From b2345a8a4342cf83316a2198fa915c7c99b7d6c7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Jan 2021 08:52:15 +0100 Subject: [PATCH 01/20] ALSA: usb-audio: Fix the missing endpoints creations for quirks MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The recent change in the endpoint management moved the endpoint object creation from the stream open time to the parser of the audio descriptor. It works fine for the standard audio, but it overlooked the other places that create audio streams via quirks (QUIRK_AUDIO_FIXED_ENDPOINT) like the reported a few Pioneer devices; those call snd_usb_add_audio_stream() manually, hence they miss the endpoints, eventually resulting in the error at opening streams. Moreover, now the sync EP setup was moved to the explicit call of snd_usb_audioformat_set_sync_ep(), and this needs to be added for those places, too. This patch addresses those regressions for quirks. It adds a local helper function add_audio_stream_from_fixed_fmt(), which does the all needed tasks, and replaces the calls of snd_usb_add_audio_stream() with this new function. Fixes: 54cb31901b83 ("ALSA: usb-audio: Create endpoint objects at parsing phase") Reported-by: František Kučera Link: https://lore.kernel.org/r/20210108075219.21463-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 54 +++++++++++++++++++++++++++++++++++++--------- 1 file changed, 44 insertions(+), 10 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e4a690bb4c99..b70e2ebc3e29 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -120,6 +120,40 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, return 0; } +/* create the audio stream and the corresponding endpoints from the fixed + * audioformat object; this is used for quirks with the fixed EPs + */ +static int add_audio_stream_from_fixed_fmt(struct snd_usb_audio *chip, + struct audioformat *fp) +{ + int stream, err; + + stream = (fp->endpoint & USB_DIR_IN) ? + SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + + snd_usb_audioformat_set_sync_ep(chip, fp); + + err = snd_usb_add_audio_stream(chip, stream, fp); + if (err < 0) + return err; + + err = snd_usb_add_endpoint(chip, fp->endpoint, + SND_USB_ENDPOINT_TYPE_DATA); + if (err < 0) + return err; + + if (fp->sync_ep) { + err = snd_usb_add_endpoint(chip, fp->sync_ep, + fp->implicit_fb ? + SND_USB_ENDPOINT_TYPE_DATA : + SND_USB_ENDPOINT_TYPE_SYNC); + if (err < 0) + return err; + } + + return 0; +} + /* * create a stream for an endpoint/altsetting without proper descriptors */ @@ -131,8 +165,8 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, struct audioformat *fp; struct usb_host_interface *alts; struct usb_interface_descriptor *altsd; - int stream, err; unsigned *rate_table = NULL; + int err; fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); if (!fp) @@ -153,11 +187,6 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, fp->rate_table = rate_table; } - stream = (fp->endpoint & USB_DIR_IN) - ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = snd_usb_add_audio_stream(chip, stream, fp); - if (err < 0) - goto error; if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber || fp->altset_idx >= iface->num_altsetting) { err = -EINVAL; @@ -176,6 +205,13 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, fp->datainterval = snd_usb_parse_datainterval(chip, alts); if (fp->maxpacksize == 0) fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + if (!fp->fmt_type) + fp->fmt_type = UAC_FORMAT_TYPE_I; + + err = add_audio_stream_from_fixed_fmt(chip, fp); + if (err < 0) + goto error; + usb_set_interface(chip->dev, fp->iface, 0); snd_usb_init_pitch(chip, fp); snd_usb_init_sample_rate(chip, fp, fp->rate_max); @@ -417,7 +453,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, struct usb_host_interface *alts; struct usb_interface_descriptor *altsd; struct audioformat *fp; - int stream, err; + int err; /* both PCM and MIDI interfaces have 2 or more altsettings */ if (iface->num_altsetting < 2) @@ -482,9 +518,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, return -ENXIO; } - stream = (fp->endpoint & USB_DIR_IN) - ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = snd_usb_add_audio_stream(chip, stream, fp); + err = add_audio_stream_from_fixed_fmt(chip, fp); if (err < 0) { list_del(&fp->list); /* unlink for avoiding double-free */ kfree(fp); From 5d15f1eb456025cf47078fdbc230d7a9f1ee4cef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Jan 2021 08:52:16 +0100 Subject: [PATCH 02/20] ALSA: usb-audio: Choose audioformat of a counter-part substream The implicit feedback mode needs to handle two endpoints and the choice of the audioformat object for the sync EP is important since this determines the compatibility of the hw_params. The current code uses the same audioformat object if both the main EP and the sync EP point to the same iface/altsetting. This was done in consideration of the non-implicit-fb sync EP handling, and it doesn't match well with the cases where actually to endpoints are defined in the sameiface / altsetting like a few Pioneer devices. Modify snd_usb_find_implicit_fb_sync_format() to pick up the audioformat that is assigned in the counter-part substreams primarily, so that the actual capture stream can be opened properly. We keep the same audioformat object only as a fallback in case nothing found, though. Fixes: 9fddc15e8039 ("ALSA: usb-audio: Factor out the implicit feedback quirk code") Link: https://lore.kernel.org/r/20210108075219.21463-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/implicit.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 931042a6a051..9724efe1cdce 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -378,20 +378,19 @@ snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip, int stream) { struct snd_usb_substream *subs; - const struct audioformat *fp, *sync_fmt; + const struct audioformat *fp, *sync_fmt = NULL; int score, high_score; - /* When sharing the same altset, use the original audioformat */ + /* Use the original audioformat as fallback for the shared altset */ if (target->iface == target->sync_iface && target->altsetting == target->sync_altsetting) - return target; + sync_fmt = target; subs = find_matching_substream(chip, stream, target->sync_ep, target->fmt_type); if (!subs) - return NULL; + return sync_fmt; - sync_fmt = NULL; high_score = 0; list_for_each_entry(fp, &subs->fmt_list, list) { score = match_endpoint_audioformats(subs, fp, From 00272c61827e37bb64c47499843d8c0d8ee136a5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Jan 2021 08:52:17 +0100 Subject: [PATCH 03/20] ALSA: usb-audio: Avoid unnecessary interface re-setup The current endpoint handling assumed (more or less) a unique 1:1 relation between the endpoint and the iface/altset. The exception was the sync EP without the implicit feedback which has usually the secondary EP of the same altset. This works fine for most devices, but it turned out that some unusual devices like Pinoeer's ones have both playback and capture endpoints in the same iface/altsetting and use both for the implicit feedback mode. For handling such a case, we need to extend the endpoint management to take the shared interface into account. This patch does that: it adds a new object snd_usb_iface_ref for managing the reference counts of the each USB interface that is used by each endpoint. The interface setup is performed only once for the (sharing) endpoints, and the doubly initialization is avoided. Along with this, the resource release of endpoints and interface refcounts are put into a single function, snd_usb_endpoint_free_all() instead of looping in the caller side. