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

[PATCH] ALSA: line6: Fix racy initialization of LINE6 MIDI
by Takashi Iwai 18 May '21

18 May '21

The initialization of MIDI devices that are found on some LINE6 drivers are currently done in a racy way; namely, the MIDI buffer instance is allocated and initialized in each private_init callback while the communication with the interface is already started via line6_init_cap_control() call before that point. This may lead to Oops in line6_data_received() when a spurious event is received, as reported by syzkaller. This patch moves the MIDI initialization to line6_init_cap_control() as well instead of the too-lately-called private_init for avoiding the race. Also this reduces slightly more lines, so it's a win-win change. Reported-by: syzbot+0d2b3feb0a2887862e06@syzkallerlkml..appspotmail.com Link: https://lore.kernel.org/r/000000000000a4be9405c28520de@google.com Link: https://lore.kernel.org/r/20210517132725.GA50495@hyeyoo Cc: Hyeonggon Yoo <42.hyeyoo(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/driver.c | 4 ++++ sound/usb/line6/pod.c | 5 ----- sound/usb/line6/variax.c | 6 ------ 3 files changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index a030dd65eb28..9602929b7de9 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -699,6 +699,10 @@ static int line6_init_cap_control(struct usb_line6 *line6) line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); if (!line6->buffer_message) return -ENOMEM; + + ret = line6_init_midi(line6); + if (ret < 0) + return ret; } else { ret = line6_hwdep_init(line6); if (ret < 0) diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c index cd44cb5f1310..16e644330c4d 100644 --- a/sound/usb/line6/pod.c +++ b/sound/usb/line6/pod.c @@ -376,11 +376,6 @@ static int pod_init(struct usb_line6 *line6, if (err < 0) return err; - /* initialize MIDI subsystem: */ - err = line6_init_midi(line6); - if (err < 0) - return err; - /* initialize PCM subsystem: */ err = line6_init_pcm(line6, &pod_pcm_properties); if (err < 0) diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c index ed158f04de80..c2245aa93b08 100644 --- a/sound/usb/line6/variax.c +++ b/sound/usb/line6/variax.c @@ -159,7 +159,6 @@ static int variax_init(struct usb_line6 *line6, const struct usb_device_id *id) { struct usb_line6_variax *variax = line6_to_variax(line6); - int err; line6->process_message = line6_variax_process_message; line6->disconnect = line6_variax_disconnect; @@ -172,11 +171,6 @@ static int variax_init(struct usb_line6 *line6, if (variax->buffer_activate == NULL) return -ENOMEM; - /* initialize MIDI subsystem: */ - err = line6_init_midi(&variax->line6); - if (err < 0) - return err; - /* initiate startup procedure: */ schedule_delayed_work(&line6->startup_work, msecs_to_jiffies(VARIAX_STARTUP_DELAY1)); -- 2.26.2

1 0

[PATCH v3 00/11] ALSA: oxfw: code refactoring for quirks specific to ASICs
by Takashi Sakamoto 18 May '21

