From fc90172ba283b598b5497e968de47e3bc2800ed5 Mon Sep 17 00:00:00 2001 From: Andrej Krutak Date: Tue, 29 Nov 2016 22:12:51 +0100 Subject: ALSA: line6: Claim pod x3 usb data interface Userspace apps have to claim USB interfaces before using endpoints in them (drivers/usb/core/devio.c:checkintf()). It's a lock mechanism so that two "drivers" don't steal data from each other. Kernel drivers don't have to claim interfaces to work - but they should, to lock out userspace. While there, fix line6_properties struct to match checkpatch.pl. Signed-off-by: Andrej Krutak Signed-off-by: Takashi Iwai --- sound/usb/line6/driver.h | 9 +++++---- sound/usb/line6/podhd.c | 26 ++++++++++++++++++++++++++ 2 files changed, 31 insertions(+), 4 deletions(-) (limited to 'sound/usb') diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 7e3a3aada222..a5c2e9ae5f17 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -98,10 +98,11 @@ struct line6_properties { int altsetting; - unsigned ep_ctrl_r; - unsigned ep_ctrl_w; - unsigned ep_audio_r; - unsigned ep_audio_w; + unsigned int ctrl_if; + unsigned int ep_ctrl_r; + unsigned int ep_ctrl_w; + unsigned int ep_audio_r; + unsigned int ep_audio_w; }; /* Capability bits */ diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 49cd4a65e390..6ab23e5aee71 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -153,6 +153,7 @@ static struct line6_pcm_properties podx3_pcm_properties = { .rats = &podhd_ratden}, .bytes_per_channel = 3 /* SNDRV_PCM_FMTBIT_S24_3LE */ }; +static struct usb_driver podhd_driver; static void podhd_startup_start_workqueue(unsigned long data); static void podhd_startup_workqueue(struct work_struct *work); @@ -291,8 +292,14 @@ static void podhd_disconnect(struct usb_line6 *line6) struct usb_line6_podhd *pod = (struct usb_line6_podhd *)line6; if (pod->line6.properties->capabilities & LINE6_CAP_CONTROL) { + struct usb_interface *intf; + del_timer_sync(&pod->startup_timer); cancel_work_sync(&pod->startup_work); + + intf = usb_ifnum_to_if(line6->usbdev, + pod->line6.properties->ctrl_if); + usb_driver_release_interface(&podhd_driver, intf); } } @@ -304,10 +311,27 @@ static int podhd_init(struct usb_line6 *line6, { int err; struct usb_line6_podhd *pod = (struct usb_line6_podhd *) line6; + struct usb_interface *intf; line6->disconnect = podhd_disconnect; if (pod->line6.properties->capabilities & LINE6_CAP_CONTROL) { + /* claim the data interface */ + intf = usb_ifnum_to_if(line6->usbdev, + pod->line6.properties->ctrl_if); + if (!intf) { + dev_err(pod->line6.ifcdev, "interface %d not found\n", + pod->line6.properties->ctrl_if); + return -ENODEV; + } + + err = usb_driver_claim_interface(&podhd_driver, intf, NULL); + if (err != 0) { + dev_err(pod->line6.ifcdev, "can't claim interface %d, error %d\n", + pod->line6.properties->ctrl_if, err); + return err; + } + /* create sysfs entries: */ err = snd_card_add_dev_attr(line6->card, &podhd_dev_attr_group); if (err < 0) @@ -406,6 +430,7 @@ static const struct line6_properties podhd_properties_table[] = { .altsetting = 1, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, + .ctrl_if = 1, .ep_audio_r = 0x86, .ep_audio_w = 0x02, }, @@ -417,6 +442,7 @@ static const struct line6_properties podhd_properties_table[] = { .altsetting = 1, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, + .ctrl_if = 1, .ep_audio_r = 0x86, .ep_audio_w = 0x02, }, -- cgit v1.2.3 From 4763601a56f155ddf94ef35fc2c41504a2de15f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Nov 2016 22:28:40 +0100 Subject: ALSA: usb-audio: Fix bogus error return in snd_usb_create_stream() The function returns -EINVAL even if it builds the stream properly. The bogus error code sneaked in during the code refactoring, but it wasn't noticed until now since the returned error code itself is ignored in anyway. Kill it here, but there is no behavior change by this patch, obviously. Fixes: e5779998bf8b ('ALSA: usb-audio: refactor code') Signed-off-by: Takashi Iwai --- sound/usb/card.