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Link: https://lore.kernel.org/r/20210108075219.21463-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.c | 5 ++- sound/usb/card.h | 2 ++ sound/usb/endpoint.c | 82 ++++++++++++++++++++++++++++++++++++++------ sound/usb/endpoint.h | 2 +- sound/usb/usbaudio.h | 1 + 5 files changed, 77 insertions(+), 15 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index d731ca62d599..e08fbf8e3ee0 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -450,10 +450,8 @@ lookup_device_name(u32 id) static void snd_usb_audio_free(struct snd_card *card) { struct snd_usb_audio *chip = card->private_data; - struct snd_usb_endpoint *ep, *n; - list_for_each_entry_safe(ep, n, &chip->ep_list, list) - snd_usb_endpoint_free(ep); + snd_usb_endpoint_free_all(chip); mutex_destroy(&chip->mutex); if (!atomic_read(&chip->shutdown)) @@ -611,6 +609,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, chip->usb_id = usb_id; INIT_LIST_HEAD(&chip->pcm_list); INIT_LIST_HEAD(&chip->ep_list); + INIT_LIST_HEAD(&chip->iface_ref_list); INIT_LIST_HEAD(&chip->midi_list); INIT_LIST_HEAD(&chip->mixer_list); diff --git a/sound/usb/card.h b/sound/usb/card.h index 6a027c349194..de0d2aa883fa 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -42,6 +42,7 @@ struct audioformat { }; struct snd_usb_substream; +struct snd_usb_iface_ref; struct snd_usb_endpoint; struct snd_usb_power_domain; @@ -58,6 +59,7 @@ struct snd_urb_ctx { struct snd_usb_endpoint { struct snd_usb_audio *chip; + struct snd_usb_iface_ref *iface_ref; int opened; /* open refcount; protect with chip->mutex */ atomic_t running; /* running status */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 162da7a50046..ae6276aded91 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -24,6 +24,14 @@ #define EP_FLAG_RUNNING 1 #define EP_FLAG_STOPPING 2 +/* interface refcounting */ +struct snd_usb_iface_ref { + unsigned char iface; + bool need_setup; + int opened; + struct list_head list; +}; + /* * snd_usb_endpoint is a model that abstracts everything related to an * USB endpoint and its streaming. @@ -488,6 +496,28 @@ exit_clear: clear_bit(ctx->index, &ep->active_mask); } +/* + * Find or create a refcount object for the given interface + * + * The objects are released altogether in snd_usb_endpoint_free_all() + */ +static struct snd_usb_iface_ref * +iface_ref_find(struct snd_usb_audio *chip, int iface) +{ + struct snd_usb_iface_ref *ip; + + list_for_each_entry(ip, &chip->iface_ref_list, list) + if (ip->iface == iface) + return ip; + + ip = kzalloc(sizeof(*ip), GFP_KERNEL); + if (!ip) + return NULL; + ip->iface = iface; + list_add_tail(&ip->list, &chip->iface_ref_list); + return ip; +} + /* * Get the existing endpoint object corresponding EP * Returns NULL if not present. @@ -520,8 +550,8 @@ snd_usb_get_endpoint(struct snd_usb_audio *chip, int ep_num) * * Returns zero on success or a negative error code. * - * New endpoints will be added to chip->ep_list and must be freed by - * calling snd_usb_endpoint_free(). + * New endpoints will be added to chip->ep_list and freed by + * calling snd_usb_endpoint_free_all(). * * For SND_USB_ENDPOINT_TYPE_SYNC, the caller needs to guarantee that * bNumEndpoints > 1 beforehand. @@ -658,6 +688,12 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, usb_audio_dbg(chip, "Open EP 0x%x, iface=%d:%d, idx=%d\n", ep_num, ep->iface, ep->altsetting, ep->ep_idx); + ep->iface_ref = iface_ref_find(chip, ep->iface); + if (!ep->iface_ref) { + ep = NULL; + goto unlock; + } + ep->cur_audiofmt = fp; ep->cur_channels = fp->channels; ep->cur_rate = params_rate(params); @@ -681,6 +717,11 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, ep->implicit_fb_sync); } else { + if (WARN_ON(!ep->iface_ref)) { + ep = NULL; + goto unlock; + } + if (!endpoint_compatible(ep, fp, params)) { usb_audio_err(chip, "Incompatible EP setup for 0x%x\n", ep_num); @@ -692,6 +733,9 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, ep_num, ep->opened); } + if (!ep->iface_ref->opened++) + ep->iface_ref->need_setup = true; + ep->opened++; unlock: @@ -760,12 +804,16 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, mutex_lock(&chip->mutex); usb_audio_dbg(chip, "Closing EP 0x%x (count %d)\n", ep->ep_num, ep->opened); - if (!--ep->opened) { + + if (!--ep->iface_ref->opened) endpoint_set_interface(chip, ep, false); + + if (!--ep->opened) { ep->iface = 0; ep->altsetting = 0; ep->cur_audiofmt = NULL; ep->cur_rate = 0; + ep->iface_ref = NULL; usb_audio_dbg(chip, "EP 0x%x closed\n", ep->ep_num); } mutex_unlock(&chip->mutex); @@ -775,6 +823,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep) { ep->need_setup = true; + if (ep->iface_ref) + ep->iface_ref->need_setup = true; } /* @@ -1195,11 +1245,13 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, int err = 0; mutex_lock(&chip->mutex); + if (WARN_ON(!ep->iface_ref)) + goto unlock; if (!ep->need_setup) goto unlock; - /* No need to (re-)configure the sync EP belonging to the same altset */ - if (ep->ep_idx) { + /* If the interface has been already set up, just set EP parameters */ + if (!ep->iface_ref->need_setup) { err = snd_usb_endpoint_set_params(chip, ep); if (err < 0) goto unlock; @@ -1242,6 +1294,8 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, goto unlock; } + ep->iface_ref->need_setup = false; + done: ep->need_setup = false; err = 1; @@ -1387,15 +1441,21 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep) } /** - * snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint + * snd_usb_endpoint_free_all: Free the resources of an snd_usb_endpoint + * @card: The chip * - * @ep: the endpoint to free - * - * This free all resources of the given ep. + * This free all endpoints and those resources */ -void snd_usb_endpoint_free(struct snd_usb_endpoint *ep) +void