18 May '21

Hi, This patchset is revised version of my take 2: * https://lore.kernel.org/alsa-devel/20210515071112.101535-1-o-takashi@sakamo… Changes from v2: * use int type instead of enumerated type for combination of enumeration-constants * fix semi-colon typo in subject line of 10th patch Regards Takashi Sakamoto (11): Revert "ALSA: bebob/oxfw: fix Kconfig entry for Mackie d.2 Pro" ALSA: firewire-lib/motu: use int type for the value of bitwise OR with enumerator-constant ALSA: oxfw: code refactoring for existent device entry with specifier_id and version ALSA: oxfw: code refactoring to detect mackie models ALSA: oxfw: add explicit device entry for Loud Technologies Tapco Link.FireWire 4x6 ALSA: oxfw: add explicit device entry for Loud Technologies Mackie Onyx Sattelite ALSA: oxfw: add comment for the type of ASICs ALSA: oxfw: code refactoring for jumbo-payload quirk in OXFW970 ALSA: firewire-lib: code refactoring for jumbo payload quirk ALSA: oxfw: code refactoring for wrong_dbs quirk ALSA: oxfw: add quirk flag for blocking transmission method sound/firewire/Kconfig | 4 +- sound/firewire/amdtp-am824.c | 4 +- sound/firewire/amdtp-am824.h | 2 +- sound/firewire/amdtp-stream.c | 11 ++- sound/firewire/amdtp-stream.h | 5 +- sound/firewire/bebob/bebob.c | 2 +- sound/firewire/digi00x/amdtp-dot.c | 2 +- sound/firewire/motu/motu.h | 3 +- sound/firewire/oxfw/oxfw-stream.c | 25 +++--- sound/firewire/oxfw/oxfw.c | 137 ++++++++++++++--------------- sound/firewire/oxfw/oxfw.h | 13 ++- 11 files changed, 110 insertions(+), 98 deletions(-) -- 2.27.0

2 15

[PATCH] sound: rawmidi: Add framing mode
by David Henningsson 18 May '21

18 May '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> --- Hi, it's been a while. Both because I've been busy with things that needed to take priority, and because I wanted to do a little more than just compile testing it before submitting; hopefully this one is mergable. That testing showed that I needed to change from "u8" to "__u8" in the headers. Other than that I think it is just the changes that Takashi Iwai pointed out in v6. include/sound/rawmidi.h | 2 + include/uapi/sound/asound.h | 30 +++++++++++- sound/core/rawmidi.c | 93 ++++++++++++++++++++++++++++++++++++- sound/core/rawmidi_compat.c | 4 +- 4 files changed, 124 insertions(+), 5 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..d17c061950df 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]; +} __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..4a6534db77d6 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) { @@ -720,8 +723,24 @@ 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; + int err; + + 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); - return resize_runtime_buffer(substream->runtime, params, true); + err = resize_runtime_buffer(substream->runtime, params, true); + if (err < 0) + return err; + + substream->framing = framing; + substream->clock_type = clock_type; + return 0; } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +982,62 @@ 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 orig_count = src_count; + 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 orig_count - src_count; +} + +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 +1052,7 @@ 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 +1063,11 @@ 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]; @@ -1541,6 +1620,8 @@ 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 +1677,14 @@ 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 3

[PATCH] ALSA: dice: fix stream format for TC Electronic Konnekt Live at high sampling transfer frequency
by Takashi Sakamoto 18 May '21

18 May '21

At high sampling transfer frequency, TC Electronic Konnekt Live transfers/receives 6 audio data frames in multi bit linear audio data channel of data block in CIP payload. Current hard-coded stream format is wrong. Cc: <stable(a)vger.kernel.org> Fixes: f1f0f330b1d0 ("ALSA: dice: add parameters of stream formats for models produced by TC Electronic") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-tcelectronic.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/dice/dice-tcelectronic.c b/sound/firewire/dice/dice-tcelectronic.c index a8875d24ba2a..43a3bcb15b3d 100644 --- a/sound/firewire/dice/dice-tcelectronic.c +++ b/sound/firewire/dice/dice-tcelectronic.c @@ -38,8 +38,8 @@ static const struct dice_tc_spec konnekt_24d = { }; static const struct dice_tc_spec konnekt_live = { - .tx_pcm_chs = {{16, 16, 16}, {0, 0, 0} }, - .rx_pcm_chs = {{16, 16, 16}, {0, 0, 0} }, + .tx_pcm_chs = {{16, 16, 6}, {0, 0, 0} }, + .rx_pcm_chs = {{16, 16, 6}, {0, 0, 0} }, .has_midi = true, }; -- 2.27.0

2 1

[PATCH] ALSA: dice: disable double_pcm_frames mode for M-Audio Profire 610, 2626 and Avid M-Box 3 Pro
by Takashi Sakamoto 18 May '21

