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

[PATCH] ALSA: hda/realtek: Enable mute/micmute LEDs and limit mic boost on EliteBook 845 G8
by Kai-Heng Feng 20 Apr '21

20 Apr '21

On HP EliteBook 845 G8, the audio LEDs can be enabled by ALC285_FIXUP_HP_MUTE_LED. So use it accordingly. In addition to that, the mic captures lots of noises, so also limits the mic boost. The quality of capture audio becomes crystal clear after limiting the mic boost. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7544b77d3f7..c71f1f2b6a8c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6427,6 +6427,7 @@ enum { ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, ALC256_FIXUP_ACER_HEADSET_MIC, + ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -7901,6 +7902,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC285_FIXUP_HP_MUTE_LED, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8079,6 +8086,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.30.2

2 1

[PATCH v6] sound: rawmidi: Add framing mode
by David Henningsson 20 Apr '21

20 Apr '21

This commit adds a new framing mode that frames all MIDI data into 32-byte frames with a timestamp. The main benefit is that we can get accurate timestamps even if userspace wakeup and processing is not immediate. Testing on a Celeron N3150 with this mode has a max jitter of 2.8 ms, compared to the in-kernel seq implementation which has a max jitter of 5 ms during idle and much worse when running scheduler stress tests in parallel. Signed-off-by: David Henningsson <coding(a)diwic.se> --- Changes since v5: Added realtime clock and changed params struct according to Jaroslav's wishes. This version of the patch has been compile tested only. include/sound/rawmidi.h | 2 + include/uapi/sound/asound.h | 30 ++++++++++++- sound/core/rawmidi.c | 86 ++++++++++++++++++++++++++++++++++++- sound/core/rawmidi_compat.c | 4 +- 4 files changed, 118 insertions(+), 4 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 334842daa904..989e1517332d 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -81,6 +81,8 @@ struct snd_rawmidi_substream { bool opened; /* open flag */ bool append; /* append flag (merge more streams) */ bool active_sensing; /* send active sensing when close */ + unsigned int framing; /* whether to frame input data */ + unsigned int clock_type; /* clock source to use for input framing */ int use_count; /* use counter (for output) */ size_t bytes; struct snd_rawmidi *rmidi; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..773a00c0a1d8 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -710,7 +710,7 @@ enum { * Raw MIDI section - /dev/snd/midi?? */ -#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 1) +#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2) enum { SNDRV_RAWMIDI_STREAM_OUTPUT = 0, @@ -736,12 +736,38 @@ struct snd_rawmidi_info { unsigned char reserved[64]; /* reserved for future use */ }; +#define SNDRV_RAWMIDI_MODE_FRAMING_MASK (7<<0) +#define SNDRV_RAWMIDI_MODE_FRAMING_SHIFT 0 +#define SNDRV_RAWMIDI_MODE_FRAMING_NONE (0<<0) +#define SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP (1<<0) +#define SNDRV_RAWMIDI_MODE_CLOCK_MASK (7<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_SHIFT 3 +#define SNDRV_RAWMIDI_MODE_CLOCK_NONE (0<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_REALTIME (1<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC (2<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW (3<<3) + +#define SNDRV_RAWMIDI_FRAMING_DATA_LENGTH 16 + +struct snd_rawmidi_framing_tstamp { + /* For now, frame_type is always 0. Midi 2.0 is expected to add new + * types here. Applications are expected to skip unknown frame types. + */ + u8 frame_type; + u8 length; /* number of valid bytes in data field */ + u8 reserved[2]; + u32 tv_nsec; /* nanoseconds */ + u64 tv_sec; /* seconds */ + u8 data[SNDRV_RAWMIDI_FRAMING_DATA_LENGTH]; +} __attribute__((packed)); + struct snd_rawmidi_params { int stream; size_t buffer_size; /* queue size in bytes */ size_t avail_min; /* minimum avail bytes for wakeup */ unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */ - unsigned char reserved[16]; /* reserved for future use */ + unsigned int mode; /* For input data only, frame incoming data */ + unsigned char reserved[12]; /* reserved for future use */ }; #ifndef __KERNEL__ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index aca00af93afe..5d5f4363e887 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -680,9 +680,12 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, bool is_input) { char *newbuf, *oldbuf; + unsigned int framing = params->mode & SNDRV_RAWMIDI_MODE_FRAMING_MASK; if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; + if (framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP && (params->buffer_size & 0x1f) != 0) + return -EINVAL; if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; if (params->buffer_size != runtime->buffer_size) { @@ -720,7 +723,18 @@ EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { + unsigned int framing = params->mode & SNDRV_RAWMIDI_MODE_FRAMING_MASK; + unsigned int clock_type = params->mode & SNDRV_RAWMIDI_MODE_CLOCK_MASK; + + if (framing == SNDRV_RAWMIDI_MODE_FRAMING_NONE && clock_type != SNDRV_RAWMIDI_MODE_CLOCK_NONE) + return -EINVAL; + else if (clock_type > SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW) + return -EINVAL; + if (framing > SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) + return -EINVAL; snd_rawmidi_drain_input(substream); + substream->framing = framing; + substream->clock_type = clock_type; return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +977,61 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } +static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream, + const unsigned char *buffer, int src_count, const struct timespec64 *tstamp) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_framing_tstamp *dest_ptr; + struct snd_rawmidi_framing_tstamp frame = { .tv_sec = tstamp->tv_sec, .tv_nsec = tstamp->tv_nsec }; + int dest_frames = 0; + int frame_size = sizeof(struct snd_rawmidi_framing_tstamp); + + BUILD_BUG_ON(frame_size != 0x20); + if (snd_BUG_ON((runtime->hw_ptr & 0x1f) != 0)) + return -EINVAL; + + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + break; + } + if (src_count >= SNDRV_RAWMIDI_FRAMING_DATA_LENGTH) + frame.length = SNDRV_RAWMIDI_FRAMING_DATA_LENGTH; + else { + frame.length = src_count; + memset(frame.data, 0, SNDRV_RAWMIDI_FRAMING_DATA_LENGTH); + } + memcpy(frame.data, buffer, frame.length); + buffer += frame.length; + src_count -= frame.length; + dest_ptr = (struct snd_rawmidi_framing_tstamp *) (runtime->buffer + runtime->hw_ptr); + *dest_ptr = frame; + runtime->avail += frame_size; + runtime->hw_ptr += frame_size; + runtime->hw_ptr %= runtime->buffer_size; + dest_frames++; + } + return dest_frames * frame_size; +} + +static struct timespec64 get_framing_tstamp(struct snd_rawmidi_substream *substream) +{ + struct timespec64 ts64 = {0, 0}; + + switch (substream->clock_type) { + case SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW: + ktime_get_raw_ts64(&ts64); + break; + case SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC: + ktime_get_ts64(&ts64); + break; + case SNDRV_RAWMIDI_MODE_CLOCK_REALTIME: + ktime_get_real_ts64(&ts64); + break; + } + return ts64; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1046,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64 = get_framing_tstamp(substream); int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -987,8 +1057,11 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, "snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; } + spin_lock_irqsave(&runtime->lock, flags); - if (count == 1) { /* special case, faster code */ + if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; @@ -1534,6 +1607,7 @@ static __poll_t snd_rawmidi_poll(struct file *file, poll_table *wait) /* */ + static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -1541,6 +1615,8 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; unsigned long buffer_size, avail, xruns; + unsigned int clock_type; + static const char *clock_names[4] = { "none", "realtime", "monotonic", "monotonic raw" }; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1596,6 +1672,14 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Avail : %lu\n" " Overruns : %lu\n", buffer_size, avail, xruns); + if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { + clock_type = substream->clock_type >> SNDRV_RAWMIDI_MODE_CLOCK_SHIFT; + if (!snd_BUG_ON(clock_type >= sizeof(clock_names))) + snd_iprintf(buffer, + " Framing : tstamp\n" + " Clock type : %s\n", + clock_names[clock_type]); + } } } } diff --git a/sound/core/rawmidi_compat.c b/sound/core/rawmidi_compat.c index 7397130976d0..68a93443583c 100644 --- a/sound/core/rawmidi_compat.c +++ b/sound/core/rawmidi_compat.c @@ -13,7 +13,8 @@ struct snd_rawmidi_params32 { u32 buffer_size; u32 avail_min; unsigned int no_active_sensing; /* avoid bit-field */ - unsigned char reserved[16]; + unsigned int mode; + unsigned char reserved[12]; } __attribute__((packed)); static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, @@ -25,6 +26,7 @@ static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, if (get_user(params.stream, &src->stream) || get_user(params.buffer_size, &src->buffer_size) || get_user(params.avail_min, &src->avail_min) || + get_user(params.mode, &src->mode) || get_user(val, &src->no_active_sensing)) return -EFAULT; params.no_active_sensing = val; -- 2.25.1