snd_usb_endpoint_free_all(struct snd_usb_audio *chip) { - kfree(ep); + struct snd_usb_endpoint *ep, *en; + struct snd_usb_iface_ref *ip, *in; + + list_for_each_entry_safe(ep, en, &chip->ep_list, list) + kfree(ep); + + list_for_each_entry_safe(ip, in, &chip->iface_ref_list, list) + kfree(ip); } /* diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 11e3bb839fd7..eea4ca49876d 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -42,7 +42,7 @@ void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep); int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); -void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); +void snd_usb_endpoint_free_all(struct snd_usb_audio *chip); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 980287aadd36..215c1771dd57 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -44,6 +44,7 @@ struct snd_usb_audio { struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ + struct list_head iface_ref_list; /* list of interface refcounts */ int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ From eae4d054f909d9e9589d0940f9b5b0cd68de1e2e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Jan 2021 08:52:18 +0100 Subject: [PATCH 04/20] ALSA: usb-audio: Annotate the endpoint index in audioformat There are devices that have multiple endpoints sharing the same iface/altset not only for sync but also for the actual streams, and the audioformat for such an endpoint needs to be handled with the proper endpoint index; otherwise it confuses the endpoint management. This patch extends the audioformat to annotate the endpoint index, and put the proper ep_idx=1 to Pioneer device quirk entries accordingly. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Link: https://lore.kernel.org/r/20210108075219.21463-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.h | 1 + sound/usb/endpoint.c | 2 +- sound/usb/quirks-table.h | 6 ++++++ sound/usb/quirks.c | 4 ++-- 4 files changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index de0d2aa883fa..37091b117614 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -18,6 +18,7 @@ struct audioformat { unsigned int frame_size; /* samples per frame for non-audio */ unsigned char iface; /* interface number */ unsigned char altsetting; /* corresponding alternate setting */ + unsigned char ep_idx; /* endpoint array index */ unsigned char altset_idx; /* array index of altenate setting */ unsigned char attributes; /* corresponding attributes of cs endpoint */ unsigned char endpoint; /* endpoint */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index ae6276aded91..fe73fe3ff2bc 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -683,7 +683,7 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, } else { ep->iface = fp->iface; ep->altsetting = fp->altsetting; - ep->ep_idx = 0; + ep->ep_idx = fp->ep_idx; } usb_audio_dbg(chip, "Open EP 0x%x, iface=%d:%d, idx=%d\n", ep_num, ep->iface, ep->altsetting, ep->ep_idx); diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 0e11cb96fa8c..c8a4bdf18207 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3362,6 +3362,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .altsetting = 1, .altset_idx = 1, .endpoint = 0x86, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC| USB_ENDPOINT_SYNC_ASYNC| USB_ENDPOINT_USAGE_IMPLICIT_FB, @@ -3450,6 +3451,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .altsetting = 1, .altset_idx = 1, .endpoint = 0x82, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC| USB_ENDPOINT_SYNC_ASYNC| USB_ENDPOINT_USAGE_IMPLICIT_FB, @@ -3506,6 +3508,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .altsetting = 1, .altset_idx = 1, .endpoint = 0x82, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC| USB_ENDPOINT_SYNC_ASYNC| USB_ENDPOINT_USAGE_IMPLICIT_FB, @@ -3562,6 +3565,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .altsetting = 1, .altset_idx = 1, .endpoint = 0x82, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC| USB_ENDPOINT_SYNC_ASYNC| USB_ENDPOINT_USAGE_IMPLICIT_FB, @@ -3619,6 +3623,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .altsetting = 1, .altset_idx = 1, .endpoint = 0x82, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC| USB_ENDPOINT_SYNC_ASYNC| USB_ENDPOINT_USAGE_IMPLICIT_FB, @@ -3679,6 +3684,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .altsetting = 1, .altset_idx = 1, .endpoint = 0x82, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC| USB_ENDPOINT_SYNC_ASYNC| USB_ENDPOINT_USAGE_IMPLICIT_FB, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b70e2ebc3e29..89e172642d98 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -194,7 +194,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, } alts = &iface->altsetting[fp->altset_idx]; altsd = get_iface_desc(alts); - if (altsd->bNumEndpoints < 1) { + if (altsd->bNumEndpoints <= fp->ep_idx) { err = -EINVAL; goto error; } @@ -204,7 +204,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, if (fp->datainterval == 0) fp->datainterval = snd_usb_parse_datainterval(chip, alts); if (fp->maxpacksize == 0) - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, fp->ep_idx)->wMaxPacketSize); if (!fp->fmt_type) fp->fmt_type = UAC_FORMAT_TYPE_I; From 167c9dc84ec384c0940359e067301883ad2b42a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Jan 2021 08:52:19 +0100 Subject: [PATCH 05/20] ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Pioneer devices have both playback and capture streams sharing the same iface/altsetting, and those need to be paired as implicit feedback. Instead of a half-baked (and broken) static quirk entry, set up more generically for those devices by checking the number of endpoints and the attribute of the secondary EP. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Reported-by: František Kučera Link: https://lore.kernel.org/r/20210108075219.21463-6-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/implicit.c | 48 +++++++++++++++++++++++++++++++++++--------- 1 file changed, 38 insertions(+), 10 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 9724efe1cdce..1ac2cc6c33fb 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -58,8 +58,6 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { IMPLICIT_FB_FIXED_DEV(0x0499, 0x172f, 0x81, 2), /* Steinberg UR22C */ IMPLICIT_FB_FIXED_DEV(0x0d9a, 0x00df, 0x81, 2), /* RTX6001 */ IMPLICIT_FB_FIXED_DEV(0x22f0, 0x0006, 0x81, 3), /* Allen&Heath Qu-16 */ - IMPLICIT_FB_FIXED_DEV(0x2b73, 0x000a, 0x82, 0), /* Pioneer DJ DJM-900NXS2 */ - IMPLICIT_FB_FIXED_DEV(0x2b73, 0x0017, 0x82, 0), /* Pioneer DJ DJM-250MK2 */ IMPLICIT_FB_FIXED_DEV(0x1686, 0xf029, 0x82, 2), /* Zoom UAC-2 */ IMPLICIT_FB_FIXED_DEV(0x2466, 0x8003, 0x86, 2), /* Fractal Audio Axe-Fx II */ IMPLICIT_FB_FIXED_DEV(0x0499, 0x172a, 0x86, 2), /* Yamaha MODX */ @@ -100,7 +98,7 @@ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { /* set up sync EP information on the audioformat */ static int add_implicit_fb_sync_ep(struct snd_usb_audio *chip, struct audioformat *fmt, - int ep, int ifnum, + int ep, int ep_idx, int ifnum, const struct usb_host_interface *alts) { struct usb_interface *iface; @@ -115,7 +113,7 @@ static int add_implicit_fb_sync_ep(struct snd_usb_audio *chip, fmt->sync_ep = ep; fmt->sync_iface = ifnum; fmt->sync_altsetting = alts->desc.bAlternateSetting; - fmt->sync_ep_idx = 0; + fmt->sync_ep_idx = ep_idx; fmt->implicit_fb = 1; usb_audio_dbg(chip, "%d:%d: added %s implicit_fb sync_ep %x, iface %d:%d\n", @@ -147,7 +145,7 @@ static int add_generic_uac2_implicit_fb(struct snd_usb_audio *chip, (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != USB_ENDPOINT_USAGE_IMPLICIT_FB) return 0; - return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, + return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 0, ifnum, alts); } @@ -173,10 +171,32 @@ static int add_roland_implicit_fb(struct snd_usb_audio *chip, (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != USB_ENDPOINT_USAGE_IMPLICIT_FB) return 0; - return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, + return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 0, ifnum, alts); } +/* Pioneer devices: playback and capture streams sharing the same iface/altset + */ +static int add_pioneer_implicit_fb(struct snd_usb_audio *chip, + struct audioformat *fmt, + struct usb_host_interface *alts) +{ + struct usb_endpoint_descriptor *epd; + + if (alts->desc.bNumEndpoints != 2) + return 0; + + epd = get_endpoint(alts, 1); + if (!usb_endpoint_is_isoc_in(epd) || + (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC || + ((epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != + USB_ENDPOINT_USAGE_DATA && + (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != + USB_ENDPOINT_USAGE_IMPLICIT_FB)) + return 0; + return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 1, + alts->desc.bInterfaceNumber, alts); +} static int __add_generic_implicit_fb(struct snd_usb_audio *chip, struct audioformat *fmt, @@ -197,7 +217,7 @@ static int __add_generic_implicit_fb(struct snd_usb_audio *chip, if (!usb_endpoint_is_isoc_in(epd) || (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC) return 0; - return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, + return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 0, iface, alts); } @@ -250,7 +270,7 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, case IMPLICIT_FB_NONE: return 0; /* No quirk */ case IMPLICIT_FB_FIXED: - return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, + return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0, p->iface, NULL); } } @@ -278,6 +298,14 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, return 1; } + /* Pioneer devices implicit feedback 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 */) { + if (add_pioneer_implicit_fb(chip, fmt, alts)) + return 1; + } + /* Try the generic implicit fb if available */ if (chip->generic_implicit_fb) return add_generic_implicit_fb(chip, fmt, alts); @@ -295,8 +323,8 @@ static int audioformat_capture_quirk(struct snd_usb_audio *chip, p = find_implicit_fb_entry(chip, capture_implicit_fb_quirks, alts); if (p && p->type == IMPLICIT_FB_FIXED) - return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, p->iface, - NULL); + return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0, + p->iface, NULL); return 0; } From 3e096a2112b7b407549020cf095e2a425f00fabb Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jonathan=20Neusch=C3=A4fer?= Date: Fri, 1 Jan 2021 23:19:42 +0100 Subject: [PATCH 06/20] ALSA: doc: Fix reference to mixart.rst MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit MIXART.txt has been converted to ReST and renamed. Fix the reference in alsa-configuration.rst. Fixes: 3d8e81862ce4 ("ALSA: doc: ReSTize MIXART.txt") Signed-off-by: Jonathan Neuschäfer Cc: Link: https://lore.kernel.org/r/20210101221942.1068388-1-j.neuschaefer@gmx.net Signed-off-by: Takashi Iwai --- Documentation/sound/alsa-configuration.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index fe52c314b763..b36af65a08ed 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -1501,7 +1501,7 @@ Module for Digigram miXart8 sound cards. This module supports multiple cards. Note: One miXart8 board will be represented as 4 alsa cards. -See MIXART.txt for details. +See Documentation/sound/cards/mixart.rst for details. When the driver is compiled as a module and the hotplug firmware is supported, the firmware data is loaded via hotplug automatically. From f4eccc7fea203cfb35205891eced1ab51836f362 Mon Sep 17 00:00:00 2001 From: Peter Geis Date: Fri, 8 Jan 2021 13:59:12 +0000 Subject: [PATCH 07/20] clk: tegra30: Add hda clock default rates to clock driver Current implementation defaults the hda clocks to clk_m. This causes hda to run too slow to operate correctly. Fix this by defaulting to pll_p and setting the frequency to the correct rate. This matches upstream t124 and downstream t30. Acked-by: Jon Hunter Tested-by: Ion Agorria Acked-by: Sameer Pujar Acked-by: Thierry Reding Signed-off-by: Peter Geis Link: https://lore.kernel.org/r/20210108135913.2421585-2-pgwipeout@gmail.com Signed-off-by: Takashi Iwai --- drivers/clk/tegra/clk-tegra30.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/clk/tegra/clk-tegra30.c b/drivers/clk/tegra/clk-tegra30.c index 37244a7e68c2..9cf249c344d9 100644 --- a/drivers/clk/tegra/clk-tegra30.c +++ b/drivers/clk/tegra/clk-tegra30.c @@ -1256,6 +1256,8 @@ static struct tegra_clk_init_table init_table[] __initdata = { { TEGRA30_CLK_I2S3_SYNC, TEGRA30_CLK_CLK_MAX, 24000000, 0 }, { TEGRA30_CLK_I2S4_SYNC, TEGRA30_CLK_CLK_MAX, 24000000, 0 }, { TEGRA30_CLK_VIMCLK_SYNC, TEGRA30_CLK_CLK_MAX, 24000000, 0 }, + { TEGRA30_CLK_HDA, TEGRA30_CLK_PLL_P, 102000000, 0 }, + { TEGRA30_CLK_HDA2CODEC_2X, TEGRA30_CLK_PLL_P, 48000000, 0 }, /* must be the last entry */ { TEGRA30_CLK_CLK_MAX, TEGRA30_CLK_CLK_MAX, 0, 0 }, }; From 615d435400435876ac68c1de37e9526a9164eaec Mon Sep 17 00:00:00 2001 From: Peter Geis Date: Fri, 8 Jan 2021 13:59:13 +0000 Subject: [PATCH 08/20] ALSA: hda/tegra: fix tegra-hda on tegra30 soc Currently hda on tegra30 fails to open a stream with an input/output error. For example: speaker-test -Dhw:0,3 -c 2 speaker-test 1.2.2 Playback device is hw:0,3 Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 16384 Period size range from 32 to 8192 Using max buffer size 16384 Periods = 4 was set period_size = 4096 was set buffer_size = 16384 0 - Front Left Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Input/output error The tegra-hda