18 May '21

ALSA dice driver detects jumbo payload at high sampling transfer frequency for below models: * Avid M-Box 3 Pro * M-Audio Profire 610 * M-Audio Profire 2626 Although many DICE-based devices have a quirk at high sampling transfer frequency to multiplex double number of PCM frames into data block than the number in IEC 61883-1/6, the above devices are just compliant to IEC 61883-1/6. This commit disables the mode of double_pcm_frames for the models. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-pcm.c | 4 ++-- sound/firewire/dice/dice-stream.c | 2 +- sound/firewire/dice/dice.c | 24 ++++++++++++++++++++++++ sound/firewire/dice/dice.h | 3 ++- 4 files changed, 29 insertions(+), 4 deletions(-) diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index af8a90ee40f3..a69ca1111b03 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -218,7 +218,7 @@ static int pcm_open(struct snd_pcm_substream *substream) if (frames_per_period > 0) { // For double_pcm_frame quirk. - if (rate > 96000) { + if (rate > 96000 && !dice->disable_double_pcm_frames) { frames_per_period *= 2; frames_per_buffer *= 2; } @@ -273,7 +273,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, mutex_lock(&dice->mutex); // For double_pcm_frame quirk. - if (rate > 96000) { + if (rate > 96000 && !dice->disable_double_pcm_frames) { events_per_period /= 2; events_per_buffer /= 2; } diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 1a14c083e8ce..c4dfe76500c2 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -181,7 +181,7 @@ static int keep_resources(struct snd_dice *dice, struct amdtp_stream *stream, // as 'Dual Wire'. // For this quirk, blocking mode is required and PCM buffer size should // be aligned to SYT_INTERVAL. - double_pcm_frames = rate > 96000; + double_pcm_frames = (rate > 96000 && !dice->disable_double_pcm_frames); if (double_pcm_frames) { rate /= 2; pcm_chs *= 2; diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 107a81691f0e..239d164b0eea 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -21,6 +21,7 @@ MODULE_LICENSE("GPL v2"); #define OUI_SSL 0x0050c2 // Actually ID reserved by IEEE. #define OUI_PRESONUS 0x000a92 #define OUI_HARMAN 0x000fd7 +#define OUI_AVID 0x00a07e #define DICE_CATEGORY_ID 0x04 #define WEISS_CATEGORY_ID 0x00 @@ -222,6 +223,14 @@ static int dice_probe(struct fw_unit *unit, (snd_dice_detect_formats_t)entry->driver_data; } + // Below models are compliant to IEC 61883-1/6 and have no quirk at high sampling transfer + // frequency. + // * Avid M-Box 3 Pro + // * M-Audio Profire 610 + // * M-Audio Profire 2626 + if (entry->vendor_id == OUI_MAUDIO || entry->vendor_id == OUI_AVID) + dice->disable_double_pcm_frames = true; + spin_lock_init(&dice->lock); mutex_init(&dice->mutex); init_completion(&dice->clock_accepted); @@ -278,7 +287,22 @@ static void dice_bus_reset(struct fw_unit *unit) #define DICE_INTERFACE 0x000001 +#define DICE_DEV_ENTRY_TYPICAL(vendor, model, data) \ + { \ + .match_flags = IEEE1394_MATCH_VENDOR_ID | \ + IEEE1394_MATCH_MODEL_ID | \ + IEEE1394_MATCH_SPECIFIER_ID | \ + IEEE1394_MATCH_VERSION, \ + .vendor_id = (vendor), \ + .model_id = (model), \ + .specifier_id = (vendor), \ + .version = DICE_INTERFACE, \ + .driver_data = (kernel_ulong_t)(data), \ + } + static const struct ieee1394_device_id dice_id_table[] = { + // Avid M-Box 3 Pro. To match in probe function. + DICE_DEV_ENTRY_TYPICAL(OUI_AVID, 0x000004, snd_dice_detect_extension_formats), /* M-Audio Profire 2626 has a different value in version field. */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index adc6f7c84460..3c967d1b3605 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -109,7 +109,8 @@ struct snd_dice { struct fw_iso_resources rx_resources[MAX_STREAMS]; struct amdtp_stream tx_stream[MAX_STREAMS]; struct amdtp_stream rx_stream[MAX_STREAMS]; - bool global_enabled; + bool global_enabled:1; + bool disable_double_pcm_frames:1; struct completion clock_accepted; unsigned int substreams_counter; -- 2.27.0

