..4ae8b9574778 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -931,8 +931,28 @@ 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); @@ -985,29 +1005,8 @@ 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; -- 2.35.1

3 3

[PATCH] ALSA: rme9652: use explicitly signed char
by Jason A. Donenfeld 25 Oct '22

25 Oct '22

With char becoming unsigned by default, and with `char` alone being ambiguous and based on architecture, signed chars need to be marked explicitly as such. This fixes warnings like: sound/pci/rme9652/hdsp.c:3953 hdsp_channel_buffer_location() warn: 'hdsp->channel_map[channel]' is unsigned sound/pci/rme9652/hdsp.c:4153 snd_hdsp_channel_info() warn: impossible condition '(hdsp->channel_map[channel] < 0) => (0-255 < 0)' sound/pci/rme9652/rme9652.c:1833 rme9652_channel_buffer_location() warn: 'rme9652->channel_map[channel]' is unsigned Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Jason A. Donenfeld <Jason(a)zx2c4.com> --- sound/pci/rme9652/hdsp.c | 26 +++++++++++++------------- sound/pci/rme9652/rme9652.c | 22 +++++++++++----------- 2 files changed, 24 insertions(+), 24 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index dcc43a81ae0e..902508e4fecc 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -433,7 +433,7 @@ struct hdsp_midi { struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *input; struct snd_rawmidi_substream *output; - char istimer; /* timer in use */ + bool istimer; /* timer in use */ struct timer_list timer; spinlock_t lock; int pending; @@ -480,7 +480,7 @@ struct hdsp { pid_t playback_pid; int running; int system_sample_rate; - const char *channel_map; + const signed char *channel_map; int dev; int irq; unsigned long port; @@ -502,7 +502,7 @@ struct hdsp { where the data for that channel can be read/written from/to. */ -static const char channel_map_df_ss[HDSP_MAX_CHANNELS] = { +static const signed char channel_map_df_ss[HDSP_MAX_CHANNELS] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25 }; @@ -517,7 +517,7 @@ static const char channel_map_mf_ss[HDSP_MAX_CHANNELS] = { /* Multiface */ -1, -1, -1, -1, -1, -1, -1, -1 }; -static const char channel_map_ds[HDSP_MAX_CHANNELS] = { +static const signed char channel_map_ds[HDSP_MAX_CHANNELS] = { /* ADAT channels are remapped */ 1, 3, 5, 7, 9, 11, 13, 15, 17, 19, 21, 23, /* channels 12 and 13 are S/PDIF */ @@ -526,7 +526,7 @@ static const char channel_map_ds[HDSP_MAX_CHANNELS] = { -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; -static const char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = { +static const signed char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = { /* ADAT channels */ 0, 1, 2, 3, 4, 5, 6, 7, /* SPDIF */ @@ -540,7 +540,7 @@ static const char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = { -1, -1 }; -static const char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = { +static const signed char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = { /* ADAT */ 1, 3, 5, 7, /* SPDIF */ @@ -554,7 +554,7 @@ static const char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = { -1, -1, -1, -1, -1, -1 }; -static const char channel_map_H9632_qs[HDSP_MAX_CHANNELS] = { +static const signed char channel_map_H9632_qs[HDSP_MAX_CHANNELS] = { /* ADAT is disabled in this mode */ /* SPDIF */ 8, 9, @@ -3939,7 +3939,7 @@ static snd_pcm_uframes_t snd_hdsp_hw_pointer(struct snd_pcm_substream *substream return hdsp_hw_pointer(hdsp); } -static char *hdsp_channel_buffer_location(struct hdsp *hdsp, +static signed char *hdsp_channel_buffer_location(struct hdsp *hdsp, int stream, int channel) @@ -3964,7 +3964,7 @@ static int snd_hdsp_playback_copy(struct snd_pcm_substream *substream, void __user *src, unsigned long count) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; if (snd_BUG_ON(pos + count > HDSP_CHANNEL_BUFFER_BYTES)) return -EINVAL; @@ -3982,7 +3982,7 @@ static int snd_hdsp_playback_copy_kernel(struct snd_pcm_substream *substream, void *src, unsigned long count) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = hdsp_channel_buffer_location(hdsp, substream->pstr->stream, channel); if (snd_BUG_ON(!channel_buf)) @@ -3996,7 +3996,7 @@ static int snd_hdsp_capture_copy(struct snd_pcm_substream *substream, void __user *dst, unsigned long count) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; if (snd_BUG_ON(pos + count > HDSP_CHANNEL_BUFFER_BYTES)) return -EINVAL; @@ -4014,7 +4014,7 @@ static int snd_hdsp_capture_copy_kernel(struct snd_pcm_substream *substream, void *dst, unsigned long count) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = hdsp_channel_buffer_location(hdsp, substream->pstr->stream, channel); if (snd_BUG_ON(!channel_buf)) @@ -4028,7 +4028,7 @@ static int snd_hdsp_hw_silence(struct snd_pcm_substream *substream, unsigned long count) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = hdsp_channel_buffer_location (hdsp, substream->pstr->stream, channel); if (snd_BUG_ON(!channel_buf)) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 1d614fe89a6a..e7c320afefe8 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -230,7 +230,7 @@ struct snd_rme9652 { int last_spdif_sample_rate; /* so that we can catch externally ... */ int last_adat_sample_rate; /* ... induced rate changes */ - const char *channel_map; + const signed char *channel_map; struct snd_card *card; struct snd_pcm *pcm; @@ -247,12 +247,12 @@ struct snd_rme9652 { where the data for that channel can be read/written from/to. */ -static const char channel_map_9652_ss[26] = { +static const signed char channel_map_9652_ss[26] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25 }; -static const char channel_map_9636_ss[26] = { +static const signed char channel_map_9636_ss[26] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, /* channels 16 and 17 are S/PDIF */ 24, 25, @@ -260,7 +260,7 @@ static const char channel_map_9636_ss[26] = { -1, -1, -1, -1, -1, -1, -1, -1 }; -static const char channel_map_9652_ds[26] = { +static const signed char channel_map_9652_ds[26] = { /* ADAT channels are remapped */ 1, 3, 5, 7, 9, 11, 13, 15, 17, 19, 21, 23, /* channels 12 and 13 are S/PDIF */ @@ -269,7 +269,7 @@ static const char channel_map_9652_ds[26] = { -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; -static const char channel_map_9636_ds[26] = { +static const signed char channel_map_9636_ds[26] = { /* ADAT channels are remapped */ 1, 3, 5, 7, 9, 11, 13, 15, /* channels 8 and 9 are S/PDIF */ @@ -1819,7 +1819,7 @@ static snd_pcm_uframes_t snd_rme9652_hw_pointer(struct snd_pcm_substream *substr return rme9652_hw_pointer(rme9652); } -static char *rme9652_channel_buffer_location(struct snd_rme9652 *rme9652, +static signed char *rme9652_channel_buffer_location(struct snd_rme9652 *rme9652, int stream, int channel) @@ -1847,7 +1847,7 @@ static int snd_rme9652_playback_copy(struct snd_pcm_substream *substream, void __user *src, unsigned long count) { struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; if (snd_BUG_ON(pos + count > RME9652_CHANNEL_BUFFER_BYTES)) return -EINVAL; @@ -1867,7 +1867,7 @@ static int snd_rme9652_playback_copy_kernel(struct snd_pcm_substream *substream, void *src, unsigned long count) { struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = rme9652_channel_buffer_location(rme9652, substream->pstr->stream, @@ -1883,7 +1883,7 @@ static int snd_rme9652_capture_copy(struct snd_pcm_substream *substream, void __user *dst, unsigned long count) { struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; if (snd_BUG_ON(pos + count > RME9652_CHANNEL_BUFFER_BYTES)) return -EINVAL; @@ -1903,7 +1903,7 @@ static int snd_rme9652_capture_copy_kernel(struct snd_pcm_substream *substream, void *dst, unsigned long count) { struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = rme9652_channel_buffer_location(rme9652, substream->pstr->stream, @@ -1919,7 +1919,7 @@ static int snd_rme9652_hw_silence(struct snd_pcm_substream *substream, unsigned long count) { struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = rme9652_channel_buffer_location (rme9652, substream->pstr->stream, -- 2.38.1