device was introduced in tegra30 but only utilized in tegra124 until recent chips. Tegra210/186 work only due to a hardware change. For this reason it is unknown when this issue first manifested. Discussions with the hardware team show this applies to all current tegra chips. It has been resolved in the tegra234, which does not have hda support at this time. The explanation from the hardware team is this: Below is the striping formula referenced from HD audio spec. { ((num_channels * bits_per_sample) / number of SDOs) >= 8 } The current issue is seen because Tegra HW has a problem with boundary condition (= 8) for striping. The reason why it is not seen on Tegra210/Tegra186 is because it uses max 2SDO lines. Max SDO lines is read from GCAP register. For the given stream (channels = 2, bps = 16); ratio = (channels * bps) / NSDO = 32 / NSDO; On Tegra30, ratio = 32/4 = 8 (FAIL) On Tegra210/186, ratio = 32/2 = 16 (PASS) On Tegra194, ratio = 32/4 = 8 (FAIL) ==> Earlier workaround was applied for it If Tegra210/186 is forced to use 4SDO, it fails there as well. So the behavior is consistent across all these chips. Applying the fix in [1] universally resolves this issue on tegra30-hda. Tested on the Ouya game console and the tf201 tablet. [1] commit 60019d8c650d ("ALSA: hda/tegra: workaround playback failure on Tegra194") Reviewed-by: Jon Hunter Tested-by: Ion Agorria Reviewed-by: Sameer Pujar Acked-by: Thierry Reding Signed-off-by: Peter Geis Link: https://lore.kernel.org/r/20210108135913.2421585-3-pgwipeout@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_tegra.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 70164d1428d4..361cf2041911 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -388,7 +388,7 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) * in powers of 2, next available ratio is 16 which can be * used as a limiting factor here. */ - if (of_device_is_compatible(np, "nvidia,tegra194-hda")) + if (of_device_is_compatible(np, "nvidia,tegra30-hda")) chip->bus.core.sdo_limit = 16; /* codec detection */ From e7c22eeaff8565d9a8374f320238c251ca31480b Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Mon, 11 Jan 2021 14:02:50 +0100 Subject: [PATCH 09/20] ALSA: fireface: Fix integer overflow in transmit_midi_msg() As snd_ff.rx_bytes[] is unsigned int, and NSEC_PER_SEC is 1000000000L, the second multiplication in ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250 always overflows on 32-bit platforms, truncating the result. Fix this by precalculating "NSEC_PER_SEC / 31250", which is an integer constant. Note that this assumes ff->rx_bytes[port] <= 16777. Fixes: 19174295788de77d ("ALSA: fireface: add transaction support") Reviewed-by: Takashi Sakamoto Signed-off-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/20210111130251.361335-2-geert+renesas@glider.be Signed-off-by: Takashi Iwai --- sound/firewire/fireface/ff-transaction.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff-transaction.c b/sound/firewire/fireface/ff-transaction.c index 7f82762ccc8c..ee7122c461d4 100644 --- a/sound/firewire/fireface/ff-transaction.c +++ b/sound/firewire/fireface/ff-transaction.c @@ -88,7 +88,7 @@ static void transmit_midi_msg(struct snd_ff *ff, unsigned int port) /* Set interval to next transaction. */ ff->next_ktime[port] = ktime_add_ns(ktime_get(), - ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250); + ff->rx_bytes[port] * 8 * (NSEC_PER_SEC / 31250)); if (quad_count == 1) tcode = TCODE_WRITE_QUADLET_REQUEST; From 9f65df9c589f249435255da37a5dd11f1bc86f4d Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Mon, 11 Jan 2021 14:02:51 +0100 Subject: [PATCH 10/20] ALSA: firewire-tascam: Fix integer overflow in midi_port_work() As snd_fw_async_midi_port.consume_bytes is unsigned int, and NSEC_PER_SEC is 1000000000L, the second multiplication in port->consume_bytes * 8 * NSEC_PER_SEC / 31250 always overflows on 32-bit platforms, truncating the result. Fix this by precalculating "NSEC_PER_SEC / 31250", which is an integer constant. Note that this assumes port->consume_bytes <= 16777. Fixes: 531f471834227d03 ("ALSA: firewire-lib/firewire-tascam: localize async midi port") Reviewed-by: Takashi Sakamoto Signed-off-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/20210111130251.361335-3-geert+renesas@glider.be Signed-off-by: Takashi Iwai --- sound/firewire/tascam/tascam-transaction.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c index 90288b4b4637..a073cece4a7d 100644 --- a/sound/firewire/tascam/tascam-transaction.c +++ b/sound/firewire/tascam/tascam-transaction.c @@ -209,7 +209,7 @@ static void midi_port_work(struct work_struct *work) /* Set interval to next transaction. */ port->next_ktime = ktime_add_ns(ktime_get(), - port->consume_bytes * 8 * NSEC_PER_SEC / 31250); + port->consume_bytes * 8 * (NSEC_PER_SEC / 31250)); /* Start this transaction. */ port->idling = false; From 20c7842ed8374e1c3ee750b2fe7ca8cdd071bda6 Mon Sep 17 00:00:00 2001 From: Alex Deucher Date: Tue, 5 Jan 2021 12:52:45 -0500 Subject: [PATCH 11/20] ALSA: hda/hdmi - enable runtime pm for CI AMD display audio We are able to power down the GPU and audio via the GPU driver so flag these asics as supporting runtime pm. Reviewed-by: Evan Quan Signed-off-by: Alex Deucher Link: https://lore.kernel.org/r/20210105175245.963451-1-alexander.deucher@amd.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 770ad25f1907..e4dd2ff5e473 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2598,7 +2598,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0x1308), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x157a), @@ -2660,9 +2661,11 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xaab0), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaac0), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0xaac8), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0xaad8), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, From 5e941fc033e411118fb3a7d9e0b97f8cf702cd39 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Wed, 13 Jan 2021 17:56:29 +0200 Subject: [PATCH 12/20] ALSA: hda: Add AlderLake-P PCI ID and HDMI codec vid Add HD Audio PCI ID and HDMI codec vendor ID for Intel AlderLake-P. Signed-off-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Guennadi Liakhovetski Link: https://lore.kernel.org/r/20210113155629.4097057-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e4dd2ff5e473..8d568277088a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2507,6 +2507,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-S */ { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-P */ + { PCI_DEVICE(0x8086, 0x51c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 74d246a0dc6d..97adff0cbcab 