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/usb') diff --git a/sound/usb/card.c b/sound/usb/card.c index 2ddc034673a8..f36cb068dad3 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -206,7 +206,6 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int if (! snd_usb_parse_audio_interface(chip, interface)) { usb_set_interface(dev, interface, 0); /* reset the current interface */ usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - return -EINVAL; } return 0; -- cgit v1.2.3 From 16200948d8353fe29a473a394d7d26790deae0e7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Dec 2016 11:19:38 +0100 Subject: ALSA: usb-audio: Fix race at stopping the stream We've got a kernel crash report showing like: Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = a1d7c000 [00000008] *pgd=31c93831, *pte=00000000, *ppte=00000000 Internal error: Oops: 17 [#1] PREEMPT SMP ARM CPU: 0 PID: 250 Comm: dbus-daemon Not tainted 3.14.51-03479-gf50bdf4 #1 task: a3ae61c0 ti: a08c8000 task.ti: a08c8000 PC is at retire_capture_urb+0x10/0x1f4 [snd_usb_audio] LR is at snd_complete_urb+0x140/0x1f0 [snd_usb_audio] pc : [<7f0eb22c>] lr : [<7f0e57fc>] psr: 200e0193 sp : a08c9c98 ip : a08c9ce8 fp : a08c9ce4 r10: 0000000a r9 : 00000102 r8 : 94cb3000 r7 : 94cb3000 r6 : 94d0f000 r5 : 94d0e8e8 r4 : 94d0e000 r3 : 7f0eb21c r2 : 00000000 r1 : 94cb3000 r0 : 00000000 Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c5387d Table: 31d7c04a DAC: 00000015 Process dbus-daemon (pid: 250, stack limit = 0xa08c8238) Stack: (0xa08c9c98 to 0xa08ca000) ... Backtrace: [<7f0eb21c>] (retire_capture_urb [snd_usb_audio]) from [<7f0e57fc>] (snd_complete_urb+0x140/0x1f0 [snd_usb_audio]) [<7f0e56bc>] (snd_complete_urb [snd_usb_audio]) from [<80371118>] (__usb_hcd_giveback_urb+0x78/0xf4) [<803710a0>] (__usb_hcd_giveback_urb) from [<80371514>] (usb_giveback_urb_bh+0x8c/0xc0) [<80371488>] (usb_giveback_urb_bh) from [<80028e3c>] (tasklet_hi_action+0xc4/0x148) [<80028d78>] (tasklet_hi_action) from [<80028358>] (__do_softirq+0x190/0x380) [<800281c8>] (__do_softirq) from [<80028858>] (irq_exit+0x8c/0xfc) [<800287cc>] (irq_exit) from [<8000ea88>] (handle_IRQ+0x8c/0xc8) [<8000e9fc>] (handle_IRQ) from [<800085e8>] (gic_handle_irq+0xbc/0xf8) [<8000852c>] (gic_handle_irq) from [<80509044>] (__irq_svc+0x44/0x78) [<80508820>] (_raw_spin_unlock_irq) from [<8004b880>] (finish_task_switch+0x5c/0x100) [<8004b824>] (finish_task_switch) from [<805052f0>] (__schedule+0x48c/0x6d8) [<80504e64>] (__schedule) from [<805055d4>] (schedule+0x98/0x9c) [<8050553c>] (schedule) from [<800116c8>] (do_work_pending+0x30/0xd0) [<80011698>] (do_work_pending) from [<8000e160>] (work_pending+0xc/0x20) Code: e1a0c00d e92ddff0 e24cb004 e24dd024 (e5902008) Kernel panic - not syncing: Fatal exception in interrupt There is a race between retire_capture_urb() and stop_endpoints(). The latter is called at stopping the stream and it sets some endpoint fields to NULL. But its call is asynchronous, thus the pending complete callback might get called after these NULL clears, and it leads the NULL dereference like the above. The fix is to move the NULL clearance after the synchronization, i.e. wait_clear_urbs(). This is called at prepare and hw_free callbacks, so it's assured to be called before the restart of the stream or the release of the stream. Also, while we're at it, put the EP_FLAG_RUNNING flag check at the beginning of snd_complete_urb() to skip the pending complete after the stream is stopped. Fixes: b2eb950de2f0 ("ALSA: usb-audio: stop both data and sync...") Reported-by: Jiada Wang Reported-by: Mark Craske Cc: Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound/usb') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c470251cea4b..f3fce9abece9 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -384,6 +384,9 @@ static void snd_complete_urb(struct urb *urb) if (unlikely(atomic_read(&ep->chip->shutdown))) goto exit_clear; + if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) + goto exit_clear; + if (usb_pipeout(ep->pipe)) { retire_outbound_urb(ep, ctx); /* can be stopped during retire callback */ @@ -534,6 +537,11 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep) alive, ep->ep_num); clear_bit(EP_FLAG_STOPPING, &ep->flags); + ep->data_subs = NULL; + ep->sync_slave = NULL; + ep->retire_data_urb = NULL; + ep->prepare_data_urb = NULL; + return 0; } @@ -1006,10 +1014,6 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep) if (--ep->use_count == 0) { deactivate_urbs(ep, false); - ep->data_subs = NULL; - ep->sync_slave = NULL; - ep->retire_data_urb = NULL; - ep->prepare_data_urb = NULL; set_bit(EP_FLAG_STOPPING, &ep->flags); } } -- cgit v1.2.3 From 1e2e3fe480064ca33186e5a923beaa160efed35d Mon Sep 17 00:00:00 2001 From: Daniel Girnus Date: Tue, 6 Dec 2016 14:46:15 +0900 Subject: ALSA: usb-audio: avoid setting of sample rate multiple times on bus Some of userland applications call 'snd_pcm_hw_params()' and 'snd_pcm_hw_prepare()' sequentially, which means 'snd_pcm_hw_prepare()' is called twice and the second 'snd_pcm_hw_prepare()' is called in 'SNDRV_PCM_STATE_PREPARED' state. Some devices are not able to manage this and they will stop playback if the sample rate will be configured several times over USB protocol. V2: updated Changelog Signed-off-by: Daniel Girnus Signed-off-by: Jens Lorenz Signed-off-by: Jiada Wang Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) (limited to 'sound/usb') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 44d178ee9177..a522c9af1f34 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -806,17 +806,18 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) if (ret < 0) goto unlock; - iface = usb_ifnum_to_if(subs->dev, subs->cur_audiofmt->iface); - alts = &iface->altsetting[subs->cur_audiofmt->altset_idx]; - ret = snd_usb_init_sample_rate(subs->stream->chip, - subs->cur_audiofmt->iface, - alts, - subs->cur_audiofmt, - subs->cur_rate); - if (ret < 0) - goto unlock; - if (subs->need_setup_ep) { + + iface = usb_ifnum_to_if(subs->dev, subs->cur_audiofmt->iface); + alts = &iface->altsetting[subs->cur_audiofmt->altset_idx]; + ret = snd_usb_init_sample_rate(subs->stream->chip, + subs->cur_audiofmt->iface, + alts, + subs->cur_audiofmt, + subs->cur_rate); + if (ret < 0) + goto unlock; + ret = configure_endpoint(subs); if (ret < 0) goto unlock; -- cgit v1.2.3 From fd1a5059610cd3887f1050171a840ca864108730 Mon Sep 17 00:00:00 2001 From: Andreas Pape Date: Tue, 6 Dec 2016 14:46:14 +0900 Subject: ALSA: usb-audio: more tolerant packetsize since commit 57e6dae1087b ("ALSA: usb-audio: do not trust too-big wMaxPacketSize values"), the expected packetsize is always limited to nominal + 25%. It was discovered, that some devices (Android audio accessory) have a much higher jitter in used packetsizes than 25% which would result in BABBLE condition and dropping of packets. A better solution is so assume the jitter to be the nominal packetsize: -one nearly empty packet followed by a almost 150% sized one. V2: changed to assume max frequency is +50 of nominal packetsize. Signed-off-by: Andreas Pape Signed-off-by: Jiada Wang Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/usb') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c470251cea4b..a2931f49a1fc 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -632,8 +632,8 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->stride = frame_bits >> 3; ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; - /* assume max. frequency is 25% higher than nominal */ - ep->freqmax = ep->freqn + (ep->freqn >> 2); + /* assume max. frequency is 50% higher than nominal */ + ep->freqmax = ep->freqn + (ep->freqn >> 1); /* Round up freqmax to nearest integer in order to calculate maximum * packet size, which must represent a whole number of frames. * This is accomplished by adding 0x0.ffff before converting the -- cgit v1.2.3 From 17f08b0d9aafccdb10038ab6dbd9ddb6433c13e2 Mon Sep 17 00:00:00 2001 From: Alberto Aguirre Date: Thu, 8 Dec 2016 00:36:48 -0600 Subject: ALSA: usb-audio: add implicit fb quirk for Axe-Fx II The Axe-Fx II implicit feedback end point and the data sync endpoint are in different interface descriptors. Add quirk to ensure a sync endpoint is properly configured. Signed-off-by: Alberto