3 10

[bug report] ASoC: SOF: ipc4-loader: Support for loading external libraries
by Dan Carpenter 25 Oct '22

25 Oct '22

Hello Peter Ujfalusi, The patch 73c091a2fe96: "ASoC: SOF: ipc4-loader: Support for loading external libraries" from Oct 20, 2022, leads to the following Smatch static checker warning: sound/soc/sof/ipc4-loader.c:207 sof_ipc4_load_library_by_uuid() warn: 'payload_offset' unsigned <= 0 sound/soc/sof/ipc4-loader.c 167 static int sof_ipc4_load_library_by_uuid(struct snd_sof_dev *sdev, 168 unsigned long lib_id, const guid_t *uuid) 169 { 170 struct sof_ipc4_fw_data *ipc4_data = sdev->private; 171 struct sof_ipc4_fw_library *fw_lib; 172 const char *fw_filename; 173 size_t payload_offset; ^^^^^^^^^^^^^^^^^^^^^ 174 int ret, i, err; 175 176 if (!sdev->pdata->fw_lib_prefix) { 177 dev_err(sdev->dev, 178 "Library loading is not supported due to not set library path\n"); 179 return -EINVAL; 180 } 181 182 if (!ipc4_data->load_library) { 183 dev_err(sdev->dev, "Library loading is not supported on this platform\n"); 184 return -EOPNOTSUPP; 185 } 186 187 fw_lib = devm_kzalloc(sdev->dev, sizeof(*fw_lib), GFP_KERNEL); 188 if (!fw_lib) 189 return -ENOMEM; 190 191 fw_filename = kasprintf(GFP_KERNEL, "%s/%pUL.bin", 192 sdev->pdata->fw_lib_prefix, uuid); 193 if (!fw_filename) { 194 ret = -ENOMEM; 195 goto free_fw_lib; 196 } 197 198 ret = request_firmware(&fw_lib->sof_fw.fw, fw_filename, sdev->dev); 199 if (ret < 0) { 200 dev_err(sdev->dev, "Library file '%s' is missing\n", fw_filename); 201 goto free_filename; 202 } else { 203 dev_dbg(sdev->dev, "Library file '%s' loaded\n", fw_filename); 204 } 205 206 payload_offset = sof_ipc4_fw_parse_ext_man(sdev, fw_lib); --> 207 if (payload_offset <= 0) { ^^^^^^^^^^^^^^^^^^^ sof_ipc4_fw_parse_ext_man() returns negative error codes but as size_t. It should just return int. If it returns > INT_MAX that can't work on 32bit systems. 208 if (!payload_offset) 209 ret = -EINVAL; 210 else 211 ret = payload_offset; 212 213 goto release; 214 } 215 216 fw_lib->sof_fw.payload_offset = payload_offset; 217 fw_lib->id = lib_id; 218 219 /* Fix up the module ID numbers within the library */ 220 for (i = 0; i < fw_lib->num_modules; i++) 221 fw_lib->modules[i].man4_module_entry.id |= (lib_id << SOF_IPC4_MOD_LIB_ID_SHIFT); 222 223 /* 224 * Make sure that the DSP is booted and stays up while attempting the 225 * loading the library for the first time 226 */ 227 ret = pm_runtime_resume_and_get(sdev->dev); 228 if (ret < 0 && ret != -EACCES) { 229 dev_err_ratelimited(sdev->dev, "%s: pm_runtime resume failed: %d\n", 230 __func__, ret); 231 goto release; 232 } 233 234 ret = ipc4_data->load_library(sdev, fw_lib, false); 235 236 pm_runtime_mark_last_busy(sdev->dev); 237 err = pm_runtime_put_autosuspend(sdev->dev); 238 if (err < 0) 239 dev_err_ratelimited(sdev->dev, "%s: pm_runtime idle failed: %d\n", 240 __func__, err); 241 242 if (ret) 243 goto release; 244 245 ret = xa_insert(&ipc4_data->fw_lib_xa, lib_id, fw_lib, GFP_KERNEL); 246 if (unlikely(ret)) 247 goto release; 248 249 kfree(fw_filename); 250 251 return 0; 252 253 release: 254 release_firmware(fw_lib->sof_fw.fw); 255 /* Allocated within sof_ipc4_fw_parse_ext_man() */ 256 devm_kfree(sdev->dev, fw_lib->modules); 257 free_filename: 258 kfree(fw_filename); 259 free_fw_lib: 260 devm_kfree(sdev->dev, fw_lib); 261 262 return ret; 263 } regards, dan carpenter