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4346,6 +4346,7 @@ HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), From e4ea77f8e53f9accb9371fba34c189d0447ecce0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 11 Jan 2021 09:16:11 +0100 Subject: [PATCH 13/20] ALSA: usb-audio: Always apply the hw constraints for implicit fb sync Since the commit 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync"), we apply the hw constraints for the implicit feedback sync to make the secondary open aligned with the already opened stream setup. This change assumed that the secondary open is performed after the first stream has been already set up, and adds the hw constraints to sync with the first stream's parameters only when the EP setup for the first stream was confirmed at the open time. However, most of applications handling the full-duplex operations do open both playback and capture streams at first, then set up both streams. This results in skipping the additional hw constraints since the counter-part stream hasn't been set up yet at the open of the second stream, and it eventually leads to "incompatible EP" error in the end. This patch corrects the behavior by always applying the hw constraints for the implicit fb sync. The hw constraint rules are defined so that they check the sync EP dynamically at each invocation, instead. This covers the concurrent stream setups better and lets the hw refine calls resolving to the right configuration. Also this patch corrects a minor error that has existed in the debug print that isn't built as default. Fixes: 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync") Link: https://lore.kernel.org/r/20210111081611.12790-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 171 ++++++++++++++++++++++++++++++------------------ 1 file changed, 108 insertions(+), 63 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 56079901769f..f71965bf815f 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -663,7 +663,7 @@ static int hw_check_valid_format(struct snd_usb_substream *subs, check_fmts.bits[1] = (u32)(fp->formats >> 32); snd_mask_intersect(&check_fmts, fmts); if (snd_mask_empty(&check_fmts)) { - hwc_debug(" > check: no supported format %d\n", fp->format); + hwc_debug(" > check: no supported format 0x%llx\n", fp->formats); return 0; } /* check the channels */ @@ -775,24 +775,11 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, return apply_hw_params_minmax(it, rmin, rmax); } -static int hw_rule_format(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int apply_hw_params_format_bits(struct snd_mask *fmt, u64 fbits) { - struct snd_usb_substream *subs = rule->private; - const struct audioformat *fp; - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - u64 fbits; u32 oldbits[2]; int changed; - hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]); - fbits = 0; - list_for_each_entry(fp, &subs->fmt_list, list) { - if (!hw_check_valid_format(subs, params, fp)) - continue; - fbits |= fp->formats; - } - oldbits[0] = fmt->bits[0]; oldbits[1] = fmt->bits[1]; fmt->bits[0] &= (u32)fbits; @@ -806,6 +793,24 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, return changed; } +static int hw_rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct audioformat *fp; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + u64 fbits; + + hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]); + fbits = 0; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (!hw_check_valid_format(subs, params, fp)) + continue; + fbits |= fp->formats; + } + return apply_hw_params_format_bits(fmt, fbits); +} + static int hw_rule_period_time(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -833,64 +838,92 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params, return apply_hw_params_minmax(it, pmin, UINT_MAX); } -/* apply PCM hw constraints from the concurrent sync EP */ -static int apply_hw_constraint_from_sync(struct snd_pcm_runtime *runtime, - struct snd_usb_substream *subs) +/* get the EP or the sync EP for implicit fb when it's already set up */ +static const struct snd_usb_endpoint * +get_sync_ep_from_substream(struct snd_usb_substream *subs) { struct snd_usb_audio *chip = subs->stream->chip; - struct snd_usb_endpoint *ep; const struct audioformat *fp; - int err; + const struct snd_usb_endpoint *ep; list_for_each_entry(fp, &subs->fmt_list, list) { ep = snd_usb_get_endpoint(chip, fp->endpoint); if (ep && ep->cur_rate) - goto found; + return ep; if (!fp->implicit_fb) continue; /* for the implicit fb, check the sync ep as well */ ep = snd_usb_get_endpoint(chip, fp->sync_ep); if (ep && ep->cur_rate) - goto found; + return ep; } - return 0; + return NULL; +} - found: - if (!find_format(&subs->fmt_list, ep->cur_format, ep->cur_rate, - ep->cur_channels, false, NULL)) { - usb_audio_dbg(chip, "EP 0x%x being used, but not applicable\n", - ep->ep_num); +/* additional hw constraints for implicit feedback mode */ +static int hw_rule_format_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + ep = get_sync_ep_from_substream(subs); + if (!ep) return 0; - } - usb_audio_dbg(chip, "EP 0x%x being used, using fixed params:\n", - ep->ep_num); - usb_audio_dbg(chip, "rate=%d, period_size=%d, periods=%d\n", - ep->cur_rate, ep->cur_period_frames, - ep->cur_buffer_periods); + hwc_debug("applying %s\n", __func__); + return apply_hw_params_format_bits(fmt, pcm_format_to_bits(ep->cur_format)); +} - runtime->hw.formats = subs->formats; - runtime->hw.rate_min = runtime->hw.rate_max = ep->cur_rate; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - runtime->hw.periods_min = runtime->hw.periods_max = - ep->cur_buffer_periods; +static int hw_rule_rate_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_RATE, - -1); - if (err < 0) - return err; + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; - err = snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - ep->cur_period_frames, - ep->cur_period_frames); - if (err < 0) - return err; + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + return apply_hw_params_minmax(it, ep->cur_rate, ep->cur_rate); +} - return 1; /* notify the finding */ +static int hw_rule_period_size_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; + + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; + + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + return apply_hw_params_minmax(it, ep->cur_period_frames, + ep->cur_period_frames); +} + +static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; + + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; + + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIODS); + return apply_hw_params_minmax(it, ep->cur_buffer_periods, + ep->cur_buffer_periods); } /* @@ -899,20 +932,11 @@ static int