2 1

[PATCH v2 00/10] ALSA: oxfw: code refactoring for quirks specific to ASICs
by Takashi Sakamoto 18 May '21

18 May '21

Hi, This patchset is reveised version of my initial post: * https://lore.kernel.org/alsa-devel/20210514085435.92807-1-o-takashi@sakamoc… Changes from v1: * revert Kconfig fix * fix detection of mackie models * add quirk flag for blocking transmission Regards Takashi Sakamoto (10): Revert "ALSA: bebob/oxfw: fix Kconfig entry for Mackie d.2 Pro" ALSA: oxfw: code refactoring for existent device entry with specifier_id and version ALSA: oxfw: code refactoring to detect mackie models ALSA: oxfw: add explicit device entry for Loud Technologies Tapco Link.FireWire 4x6 ALSA: oxfw: add explicit device entry for Loud Technologies Mackie Onyx Sattelite ALSA: oxfw: add comment for the type of ASICs ALSA: oxfw: code refactoring for jumbo-payload quirk in OXFW970 ALSA: firewire-lib: code refactoring for jumbo payload quirk ALSA: oxfw; code refactoring for wrong_dbs quirk ALSA: oxfw: add quirk flag for blocking transmission method sound/firewire/Kconfig | 4 +- sound/firewire/amdtp-stream.c | 7 +- sound/firewire/bebob/bebob.c | 2 +- sound/firewire/oxfw/oxfw-stream.c | 25 +++--- sound/firewire/oxfw/oxfw.c | 137 ++++++++++++++---------------- sound/firewire/oxfw/oxfw.h | 12 ++- 6 files changed, 98 insertions(+), 89 deletions(-) -- 2.27.0

2 13

[PATCH v5 0/9] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 18 May '21

18 May '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Eventhough this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Thanks, srini Changes since v4: - updated dt-bindings and example as suggested by RobH - moved SoundWire specific functions to sdw driver. - fixed wrong indentation reported by kernel test robot Srinivas Kandagatla (9): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw ASoC: codecs: wcd938x-sdw: add SoundWire driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x-sdw.yaml | 70 + .../bindings/sound/qcom,wcd938x.yaml | 149 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wcd-clsh-v2.c | 348 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 298 ++ sound/soc/codecs/wcd938x.c | 3753 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 705 ++++ 9 files changed, 5346 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

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[PATCH] dt-bindings: More removals of type references on common properties
by Rob Herring 18 May '21