3 2

[PATCH] ALSA: ctxfi: Avoid writing uninitialized variable in uaa_to_xfi()
by Yizhuo Zhai 20 Apr '21

20 Apr '21

Inside the function uaa_to_xfi(), variable "l_timer" could be uninitialized if pci_bus_read_config_dword() returns PCIBIOS_BAD_REGISTER_NUMBER. However, it is write back to PCI config space via pci_write_config_dword(), which is potentially unsafe. Signed-off-by: Yizhuo <yzhai003(a)ucr.edu> --- sound/pci/ctxfi/cthw20k1.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 0cea4982ed7d..7c3436499974 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1789,7 +1789,7 @@ static struct capabilities hw_capabilities(struct hw *hw) static int uaa_to_xfi(struct pci_dev *pci) { unsigned int bar0, bar1, bar2, bar3, bar4, bar5; - unsigned int cmd, irq, cl_size, l_timer, pwr; + unsigned int cmd, irq, cl_size, l_timer = ~0, pwr; unsigned int is_uaa; unsigned int data[4] = {0}; unsigned int io_base; -- 2.31.1 -- Kind Regards, *Yizhuo Zhai* *Computer Science, Graduate Student* *University of California, Riverside *

2 1

Headset MIC issue on HP ThinClient
by Kailang 20 Apr '21

20 Apr '21

Hi Takashi, Some HP machines, boot with plugged headset, the headset MIC will lose. Attach patch will solve it. BR, Kailang

2 1

[PATCH] ASoC: rt711-sdca: remove capture switch controls
by shumingf＠realtek.com 20 Apr '21

20 Apr '21

From: Shuming Fan <shumingf(a)realtek.com> The settings of the switch control already set by DAPM event. These switch controls might the user confused why it can't disable the capture. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index bfb7f1c8ec8f..2a09c305f4e4 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -652,14 +652,6 @@ static const struct snd_kcontrol_new rt711_sdca_snd_controls[] = { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_VOLUME, CH_R), 0x57, 0x57, 0, rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, out_vol_tlv), - SOC_DOUBLE_R("FU1E Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), - SOC_DOUBLE_R("FU0F Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), SOC_DOUBLE_R_EXT_TLV("FU1E Capture Volume", SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_L), SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_R), -- 2.29.0

5 10

[PATCH 1/2] ASoC: simple-card-utils: Propagate errors on too many links
by Thierry Reding 20 Apr '21