2 1

[PATCH v4 00/11] ASoC: cleanups and improvements for jz4740-i2s
by Aidan MacDonald 25 Oct '22

25 Oct '22

This series is a preparatory cleanup of the jz4740-i2s driver before adding support for a new SoC. The two improvements are lifting unnecessary restrictions on sample rates and formats -- the existing ones appear to be derived from the limitations of the JZ4740's internal codec and don't reflect the actual capabilities of the I2S controller. I'm unable to test the series on any JZ47xx SoCs, but I have tested on an X1000 (which is the SoC I'll be adding in a followup series). Changes in v2: * Drop two patches already in sound for-next. * Squash two removal patches into the regmap fields patch. * Remove the unused 'mem' resource in the driver private struct. * Use regmap_set_bits() and regmap_clear_bits() to improve readability. * Add fix for SoCs with independent FIFO flush bits (ie. most of them). * Update sample formats patch with a more informative commit message. * Add two new patches to refactor DAI/component probing. Changes in v3: * Fix missing 'ret' in patch 11 (yes, that was pretty silly of me) Changes in v4: * Refactor FIFO flush bits fix so it doesn't depend on regmap conversion. Aidan MacDonald (11): ASoC: jz4740-i2s: Handle independent FIFO flush bits ASoC: jz4740-i2s: Remove unused 'mem' resource ASoC: jz4740-i2s: Convert to regmap API ASoC: jz4740-i2s: Simplify using regmap fields ASoC: jz4740-i2s: Use FIELD_PREP() macros in hw_params callback ASoC: jz4740-i2s: Align macro values and sort includes ASoC: jz4740-i2s: Make the PLL clock name SoC-specific ASoC: jz4740-i2s: Support S20_LE and S24_LE sample formats ASoC: jz4740-i2s: Support continuous sample rate ASoC: jz4740-i2s: Move component functions near the component driver ASoC: jz4740-i2s: Refactor DAI probe/remove ops as component ops sound/soc/jz4740/Kconfig | 1 + sound/soc/jz4740/jz4740-i2s.c | 461 ++++++++++++++++++---------------- 2 files changed, 248 insertions(+), 214 deletions(-) -- 2.35.1

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[PATCH] ALSA: au88x0: use explicitly signed char
by Jason A. Donenfeld 25 Oct '22