apply_hw_constraint_from_sync(struct snd_pcm_runtime *runtime, static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) { - struct snd_usb_audio *chip = subs->stream->chip; const struct audioformat *fp; unsigned int pt, ptmin; int param_period_time_if_needed = -1; int err; - mutex_lock(&chip->mutex); - err = apply_hw_constraint_from_sync(runtime, subs); - mutex_unlock(&chip->mutex); - if (err < 0) - return err; - if (err > 0) /* found the matching? */ - goto add_extra_rules; - runtime->hw.formats = subs->formats; runtime->hw.rate_min = 0x7fffffff; @@ -964,7 +988,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err; -add_extra_rules: err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, subs, SNDRV_PCM_HW_PARAM_FORMAT, @@ -993,6 +1016,28 @@ add_extra_rules: return err; } + /* additional hw constraints for implicit fb */ + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + hw_rule_period_size_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, + hw_rule_periods_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_PERIODS, -1); + if (err < 0) + return err; + return 0; } From 495dc7637cb5ca8e39c46db818328410bb6e73a1 Mon Sep 17 00:00:00 2001 From: Chris Chiu Date: Thu, 14 Jan 2021 16:27:28 +0800 Subject: [PATCH 14/20] ALSA: hda/realtek - Limit int mic boost on Acer Aspire E5-575T The Acer Apire E5-575T laptop with codec ALC255 has a terrible background noise comes from internal mic capture. And the jack sensing dose not work for headset like some other Acer laptops. This patch limits the internal mic boost on top of the existing ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk for Acer Aspire E5-575T. Signed-off-by: Chris Chiu Cc: Link: https://lore.kernel.org/r/20210114082728.74729-1-chiu@endlessos.org Signed-off-by: Takashi Iwai --- 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 dd82ff2bd5d6..ed5b6b894dc1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6371,6 +6371,7 @@ enum { ALC256_FIXUP_HP_HEADSET_MIC, ALC236_FIXUP_DELL_AIO_HEADSET_MIC, ALC282_FIXUP_ACER_DISABLE_LINEOUT, + ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -7808,6 +7809,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7826,6 +7833,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1025, 0x1094, "Acer Aspire E5-575T", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1166, "Acer Veriton N4640G", ALC269_FIXUP_LIFEBOOK), From 67ea698c3950d10925be33c21ca49ffb64e21842 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2021 08:24:53 +0100 Subject: [PATCH 15/20] ALSA: hda/via: Add minimum mute flag It turned out that VIA codecs also mute the sound in the lowest mixer level. Turn on the dac_min_mute flag to indicate the mute-as-minimum in TLV like already done in Conexant and IDT codecs. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210559 Cc: Link: https://lore.kernel.org/r/20210114072453.11379-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 0ab40a8a68fb..834367dd54e1 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -113,6 +113,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->codec_type = VT1708S; spec->gen.indep_hp = 1; spec->gen.keep_eapd_on = 1; + spec->gen.dac_min_mute = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; codec->power_save_node = 1; From 217bfbb8b0bfa24619b11ab75c135fec99b99b20 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Jan 2021 10:34:28 +0100 Subject: [PATCH 16/20] ALSA: seq: oss: Fix missing error check in snd_seq_oss_synth_make_info() snd_seq_oss_synth_make_info() didn't check the error code from snd_seq_oss_midi_make_info(), and this leads to the call of strlcpy() with the uninitialized string as the source, which may lead to the access over the limit. Add the proper error check for avoiding the failure. Reported-by: syzbot+e42504ff21cff05a595f@syzkaller.appspotmail.com Cc: Link: https://lore.kernel.org/r/20210115093428.15882-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_synth.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 11554d0412f0..1b8409ec2c97 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -611,7 +611,8 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in if (info->is_midi) { struct midi_info minf; - snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf); + if (snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf)) + return -ENXIO; inf->synth_type = SYNTH_TYPE_MIDI; inf->synth_subtype = 0; inf->nr_voices = 16; From f84d3a1ec375e46a55cc3ba85c04272b24bd3921 Mon Sep 17 00:00:00 2001 From: Kai-Chuan Hsieh Date: Fri, 15 Jan 2021 11:15:15 +0800 Subject: [PATCH 17/20] ALSA: hda: Add Cometlake-R PCI ID Add HD Audio Device PCI ID for the Intel Cometlake-R platform Reviewed-by: Kai Vehmanen Signed-off-by: Kai-Chuan Hsieh Link: https://lore.kernel.org/r/20210115031515.13100-1-kaichuan.hsieh@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8d568277088a..5a50d3a46445 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2484,6 +2484,9 @@ static const struct pci_device_id azx_ids[] = { /* CometLake-S */ { PCI_DEVICE(0x8086, 0xa3f0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* CometLake-R */ + { PCI_DEVICE(0x8086, 0xf0c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, From 87cb9af9f8a2b242cea7f828206d619e8cbb6a1a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2021 08:58:14 +0100 Subject: [PATCH 18/20] ALSA: usb-audio: Fix UAC1 rate setup for secondary endpoints The current sample rate setup function for UAC1 assumes only the first endpoint retrieved from the interface:altset pair, but the rate set up may be needed also for the secondary endpoint. Also, retrieving the endpoint number from the interface descriptor is redundant; we have already the target endpoint in the given audioformat object. This patch simplifies the code and corrects the target endpoint as described in the above. It simply refers to fmt->endpoint directly. Also, this patch drops the pioneer_djm_set_format_quirk() that is caleld from snd_usb_set_format_quirk(); this function does the sample rate setup but for the capture endpoint (0x82), and that's exactly what the change above fixes. Link: https://lore.kernel.org/r/20210118075816.25068-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 21 ++++++--------------- sound/usb/quirks.c | 28 ---------------------------- 2 files changed, 6 insertions(+), 43 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 31051f2be46d..dc68ed65e478 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -485,18 +485,9 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, const struct audioformat *fmt, int rate) { struct usb_device *dev = chip->dev; - struct usb_host_interface *alts; - unsigned int ep; unsigned char data[3]; int err, crate; - alts = snd_usb_get_host_interface(chip, fmt->iface, fmt->altsetting); - if (!alts) - return -EINVAL; - if (get_iface_desc(alts)->bNumEndpoints < 1) - return -EINVAL; - ep = get_endpoint(alts, 0)->bEndpointAddress; - /* if endpoint doesn't have sampling rate control, bail out */ if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) return 0; @@ -506,11 +497,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, data[2] = rate >> 16; err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data)); + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, + fmt->endpoint, data, sizeof(data)); if (err < 0) { dev_err(&dev->dev, "%d:%d: cannot set freq %d to ep %#x\n", - fmt->iface, fmt->altsetting, rate, ep); + fmt->iface, fmt->altsetting, rate, fmt->endpoint); return err; } @@ -524,11 +515,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data)); + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, + fmt->endpoint, data, sizeof(data)); if (err < 0) { dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", - fmt->iface, fmt->altsetting, ep); + fmt->iface, fmt->altsetting, fmt->endpoint); chip->sample_rate_read_error++; return 0; /* some devices don't support reading */ } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 89e172642d98..e196e364cef1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1470,30 +1470,6 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } - -/* - * Pioneer DJ DJM-900NXS2 - * Device needs to know the sample rate each time substream is started - */ -static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs) -{ - unsigned int cur_rate = subs->data_endpoint->cur_rate; - /* Convert sample rate value to little endian */ - u8 sr[3]; - - sr[0] = cur_rate & 0xff; - sr[1] = (cur_rate >> 8) & 0xff; - sr[2] = (cur_rate >> 16) & 0xff; - - /* Configure device */ - usb_set_interface(subs->dev, 0, 1); - snd_usb_ctl_msg(subs->stream->chip->dev, - usb_rcvctrlpipe(subs->stream->chip->dev, 0), - 0x01, 0x22, 0x0100, 0x0082, &sr, 0x0003); - - return 0; -} - void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1504,10 +1480,6 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; - case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ - case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ - pioneer_djm_set_format_quirk(subs); - break; case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; From 3784d449d795ba11a92681bd22d183329f976421 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2021 08:58:15 +0100 Subject: [PATCH 19/20] ALSA: usb-audio: Set sample rate for all sharing EPs on UAC1 The UAC2/3 sample rate setup is based on the clock node, which is usually shared in the interface, and can't be re-setup without deselecting the interface once, and that's how the current code behaves. OTOH, the sample rate setup of UAC1 is per endpoint, hence we basically need to call for each endpoint usage even if those share the same interface. This patch fixes the behavior of UAC1 to call always snd_usb_init_sample_rate() in snd_usb_endpoint_configure(). Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Link: https://lore.kernel.org/r/20210118075816.25068-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index fe73fe3ff2bc..8e568823c992 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1252,6 +1252,15 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, /* If the interface has been already set up, just set EP parameters */ if (!ep->iface_ref->need_setup) { + /* sample rate setup of UAC1 is per endpoint, and we need + * to update at each EP configuration + */ + if (ep->cur_audiofmt->protocol == UAC_VERSION_1) { + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, + ep->cur_rate); + if (err < 0) + goto unlock; + } err = snd_usb_endpoint_set_params(chip, ep); if (err < 0) goto unlock; From 532a208ad61018b586cebfca8431291fe9c10ce7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2021 08:58:16 +0100 Subject: [PATCH 20/20] ALSA: usb-audio: Avoid implicit feedback on Pioneer devices For addressing the regression on Pioneer devices, we recently corrected the quirk code to enable the implicit feedback mode on those devices properly. However, the devices still showed problems with the full duplex operations with JACK, and after debug sessions, we figured out that the older kernels that had worked with JACK also didn't use the implicit feedback mode at all although they had the quirk code to enable it; instead, the old code worked just to skip the normal sync endpoint setup that would have been detected without it. IOW, what broke without the implicit-fb quirk in the past was the application of the normal sync endpoint that is actually the capture data endpoint on these devices. This patch covers the overseen piece: it modifies the quirk code again not to enable the implicit feedback mode but just to make the driver skipping the sync endpoint detection. This made the driver working with JACK full-duplex mode again. Still it's not quite clear why the implicit feedback doesn't work on those devices yet; maybe it's about some issues in the URB setup. But at least, with this patch, the driver should work in the level of the older kernels again. Fixes: 167c9dc84ec3 ("ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices") Link: https://lore.kernel.org/r/20210118075816.25068-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/implicit.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 1ac2cc6c33fb..521cc846d9d9 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -175,11 +175,13 @@ static int add_roland_implicit_fb(struct snd_usb_audio *chip, ifnum, alts); } -/* Pioneer devices: playback and capture streams sharing the same iface/altset +/* 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. */ -static int add_pioneer_implicit_fb(struct snd_usb_audio *chip, - struct audioformat *fmt, - struct usb_host_interface *alts) +static int skip_pioneer_sync_ep(struct snd_usb_audio *chip, + struct audioformat *fmt, + struct usb_host_interface *alts) { struct usb_endpoint_descriptor *epd; @@ -194,8 +196,7 @@ static int add_pioneer_implicit_fb(struct snd_usb_audio *chip, (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != USB_ENDPOINT_USAGE_IMPLICIT_FB)) return 0; - return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 1, - alts->desc.bInterfaceNumber, alts); + return 1; /* don't handle with the implicit fb, just skip sync EP */ } static int __add_generic_implicit_fb(struct snd_usb_audio *chip, @@ -298,11 +299,11 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, return 1; } - /* Pioneer devices implicit feedback with vendor spec class */ + /* 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 */) { - if (add_pioneer_implicit_fb(chip, fmt, alts)) + if (skip_pioneer_sync_ep(chip, fmt, alts)) return 1; }