Aguirre Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/usb') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a522c9af1f34..34c6d4f2c0b6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -348,6 +348,16 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, alts = &iface->altsetting[1]; goto add_sync_ep; + case USB_ID(0x2466, 0x8003): + ep = 0x86; + iface = usb_ifnum_to_if(dev, 2); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && -- cgit v1.2.3 From 82ffb6fc637150b279f49e174166d2aa3853eaf4 Mon Sep 17 00:00:00 2001 From: Con Kolivas Date: Fri, 9 Dec 2016 15:15:57 +1100 Subject: ALSA: usb-audio: Add QuickCam Communicate Deluxe/S7500 to volume_control_quirks The Logitech QuickCam Communicate Deluxe/S7500 microphone fails with the following warning. [ 6.778995] usb 2-1.2.2.2: Warning! Unlikely big volume range (=3072), cval->res is probably wrong. [ 6.778996] usb 2-1.2.2.2: [5] FU [Mic Capture Volume] ch = 1, val = 4608/7680/1 Adding it to the list of devices in volume_control_quirks makes it work properly, fixing related typo. Signed-off-by: Con Kolivas Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/usb') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 2f8c388ef84f..4703caea56b2 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -932,9 +932,10 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x08ca): /* Logitech Quickcam Fusion */ case USB_ID(0x046d, 0x0991): + case USB_ID(0x046d, 0x09a2): /* QuickCam Communicate Deluxe/S7500 */ /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. - * Proboly there is some logitech magic behind this number --fishor + * Probably there is some logitech magic behind this number --fishor */ if (!strcmp(kctl->id.name, "Mic Capture Volume")) { usb_audio_info(chip, -- cgit v1.2.3 From 7f38ca047b0cb54df7f6d9e4110e292e45dba6ad Mon Sep 17 00:00:00 2001 From: Nobutaka Okabe Date: Tue, 13 Dec 2016 01:24:08 +0900 Subject: ALSA: usb-audio: Add native DSD support for TEAC 501/503 DAC This patch adds native DSD support for the following devices. - TEAC NT-503 - TEAC UD-503 - TEAC UD-501 (1) Add quirks for native DSD support for TEAC devices. (2) A specific vendor command is needed to switch between PCM/DOP and DSD mode, same as Denon/Marantz devices. Signed-off-by: Nobutaka Okabe Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 38 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) (limited to 'sound/usb') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2782155ae3ce..b3fd2382fdd9 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1165,6 +1165,18 @@ static bool is_marantz_denon_dac(unsigned int id) return false; } +/* TEAC UD-501/UD-503/NT-503 USB DACs need a vendor cmd to switch + * between PCM/DOP and native DSD mode + */ +static bool is_teac_50X_dac(unsigned int id) +{ + switch (id) { + case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ + return true; + } + return false; +} + int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, struct audioformat *fmt) { @@ -1192,6 +1204,26 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, break; } mdelay(20); + } else if (is_teac_50X_dac(subs->stream->chip->usb_id)) { + /* Vendor mode switch cmd is required. */ + switch (fmt->altsetting) { + case 3: /* DSD mode (DSD_U32) requested */ + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0, + USB_DIR_OUT|USB_TYPE_VENDOR|USB_RECIP_INTERFACE, + 1, 1, NULL, 0); + if (err < 0) + return err; + break; + + case 2: /* PCM or DOP mode (S32) requested */ + case 1: /* PCM mode (S16) requested */ + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0, + USB_DIR_OUT|USB_TYPE_VENDOR|USB_RECIP_INTERFACE, + 0, 1, NULL, 0); + if (err < 0) + return err; + break; + } } return 0; } @@ -1337,5 +1369,11 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; } + /* TEAC devices with USB DAC functionality */ + if (is_teac_50X_dac(chip->usb_id)) { + if (fp->altsetting == 3) + return SNDRV_PCM_FMTBIT_DSD_U32_BE; + } + return 0; } -- cgit v1.2.3 From 012007309133d21a5a7eae3f552c03ac061a2b51 Mon Sep 17 00:00:00 2001 From: Nobutaka Okabe Date: Tue, 13 Dec 2016 02:52:58 +0900 Subject: ALSA: usb-audio: Eliminate noise at the start of DSD playback. [Problem] In some USB DACs, a terrible pop noise