25 Oct '22

With char becoming unsigned by default, and with `char` alone being ambiguous and based on architecture, signed chars need to be marked explicitly as such. This fixes warnings like: sound/pci/au88x0/au88x0_core.c:2029 vortex_adb_checkinout() warn: signedness bug returning '(-22)' sound/pci/au88x0/au88x0_core.c:2046 vortex_adb_checkinout() warn: signedness bug returning '(-12)' sound/pci/au88x0/au88x0_core.c:2125 vortex_adb_allocroute() warn: 'vortex_adb_checkinout(vortex, (0), en, 0)' is unsigned sound/pci/au88x0/au88x0_core.c:2170 vortex_adb_allocroute() warn: 'vortex_adb_checkinout(vortex, stream->resources, en, 4)' is unsigned As well, since one function returns errnos, return an `int` rather than a `signed char`. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Jason A. Donenfeld <Jason(a)zx2c4.com> --- sound/pci/au88x0/au88x0.h | 6 +++--- sound/pci/au88x0/au88x0_core.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index 0aa7af049b1b..6cbb2bc4a048 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -141,7 +141,7 @@ struct snd_vortex { #ifndef CHIP_AU8810 stream_t dma_wt[NR_WT]; wt_voice_t wt_voice[NR_WT]; /* WT register cache. */ - char mixwt[(NR_WT / NR_WTPB) * 6]; /* WT mixin objects */ + s8 mixwt[(NR_WT / NR_WTPB) * 6]; /* WT mixin objects */ #endif /* Global resources */ @@ -235,8 +235,8 @@ static int vortex_alsafmt_aspfmt(snd_pcm_format_t alsafmt, vortex_t *v); static void vortex_connect_default(vortex_t * vortex, int en); static int vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type, int subdev); -static char vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, - int restype); +static int vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, + int restype); #ifndef CHIP_AU8810 static int vortex_wt_allocroute(vortex_t * vortex, int dma, int nr_ch); static void vortex_wt_connect(vortex_t * vortex, int en); diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 2ed5100b8cae..f217c02dfdfa 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1998,7 +1998,7 @@ static const int resnum[VORTEX_RESOURCE_LAST] = out: Mean checkout if != 0. Else mean Checkin resource. restype: Indicates type of resource to be checked in or out. */ -static char +static int vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) { int i, qty = resnum[restype], resinuse = 0; -- 2.38.1

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[PATCH] MAINTAINERS: update Tzung-Bi's email address
by Tzung-Bi Shih 25 Oct '22

25 Oct '22

Use kernel.org account instead. Signed-off-by: Tzung-Bi Shih <tzungbi(a)kernel.org> --- .../devicetree/bindings/sound/google,cros-ec-codec.yaml | 2 +- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 2 +- MAINTAINERS | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index c3e9f3485449..dea293f403d9 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -8,7 +8,7 @@ title: Audio codec controlled by ChromeOS EC maintainers: - Cheng-Yi Chiang <cychiang(a)chromium.org> - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Google's ChromeOS EC codec is a digital mic codec provided by the diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 1d73204451b1..ea7d4900ee4a 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Realtek rt1015p codec devicetree bindings maintainers: - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Rt1015p is a rt1015 variant which does not support I2C and diff --git a/MAINTAINERS b/MAINTAINERS index cf0f18502372..f9749afc0b9d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4906,7 +4906,7 @@ F: drivers/platform/chrome/ CHROMEOS EC CODEC DRIVER M: Cheng-Yi Chiang <cychiang(a)chromium.org> -M: Tzung-Bi Shih <tzungbi(a)google.com> +M: Tzung-Bi Shih <tzungbi(a)kernel.org> R: Guenter Roeck <groeck(a)chromium.org> L: chrome-platform(a)lists.linux.dev S: Maintained -- 2.38.0.135.g90850a2211-goog

2 2

[PATCH 0/3] Add missing dt-binding properties to rt5682(s)
by Nícolas F. R. A. Prado 24 Oct '22

24 Oct '22

This series adds the missing sound-dai-cells to rt5682s and supplies for both rt5682s and rt5682. Nícolas F. R. A. Prado (3): ASoC: dt-bindings: realtek,rt5682s: Add #sound-dai-cells ASoC: dt-bindings: realtek,rt5682s: Add AVDD and MICVDD supplies ASoC: dt-bindings: rt5682: Add AVDD, MICVDD and VBAT supplies .../devicetree/bindings/sound/realtek,rt5682s.yaml | 7 +++++++ Documentation/devicetree/bindings/sound/rt5682.txt | 6 ++++++ 2 files changed, 13 insertions(+) -- 2.38.1

3 13