From 5afc13af36d2fdaa48bc54386c6ad43590d88be5 Mon Sep 17 00:00:00 2001 From: Gustavo Maciel Dias Vieira Date: Fri, 26 Oct 2012 12:51:53 -0200 Subject: ALSA: hda - Fix mute-LED setup for HP dv5 laptop MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The BIOS on HP dv5 doesn't have the DMI string to guide the setup of mute led GPIO and polarity. Associate this laptop with the hp-inv-led model. Signed-off-by: Gustavo Maciel Dias Vieira Tested-by: Vinícius Angiolucci Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 770013f..9ba8af0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1763,6 +1763,8 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "HP", STAC_HP_ZEPHYR), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3660, "HP Mini", STAC_92HD83XXX_HP_LED), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x144E, + "HP Pavilion dv5", STAC_92HD83XXX_HP_INV_LED), {} /* terminator */ }; -- cgit v1.1 From 257d36fd696d76b622539c26af652d2a8a931ce9 Mon Sep 17 00:00:00 2001 From: Tony Lindgren Date: Wed, 3 Oct 2012 09:31:02 -0700 Subject: ASoC: zoom2: Fix compile error by including correct header files Also drop the includes that are no longer needed and just cause problems for the ARM common zImage. Acked-by: Peter Ujfalusi Signed-off-by: Tim Gardner [tony@atomide.com: updated to drop unneeded headers] Signed-off-by: Tony Lindgren Signed-off-by: Mark Brown --- sound/soc/omap/zoom2.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 677b567..1ff6bb9 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -21,15 +21,14 @@ #include #include +#include #include #include #include #include -#include -#include -#include #include +#include /* Register descriptions for twl4030 codec part */ #include -- cgit v1.1 From 19118eb8dc3cd6bb1b1fdf0e4ad62070c6683158 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 26 Oct 2012 12:33:08 +0200 Subject: ASoC: omap-dmic: Correct functional clock name We should really use "fck" when asking for the functional clock and not "dmic_fck". This way we can ensure that multiple dmic modules can exist in the system. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-dmic.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 68f2cd1..5a6aeaf 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -464,9 +464,9 @@ static __devinit int asoc_dmic_probe(struct platform_device *pdev) mutex_init(&dmic->mutex); - dmic->fclk = clk_get(dmic->dev, "dmic_fck"); + dmic->fclk = clk_get(dmic->dev, "fck"); if (IS_ERR(dmic->fclk)) { - dev_err(dmic->dev, "cant get dmic_fck\n"); + dev_err(dmic->dev, "cant get fck\n"); return -ENODEV; } -- cgit v1.1 From 9b0573c07f278e9888c352aa9724035c75784ea0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Oct 2012 15:07:34 +0200 Subject: ALSA: PCM: Fix some races at disconnection Fix races at PCM disconnection: - while a PCM device is being opened or closed - while the PCM state is being changed without lock in prepare, hw_params, hw_free ops Reported-by: Matthieu CASTET Cc: Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 7 ++++++- sound/core/pcm_native.c | 16 ++++++++++++---- 2 files changed, 18 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index f299194..993b240 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1086,11 +1086,15 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) if (list_empty(&pcm->list)) goto unlock; + mutex_lock(&pcm->open_mutex); list_del_init(&pcm->list); for (cidx = 0; cidx < 2; cidx++) - for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) + for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) { + snd_pcm_stream_lock_irq(substream); if (substream->runtime) substream->runtime->status->state = SNDRV_PCM_STATE_DISCONNECTED; + snd_pcm_stream_unlock_irq(substream); + } list_for_each_entry(notify, &snd_pcm_notify_list, list) { notify->n_disconnect(pcm); } @@ -1110,6 +1114,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) pcm->streams[cidx].chmap_kctl = NULL; } } + mutex_unlock(&pcm->open_mutex); unlock: mutex_unlock(®ister_mutex); return 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 5e12e5b..8753c89 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -369,6 +369,14 @@ static int period_to_usecs(struct snd_pcm_runtime *runtime) return usecs; } +static void snd_pcm_set_state(struct snd_pcm_substream *substream, int state) +{ + snd_pcm_stream_lock_irq(substream); + if (substream->runtime->status->state != SNDRV_PCM_STATE_DISCONNECTED) + substream->runtime->status->state = state; + snd_pcm_stream_unlock_irq(substream); +} + static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -452,7 +460,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->boundary *= 2; snd_pcm_timer_resolution_change(substream); - runtime->status->state = SNDRV_PCM_STATE_SETUP; + snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); if (pm_qos_request_active(&substream->latency_pm_qos_req)) pm_qos_remove_request(&substream->latency_pm_qos_req); @@ -464,7 +472,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, /* hardware might be unusable from this time, so we force application to retry to set the correct hardware parameter settings */ - runtime->status->state = SNDRV_PCM_STATE_OPEN; + snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); if (substream->ops->hw_free != NULL) substream->ops->hw_free(substream); return err; @@ -512,7 +520,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) return -EBADFD; if (substream->ops->hw_free) result = substream->ops->hw_free(substream); - runtime->status->state = SNDRV_PCM_STATE_OPEN; + snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); pm_qos_remove_request(&substream->latency_pm_qos_req); return result; } @@ -1320,7 +1328,7 @@ static void snd_pcm_post_prepare(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; runtime->control->appl_ptr = runtime->status->hw_ptr; - runtime->status->state = SNDRV_PCM_STATE_PREPARED; + snd_pcm_set_state(substream, SNDRV_PCM_STATE_PREPARED); } static struct action_ops snd_pcm_action_prepare = { -- cgit v1.1 From 978520b75f0a1ce82b17e1e8186417250de6d545 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Oct 2012 15:12:55 +0200 Subject: ALSA: usb-audio: Fix races at disconnection Close some races at disconnection of a USB audio device by adding the chip->shutdown_mutex and chip->shutdown check at appropriate places. The spots to put bandaids are: - PCM prepare, hw_params and hw_free - where the usb device is accessed for communication or get speed, in mixer.c and others; the device speed is now cached in subs->speed instead of accessing to chip->dev The accesses in PCM open and close don't need the mutex protection because these are already handled in the core PCM disconnection code. The autosuspend/autoresume codes are still uncovered by this patch because of possible mutex deadlocks. They'll be covered by the upcoming change to rwsem. Also the mixer codes are untouched, too. These will be fixed in another patch, too. Reported-by: Matthieu CASTET Cc: Signed-off-by: Takashi Iwai --- sound/usb/card.h | 1 + sound/usb/mixer.c | 65 ++++++++++++++++++++++++++++++++++++------------------ sound/usb/pcm.c | 49 ++++++++++++++++++++++++++-------------- sound/usb/proc.c | 4 ++-- sound/usb/stream.c | 1 + 5 files changed, 79 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.h b/sound/usb/card.h index afa4f9e..814cb35 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -126,6 +126,7 @@ struct snd_usb_substream { struct snd_usb_endpoint *sync_endpoint; unsigned long flags; bool need_setup_ep; /* (re)configure EP at prepare? */ + unsigned int speed; /* USB_SPEED_XXX */ u64 formats; /* format bitmasks (all or'ed) */ unsigned int num_formats; /* number of supported audio formats (list) */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index fe56c9d..c2ef11c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -287,25 +287,32 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v unsigned char buf[2]; int val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; int timeout = 10; - int err; + int idx = 0, err; err = snd_usb_autoresume(cval->mixer->chip); if (err < 0) return -EIO; + mutex_lock(&chip->shutdown_mutex); while (timeout-- > 0) { + if (chip->shutdown) + break; + idx = snd_usb_ctrl_intf(chip) | (cval->id << 8); if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, - validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), - buf, val_len) >= val_len) { + validx, idx, buf, val_len) >= val_len) { *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); - snd_usb_autosuspend(cval->mixer->chip); - return 0; + err = 0; + goto out; } } - snd_usb_autosuspend(cval->mixer->chip); snd_printdd(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", - request, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), cval->val_type); - return -EINVAL; + request, validx, idx, cval->val_type); + err = -EINVAL; + + out: + mutex_unlock(&chip->shutdown_mutex); + snd_usb_autosuspend(cval->mixer->chip); + return err; } static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) @@ -313,7 +320,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v struct snd_usb_audio *chip = cval->mixer->chip; unsigned char buf[2 + 3*sizeof(__u16)]; /* enough space for one range */ unsigned char *val; - int ret, size; + int idx = 0, ret, size; __u8 bRequest; if (request == UAC_GET_CUR) { @@ -330,16 +337,22 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v if (ret) goto error; - ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, + mutex_lock(&chip->shutdown_mutex); + if (chip->shutdown) + ret = -ENODEV; + else { + idx = snd_usb_ctrl_intf(chip) | (cval->id << 8); + ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, - validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), - buf, size); + validx, idx, buf, size); + } + mutex_unlock(&chip->shutdown_mutex); snd_usb_autosuspend(chip); if (ret < 0) { error: snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", - request, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), cval->val_type); + request, validx, idx, cval->val_type); return ret; } @@ -417,7 +430,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, { struct snd_usb_audio *chip = cval->mixer->chip; unsigned char buf[2]; - int val_len, err, timeout = 10; + int idx = 0, val_len, err, timeout = 10; if (cval->mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; @@ -440,19 +453,27 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, err = snd_usb_autoresume(chip); if (err < 0) return -EIO; - while (timeout-- > 0) + mutex_lock(&chip->shutdown_mutex); + while (timeout-- > 0) { + if (chip->shutdown) + break; + idx = snd_usb_ctrl_intf(chip) | (cval->id << 8); if (snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, - validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), - buf, val_len) >= 0) { - snd_usb_autosuspend(chip); - return 0; + validx, idx, buf, val_len) >= 0) { + err = 0; + goto out; } - snd_usb_autosuspend(chip); + } snd_printdd(KERN_ERR "cannot set ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, data = %#x/%#x\n", - request, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), cval->val_type, buf[0], buf[1]); - return -EINVAL; + request, validx, idx, cval->val_type, buf[0], buf[1]); + err = -EINVAL; + + out: + mutex_unlock(&chip->shutdown_mutex); + snd_usb_autosuspend(chip); + return err; } static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int value) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 55e19e1..55e741c 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -71,6 +71,8 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream unsigned int hwptr_done; subs = (struct snd_usb_substream *)substream->runtime->private_data; + if (subs->stream->chip->shutdown) + return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; substream->runtime->delay = snd_usb_pcm_delay(subs, @@ -444,7 +446,6 @@ static int configure_endpoint(struct snd_usb_substream *subs) { int ret; - mutex_lock(&subs->stream->chip->shutdown_mutex); /* format changed */ stop_endpoints(subs, 0, 0, 0); ret = snd_usb_endpoint_set_params(subs->data_endpoint, @@ -455,7 +456,7 @@ static int configure_endpoint(struct snd_usb_substream *subs) subs->cur_audiofmt, subs->sync_endpoint); if (ret < 0) - goto unlock; + return ret; if (subs->sync_endpoint) ret = snd_usb_endpoint_set_params(subs->data_endpoint, @@ -465,9 +466,6 @@ static int configure_endpoint(struct snd_usb_substream *subs) subs->cur_rate, subs->cur_audiofmt, NULL); - -unlock: - mutex_unlock(&subs->stream->chip->shutdown_mutex); return ret; } @@ -505,7 +503,13 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if ((ret = set_format(subs, fmt)) < 0) + mutex_lock(&subs->stream->chip->shutdown_mutex); + if (subs->stream->chip->shutdown) + ret = -ENODEV; + else + ret = set_format(subs, fmt); + mutex_unlock(&subs->stream->chip->shutdown_mutex); + if (ret < 0) return ret; subs->interface = fmt->iface; @@ -528,8 +532,10 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->cur_rate = 0; subs->period_bytes = 0; mutex_lock(&subs->stream->chip->shutdown_mutex); - stop_endpoints(subs, 0, 1, 1); - deactivate_endpoints(subs); + if (!subs->stream->chip->shutdown) { + stop_endpoints(subs, 0, 1, 1); + deactivate_endpoints(subs); + } mutex_unlock(&subs->stream->chip->shutdown_mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); } @@ -552,12 +558,19 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) return -ENXIO; } - if (snd_BUG_ON(!subs->data_endpoint)) - return -EIO; + mutex_lock(&subs->stream->chip->shutdown_mutex); + if (subs->stream->chip->shutdown) { + ret = -ENODEV; + goto unlock; + } + if (snd_BUG_ON(!subs->data_endpoint)) { + ret = -EIO; + goto unlock; + } ret = set_format(subs, subs->cur_audiofmt); if (ret < 0) - return ret; + goto unlock; iface = usb_ifnum_to_if(subs->dev, subs->cur_audiofmt->iface); alts = &iface->altsetting[subs->cur_audiofmt->altset_idx]; @@ -567,12 +580,12 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->cur_audiofmt, subs->cur_rate); if (ret < 0) - return ret; + goto unlock; if (subs->need_setup_ep) { ret = configure_endpoint(subs); if (ret < 0) - return ret; + goto unlock; subs->need_setup_ep = false; } @@ -592,9 +605,11 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) /* for playback, submit the URBs now; otherwise, the first hwptr_done * updates for all URBs would happen at the same time when starting */ if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) - return start_endpoints(subs, 1); + ret = start_endpoints(subs, 1); - return 0; + unlock: + mutex_unlock(&subs->stream->chip->shutdown_mutex); + return ret; } static struct snd_pcm_hardware snd_usb_hardware = @@ -647,7 +662,7 @@ static int hw_check_valid_format(struct