comes to be heard at the start of DSD playback (in the following situations). - play first DSD track - change from PCM track to DSD track - change from DSD64 track to DSD128 track (and etc...) - seek DSD track - Fast-Forward/Rewind DSD track [Cause] At the start of playback, there is a little silence. The silence bit pattern "0x69" is required on DSD mode, but it is not like that. [Solution] This patch adds DSD silence pattern to the endpoint settings. Signed-off-by: Nobutaka Okabe Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) (limited to 'sound/usb') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 57b0d9968ec2..a2cdf3370afe 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -638,7 +638,21 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->datainterval = fmt->datainterval; ep->stride = frame_bits >> 3; - ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; + + switch (pcm_format) { + case SNDRV_PCM_FORMAT_U8: + ep->silence_value = 0x80; + break; + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + ep->silence_value = 0x69; + break; + default: + ep->silence_value = 0; + } /* assume max. frequency is 50% higher than nominal */ ep->freqmax = ep->freqn + (ep->freqn >> 1); -- cgit v1.2.3 From 995c6a7fd9b9212abdf01160f6ce3193176be503 Mon Sep 17 00:00:00 2001 From: Jussi Laako Date: Mon, 28 Nov 2016 11:27:45 +0200 Subject: ALSA: hiface: Fix M2Tech hiFace driver sampling rate change Sampling rate changes after first set one are not reflected to the hardware, while driver and ALSA think the rate has been changed. Fix the problem by properly stopping the interface at the beginning of prepare call, allowing new rate to be set to the hardware. This keeps the hardware in sync with the driver. Signed-off-by: Jussi Laako Cc: Signed-off-by: Takashi Iwai --- sound/usb/hiface/pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/usb') diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 2c44139b4041..33db205dd12b 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -445,6 +445,8 @@ static int hiface_pcm_prepare(struct snd_pcm_substream *alsa_sub) mutex_lock(&rt->stream_mutex); + hiface_pcm_stream_stop(rt); + sub->dma_off = 0; sub->period_off = 0; -- cgit v1.2.3 From f8114f8583bb18a467c04ddc1e8978330e445801 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 21 Dec 2016 11:28:28 +0100 Subject: Revert "ALSA: usb-audio: Fix race at stopping the stream" This reverts commit 16200948d8353fe29a473a394d7d26790deae0e7. The commit was intended to cover the race condition, but it introduced yet another regression for devices with the implicit feedback, leading to a kernel panic due to NULL-dereference in an irq context. As the race condition that was addressed by the commit is very rare and the regression is much worse, let's revert the commit for rc1, and fix the issue properly in a later patch. Fixes: 16200948d835 ("ALSA: usb-audio: Fix race at stopping the stream") Reported-by: Ioan-Adrian Ratiu Cc: Signed-off-by: Takashi Iwai Signed-off-by: Linus Torvalds --- sound/usb/endpoint.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound/usb') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a2cdf3370afe..15d1d5c63c3c 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -384,9 +384,6 @@ static void snd_complete_urb(struct urb *urb) if (unlikely(atomic_read(&ep->chip->shutdown))) goto exit_clear; - if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) - goto exit_clear; - if (usb_pipeout(ep->pipe)) { retire_outbound_urb(ep, ctx); /* can be stopped during retire callback */ @@ -537,11 +534,6 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep) alive, ep->ep_num); clear_bit(EP_FLAG_STOPPING, &ep->flags); - ep->data_subs = NULL; - ep->sync_slave = NULL; - ep->retire_data_urb = NULL; - ep->prepare_data_urb = NULL; - return 0; } @@ -1028,6 +1020,10 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep) if (--ep->use_count == 0) { deactivate_urbs(ep, false); + ep->data_subs = NULL; + ep->sync_slave = NULL; + ep->retire_data_urb = NULL; + ep->prepare_data_urb = NULL; set_bit(EP_FLAG_STOPPING, &ep->flags); } } -- cgit v1.2.3