20 Apr '21

From: Thierry Reding <treding(a)nvidia.com> The DAI counting code doesn't propagate errors when the number of maximum links is exceeded, which causes subsequent initialization code to continue to run and that eventually leads to memory corruption with the code trying to access memory that is out of bounds. Fix this by propgating errors when the maximum number of links is reached, which ensures that the driver fails to load and prevents the memory corruption. Fixes: f2138aed231c ("ASoC: simple-card-utils: enable flexible CPU/Codec/Platform") Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/soc/generic/audio-graph-card.c | 19 +++++++++++-------- sound/soc/generic/simple-card.c | 17 ++++++++++------- 2 files changed, 21 insertions(+), 15 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 0582fe296471..2401212281c2 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -539,8 +539,8 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, return ret; } -static void graph_get_dais_count(struct asoc_simple_priv *priv, - struct link_info *li); +static int graph_get_dais_count(struct asoc_simple_priv *priv, + struct link_info *li); int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) { @@ -552,7 +552,10 @@ int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) card->dev = dev; memset(&li, 0, sizeof(li)); - graph_get_dais_count(priv, &li); + ret = graph_get_dais_count(priv, &li); + if (ret < 0) + return ret; + if (!li.link) return -EINVAL; @@ -657,8 +660,8 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, return 0; } -static void graph_get_dais_count(struct asoc_simple_priv *priv, - struct link_info *li) +static int graph_get_dais_count(struct asoc_simple_priv *priv, + struct link_info *li) { /* * link_num : number of links. @@ -706,9 +709,9 @@ static void graph_get_dais_count(struct asoc_simple_priv *priv, * => 4 DAIs = 2xCPU + 2xCodec * => 1 ccnf = 1xdummy-Codec */ - graph_for_each_link(priv, li, - graph_count_noml, - graph_count_dpcm); + return graph_for_each_link(priv, li, + graph_count_noml, + graph_count_dpcm); } int audio_graph_card_probe(struct snd_soc_card *card) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bf5ddf1ea65f..1d1c4309b582 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -526,8 +526,8 @@ static int simple_count_dpcm(struct asoc_simple_priv *priv, return 0; } -static void simple_get_dais_count(struct asoc_simple_priv *priv, - struct link_info *li) +static int simple_get_dais_count(struct asoc_simple_priv *priv, + struct link_info *li) { struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; @@ -584,12 +584,12 @@ static void simple_get_dais_count(struct asoc_simple_priv *priv, li->num[0].platforms = 1; li->link = 1; - return; + return 0; } - simple_for_each_link(priv, li, - simple_count_noml, - simple_count_dpcm); + return simple_for_each_link(priv, li, + simple_count_noml, + simple_count_dpcm); } static int simple_soc_probe(struct snd_soc_card *card) @@ -628,7 +628,10 @@ static int asoc_simple_probe(struct platform_device *pdev) card->probe = simple_soc_probe; memset(&li, 0, sizeof(li)); - simple_get_dais_count(priv, &li); + ret = simple_get_dais_count(priv, &li); + if (ret < 0) + return ret; + if (!li.link) return -EINVAL; -- 2.30.2

4 9

[PATCH v2] ALSA: usb-audio: Re-apply implicit feedback mode to Pioneer devices
by Takashi Iwai 20 Apr '21