18 May '21

Users of common properties shouldn't have a type definition as the common schemas already have one. A few new ones slipped in and *-names was missed in the last clean-up pass. Drop all the unnecessary type references in the tree. A meta-schema update to catch these is pending. Cc: Luca Ceresoli <luca(a)lucaceresoli.net> Cc: Stephen Boyd <sboyd(a)kernel.org> Cc: Olivier Moysan <olivier.moysan(a)foss.st.com> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> Cc: Jonathan Cameron <jic23(a)kernel.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Bjorn Andersson <bjorn.andersson(a)linaro.org> Cc: Georgi Djakov <djakov(a)kernel.org> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Sebastian Reichel <sre(a)kernel.org> Cc: Orson Zhai <orsonzhai(a)gmail.com> Cc: Baolin Wang <baolin.wang7(a)gmail.com> Cc: Chunyan Zhang <zhang.lyra(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Fabrice Gasnier <fabrice.gasnier(a)st.com> Cc: Odelu Kukatla <okukatla(a)codeaurora.org> Cc: Alex Elder <elder(a)kernel.org> Cc: Shengjiu Wang <shengjiu.wang(a)nxp.com> Cc: linux-clk(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-iio(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-input(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/clock/idt,versaclock5.yaml | 2 -- .../devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 1 - Documentation/devicetree/bindings/input/input.yaml | 1 - Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml | 1 - Documentation/devicetree/bindings/net/qcom,ipa.yaml | 1 - .../devicetree/bindings/power/supply/sc2731-charger.yaml | 2 +- Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml | 2 +- 7 files changed, 2 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml b/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml index c268debe5b8d..28675b0b80f1 100644 --- a/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml +++ b/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml @@ -60,7 +60,6 @@ properties: maxItems: 2 idt,xtal-load-femtofarads: - $ref: /schemas/types.yaml#/definitions/uint32 minimum: 9000 maximum: 22760 description: Optional load capacitor for XTAL1 and XTAL2 @@ -84,7 +83,6 @@ patternProperties: enum: [ 1800000, 2500000, 3300000 ] idt,slew-percent: description: The Slew rate control for CMOS single-ended. - $ref: /schemas/types.yaml#/definitions/uint32 enum: [ 80, 85, 90, 100 ] required: diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml index 6f2398cdc82d..1e7894e524f9 100644 --- a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml @@ -102,7 +102,6 @@ patternProperties: st,adc-channel-names: description: List of single-ended channel names. - $ref: /schemas/types.yaml#/definitions/string-array st,filter-order: description: | diff --git a/Documentation/devicetree/bindings/input/input.yaml b/Documentation/devicetree/bindings/input/input.yaml index 74244d21d2b3..d41d8743aad4 100644 --- a/Documentation/devicetree/bindings/input/input.yaml +++ b/Documentation/devicetree/bindings/input/input.yaml @@ -38,6 +38,5 @@ properties: Duration in seconds which the key should be kept pressed for device to reset automatically. Device with key pressed reset feature can specify this property. - $ref: /schemas/types.yaml#/definitions/uint32 additionalProperties: true diff --git a/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml b/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml index cb6498108b78..36c955965d90 100644 --- a/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml +++ b/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml @@ -92,7 +92,6 @@ properties: this interconnect to send RPMh commands. qcom,bcm-voter-names: - $ref: /schemas/types.yaml#/definitions/string-array description: | Names for each of the qcom,bcm-voters specified. diff --git a/Documentation/devicetree/bindings/net/qcom,ipa.yaml b/Documentation/devicetree/bindings/net/qcom,ipa.yaml index 7443490d4cc6..5fe6d3dceb08 100644 --- a/Documentation/devicetree/bindings/net/qcom,ipa.yaml +++ b/Documentation/devicetree/bindings/net/qcom,ipa.yaml @@ -105,7 +105,6 @@ properties: - description: Whether the IPA clock is enabled (if valid) qcom,smem-state-names: - $ref: /schemas/types.yaml#/definitions/string-array description: The names of the state bits used for SMP2P output items: - const: ipa-clock-enabled-valid diff --git a/Documentation/devicetree/bindings/power/supply/sc2731-charger.yaml b/Documentation/devicetree/bindings/power/supply/sc2731-charger.yaml index db1aa238cda5..b62c2431f94e 100644 --- a/Documentation/devicetree/bindings/power/supply/sc2731-charger.yaml +++ b/Documentation/devicetree/bindings/power/supply/sc2731-charger.yaml @@ -20,7 +20,7 @@ properties: maxItems: 1 phys: - $ref: /schemas/types.yaml#/definitions/phandle + maxItems: 1 description: phandle to the USB phy monitored-battery: diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml index b4c190bddd84..61802a11baf4 100644 --- a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -49,7 +49,7 @@ properties: maxItems: 1 memory-region: - $ref: /schemas/types.yaml#/definitions/phandle + maxItems: 1 description: phandle to a node describing reserved memory (System RAM memory) The M core can't access all the DDR memory space on some platform, -- 2.27.0