20 Apr '21

Pioneer devices are supposed to be working with the implicit feedback mode, but so far the attempt to apply the implicit feedback caused issues, hence we explicitly skipped the implicit feedback mode for them. Recently, Geraldo discovered that the device actually works if you skip the generic matching of the sync EPs for the capture stream. That is, we should apply the implicit feedback setup for the playback like other similar devices, while we need to return 1 from audioformat_capture_quirk() so that no further matching will be done. And, later on, Olivia reported later that the fiddling with the capture quirk alone doesn't suffice for the test with speaker-test program. This seems to be a similar case like the recently fixed BOSS devices. Indeed, the problem could be addressed by setting playback_first flag, which indicates that the playback URBs have to be sent out at first even in the implicit feedback mode. This patch implements the application of the implicit feedback to Pioneer devices as described in the above. The former skip_pioneer_sync_ep() was dropped, and instead we provide is_pioneer_implicit_fb() to check the Pioneer devices that need the implicit feedback. In the audioformat_implicit_fb_quirk(), simply apply the implicit fb for playback and set chip->playback_first flag if matching, and in audioformat_capture_quirk()(), it returns 1 for skipping the generic EP sync handling. Reported-by: Geraldo <geraldogabriel(a)gmail.com> Tested-by: Olivia Mackintosh <livvy(a)base.nu> Link: https://lore.kernel.org/r/s5ha6pygqfz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/implicit.c | 42 +++++++++++++++++++++++++++--------------- 1 file changed, 27 insertions(+), 15 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 428e3a449f5d..77ffcea294dd 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -230,18 +230,27 @@ static int add_roland_implicit_fb(struct snd_usb_audio *chip, ifnum, alts); } -/* Playback and capture EPs on Pioneer devices share the same iface/altset, - * but they don't seem working with the implicit fb mode well, hence we - * just return as if the sync were already set up. +/* Playback and capture EPs on Pioneer devices share the same iface/altset + * for the implicit feedback operation */ -static int skip_pioneer_sync_ep(struct snd_usb_audio *chip, - struct audioformat *fmt, - struct usb_host_interface *alts) +static bool is_pioneer_implicit_fb(struct snd_usb_audio *chip, + struct usb_host_interface *alts) + { struct usb_endpoint_descriptor *epd; + if (USB_ID_VENDOR(chip->usb_id) != 0x2b73 && + USB_ID_VENDOR(chip->usb_id) != 0x08e4) + return false; + if (alts->desc.bInterfaceClass != USB_CLASS_VENDOR_SPEC) + return false; if (alts->desc.bNumEndpoints != 2) - return 0; + return false; + + epd = get_endpoint(alts, 0); + if (!usb_endpoint_is_isoc_out(epd) || + (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC) + return false; epd = get_endpoint(alts, 1); if (!usb_endpoint_is_isoc_in(epd) || @@ -250,8 +259,9 @@ static int skip_pioneer_sync_ep(struct snd_usb_audio *chip, USB_ENDPOINT_USAGE_DATA && (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != USB_ENDPOINT_USAGE_IMPLICIT_FB)) - return 0; - return 1; /* don't handle with the implicit fb, just skip sync EP */ + return false; + + return true; } static int __add_generic_implicit_fb(struct snd_usb_audio *chip, @@ -367,12 +377,12 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } /* Pioneer devices with vendor spec class */ - if (attr == USB_ENDPOINT_SYNC_ASYNC && - alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && - (USB_ID_VENDOR(chip->usb_id) == 0x2b73 || /* Pioneer */ - USB_ID_VENDOR(chip->usb_id) == 0x08e4 /* Pioneer */)) { - if (skip_pioneer_sync_ep(chip, fmt, alts)) - return 1; + if (is_pioneer_implicit_fb(chip, alts)) { + chip->playback_first = 1; + return add_implicit_fb_sync_ep(chip, fmt, + get_endpoint(alts, 1)->bEndpointAddress, + 1, alts->desc.bInterfaceNumber, + alts); } /* Try the generic implicit fb if available */ @@ -394,6 +404,8 @@ static int audioformat_capture_quirk(struct snd_usb_audio *chip, if (p && (p->type == IMPLICIT_FB_FIXED || p->type == IMPLICIT_FB_BOTH)) return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0, p->iface, NULL); + if (is_pioneer_implicit_fb(chip, alts)) + return 1; /* skip the quirk, also don't handle generic sync EP */ return 0; } -- 2.26.2

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[PATCH 0/5] ASoC: rt286/rt298: Fixes for DMIC2 config and combo jack
by David Ward 19 Apr '21

19 Apr '21

The last two patches in this series fix a longstanding issue that prevented the ALC3263 codec from using a headset mic. This codec can be found on Dell systems including the Latitude 13 7350, Venue 11 Pro 7140, and XPS 13 9343. In fact, there is an ACPI quirk for the XPS 13 9343, which forces it to use legacy HD Audio just to avoid this issue: https://lore.kernel.org/alsa-devel/CAPeXnHv07HkvcHrYFmZMr8OTp7U7F=k_k=LPYnU… This may allow that ACPI quirk to be removed. Either way, the other systems mentioned above do not support this quirk and already use the ASoC driver, so this fix is necessary for headset mic support on those systems. Note: there is likely other handling for this codec that only exists in the HDA driver, but which also belongs in the ASoC driver. Commit 394c97f824fa ("ALSA: hda/realtek - Change EAPD to verb control") describes an issue that does not seem to be resolved in the ASoC driver, to give an example. Other patches in this series are not specific to the ALC3263. These patches set the correct combo jack configuration when headphones are inserted, and fix a misaligned value set in the DMIC2 Configuration Default register. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=114171 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> David Ward (5): ASoC: rt286: Fix upper byte in DMIC2 configuration ASoC: rt286: Configure combo jack for headphones ASoC: rt298: Configure combo jack for headphones ASoC: rt286: Make RT286_SET_GPIO_* readable and writable ASoC: rt286: Generalize support for ALC3263 codec sound/soc/codecs/rt286.c | 34 +++++++++++++++++++++------------- sound/soc/codecs/rt298.c | 9 +++++++-- 2 files changed, 28 insertions(+), 15 deletions(-) -- 2.31.1

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[PATCH 0/5] ASoC: remove more cppcheck warnings
by Pierre-Louis Bossart 19 Apr '21

19 Apr '21

I missed those warnings in the previous rounds, they are mostly trivial and shouldn't change the behavior. Pierre-Louis Bossart (5): ASoC: soc-acpi: remove useless initialization ASoC: soc-core: fix signed/unsigned issue ASoC: soc-core: fix always-false condition ASoC: codecs: lpass-rx-macro: remove useless return ASoC: codecs: rt5682: clarify expression sound/soc/codecs/lpass-rx-macro.c | 2 -- sound/soc/codecs/rt5682.c | 2 +- sound/soc/soc-acpi.c | 2 +- sound/soc/soc-core.c | 7 +------ 4 files changed, 3 insertions(+), 10 deletions(-) -- 2.25.1

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[PATCH] ASoC: sigmadsp: Disable cache mechanism for readbacks
by Niklas Carlsson 19 Apr '21

19 Apr '21

From: Niklas Carlsson <niklasc(a)axis.com> The ALSA control readback functionality only works for non-volatile controls, i.e. control values that does not change on their own without driver interaction. This doesn't work for readbacks since the DSP firmware updates the control value. Disable the cache mechanism in the driver if the control name matches the prefix used for readbacks to ensure that the control value is valid. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/sigmadsp.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 76c77dc8ecf7..b992216aee55 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -24,6 +24,8 @@ #define SIGMA_FW_CHUNK_TYPE_CONTROL 1 #define SIGMA_FW_CHUNK_TYPE_SAMPLERATES 2 +#define READBACK_CTRL_NAME "ReadBack" + struct sigmadsp_control { struct list_head head; uint32_t samplerates; @@ -31,6 +33,7 @@ struct sigmadsp_control { unsigned int num_bytes; const char *name; struct snd_kcontrol *kcontrol; + bool is_readback; bool cached; uint8_t cache[]; }; @@ -141,7 +144,8 @@ static int sigmadsp_ctrl_put(struct snd_kcontrol *kcontrol, if (ret == 0) { memcpy(ctrl->cache, data, ctrl->num_bytes); - ctrl->cached = true; + if (!ctrl->is_readback) + ctrl->cached = true; } mutex_unlock(&sigmadsp->lock); @@ -164,7 +168,8 @@ static int sigmadsp_ctrl_get(struct snd_kcontrol *kcontrol, } if (ret == 0) { - ctrl->cached = true; + if (!ctrl->is_readback) + ctrl->cached = true; memcpy(ucontrol->value.bytes.data, ctrl->cache, ctrl->num_bytes); } @@ -231,6 +236,15 @@ static int sigma_fw_load_control(struct sigmadsp *sigmadsp, name[name_len] = '\0'; ctrl->name = name; + /* + * Readbacks doesn't work with non-volatile controls, since the + * firmware updates the control value without driver interaction. Mark + * the readbacks to ensure that the values are not cached. + */ + if (ctrl->name && strncmp(ctrl->name, READBACK_CTRL_NAME, + (sizeof(READBACK_CTRL_NAME) - 1)) == 0) + ctrl->is_readback = true; + ctrl->addr = le16_to_cpu(ctrl_chunk->addr); ctrl->num_bytes = num_bytes; ctrl->samplerates = le32_to_cpu(chunk->samplerates); -- 2.20.1

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