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[RFC PATCH] sound: line6: Fix race condition in line6_probe
by Hyeonggon Yoo 17 May '21

17 May '21

syzbot reported general protection fault in midibuf_is_full. the cause is linemidi pointer in struct usb_line6 isn't properly initialized. the pointer isn't initialized because there is race condition in line6_probe. it calls line6_init_cap_control first, which submits urb. and then it initializes it's data using private_init function. so it's possible line6_data_received is called before it's data isn't initialized. Link: https://lkml.org/lkml/2021/5/17/543 Reported-by: syzbot+0d2b3feb0a2887862e06@syzkallerlkml..appspotmail.com Signed-off-by: Hyeonggon Yoo <42.hyeyoo(a)gmail.com> --- sound/usb/line6/driver.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index a030dd65eb28..2c183a2a30f0 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -788,17 +788,17 @@ int line6_probe(struct usb_interface *interface, line6_get_usb_properties(line6); + /* initialize device data based on device: */ + ret = private_init(line6, id); + if (ret < 0) + goto error; + if (properties->capabilities & LINE6_CAP_CONTROL) { ret = line6_init_cap_control(line6); if (ret < 0) goto error; } - /* initialize device data based on device: */ - ret = private_init(line6, id); - if (ret < 0) - goto error; - /* creation of additional special files should go here */ dev_info(&interface->dev, "Line 6 %s now attached\n", -- 2.25.1

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[PATCH 00/11] ALSA: usx2y: Fixes and cleanups
by Takashi Iwai 17 May '21

17 May '21

Hi, this is a patch series for usx2y drivers. Most part of changes are about cleanup of the code. The current usx2y driver is fairly unmaintained code, and this was the cause of some overseen bugs (or rather confusion) as happened in UMN fiasco. So the initial goal is to make a bit more readable for further reviews. The remaining part of changes are actual fixes. At this time, I concentrated only on the obvious issues that could be spotted out after the cleanup. The driver is likely broken in other ways, but the identification and the fix would need the actual device, so I leave them for now. Takashi === Takashi Iwai (11): ALSA: usx2y: Avoid camelCase ALSA: usx2y: Fix spaces ALSA: usx2y: Coding style fixes ALSA: usx2y: Fix potential leaks of uninitialized memory ALSA: usx2y: Avoid self-killing ALSA: usx2y: Fix potential memory leaks ALSA: usxy2: Fix potential doubly allocations ALSA: usx2y: Fix shmem initialization ALSA: usx2y: Don't call free_pages_exact() with NULL address ALSA: usx2y: Cleanup probe and disconnect callbacks ALSA: usx2y: Nuke pcm_list sound/usb/usx2y/us122l.c | 61 +-- sound/usb/usx2y/us122l.h | 2 +- sound/usb/usx2y/usX2Yhwdep.c | 135 +++--- sound/usb/usx2y/usX2Yhwdep.h | 2 +- sound/usb/usx2y/usb_stream.c | 82 ++-- sound/usb/usx2y/usb_stream.h | 23 +- sound/usb/usx2y/usbus428ctldefs.h | 104 ++--- sound/usb/usx2y/usbusx2y.c | 389 +++++++++-------- sound/usb/usx2y/usbusx2y.h | 65 ++- sound/usb/usx2y/usbusx2yaudio.c | 684 ++++++++++++++++-------------- sound/usb/usx2y/usx2yhwdeppcm.c | 627 ++++++++++++++------------- sound/usb/usx2y/usx2yhwdeppcm.h | 4 +- 12 files changed, 1146 insertions(+), 1032 deletions(-) -- 2.26.2

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