snd_usb_substream *subs, return 0; } /* check whether the period time is >= the data packet interval */ - if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) { + if (subs->speed != USB_SPEED_FULL) { ptime = 125 * (1 << fp->datainterval); if (ptime > pt->max || (ptime == pt->max && pt->openmax)) { hwc_debug(" > check: ptime %u > max %u\n", ptime, pt->max); @@ -925,7 +940,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre return err; param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) + if (subs->speed == USB_SPEED_FULL) /* full speed devices have fixed data packet interval */ ptmin = 1000; if (ptmin == 1000) diff --git a/sound/usb/proc.c b/sound/usb/proc.c index ebc1a5b..d218f76 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -108,7 +108,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s } snd_iprintf(buffer, "\n"); } - if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) + if (subs->speed != USB_SPEED_FULL) snd_iprintf(buffer, " Data packet interval: %d us\n", 125 * (1 << fp->datainterval)); // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); @@ -124,7 +124,7 @@ static void proc_dump_ep_status(struct snd_usb_substream *subs, return; snd_iprintf(buffer, " Packet Size = %d\n", ep->curpacksize); snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n", - snd_usb_get_speed(subs->dev) == USB_SPEED_FULL + subs->speed == USB_SPEED_FULL ? get_full_speed_hz(ep->freqm) : get_high_speed_hz(ep->freqm), ep->freqm >> 16, ep->freqm & 0xffff); diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 083ed81..1de0c8c 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -90,6 +90,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->direction = stream; subs->dev = as->chip->dev; subs->txfr_quirk = as->chip->txfr_quirk; + subs->speed = snd_usb_get_speed(subs->dev); snd_usb_set_pcm_ops(as->pcm, stream); -- cgit v1.1 From 34f3c89fda4fba9fe689db22253ca8db2f5e6386 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Oct 2012 12:16:02 +0200 Subject: ALSA: usb-audio: Use rwsem for disconnect protection Replace mutex with rwsem for codec->shutdown protection so that concurrent accesses are allowed. Also add the protection to snd_usb_autosuspend() and snd_usb_autoresume(), too. Reported-by: Matthieu CASTET Cc: Signed-off-by: Takashi Iwai --- sound/usb/card.c | 12 ++++++++---- sound/usb/mixer.c | 12 ++++++------ sound/usb/pcm.c | 12 ++++++------ sound/usb/usbaudio.h | 2 +- 4 files changed, 21 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 561bb74..282f0fc 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -339,7 +339,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, } mutex_init(&chip->mutex); - mutex_init(&chip->shutdown_mutex); + init_rwsem(&chip->shutdown_rwsem); chip->index = idx; chip->dev = dev; chip->card = card; @@ -560,7 +560,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, card = chip->card; mutex_lock(®ister_mutex); - mutex_lock(&chip->shutdown_mutex); + down_write(&chip->shutdown_rwsem); chip->shutdown = 1; chip->num_interfaces--; if (chip->num_interfaces <= 0) { @@ -582,11 +582,11 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, snd_usb_mixer_disconnect(p); } usb_chip[chip->index] = NULL; - mutex_unlock(&chip->shutdown_mutex); + up_write(&chip->shutdown_rwsem); mutex_unlock(®ister_mutex); snd_card_free_when_closed(card); } else { - mutex_unlock(&chip->shutdown_mutex); + up_write(&chip->shutdown_rwsem); mutex_unlock(®ister_mutex); } } @@ -618,16 +618,20 @@ int snd_usb_autoresume(struct snd_usb_audio *chip) { int err = -ENODEV; + down_read(&chip->shutdown_rwsem); if (!chip->shutdown && !chip->probing) err = usb_autopm_get_interface(chip->pm_intf); + up_read(&chip->shutdown_rwsem); return err; } void snd_usb_autosuspend(struct snd_usb_audio *chip) { + down_read(&chip->shutdown_rwsem); if (!chip->shutdown && !chip->probing) usb_autopm_put_interface(chip->pm_intf); + up_read(&chip->shutdown_rwsem); } static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c2ef11c..298070e 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -292,7 +292,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v err = snd_usb_autoresume(cval->mixer->chip); if (err < 0) return -EIO; - mutex_lock(&chip->shutdown_mutex); + down_read(&chip->shutdown_rwsem); while (timeout-- > 0) { if (chip->shutdown) break; @@ -310,7 +310,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v err = -EINVAL; out: - mutex_unlock(&chip->shutdown_mutex); + up_read(&chip->shutdown_rwsem); snd_usb_autosuspend(cval->mixer->chip); return err; } @@ -337,7 +337,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v if (ret) goto error; - mutex_lock(&chip->shutdown_mutex); + down_read(&chip->shutdown_rwsem); if (chip->shutdown) ret = -ENODEV; else { @@ -346,7 +346,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, buf, size); } - mutex_unlock(&chip->shutdown_mutex); + up_read(&chip->shutdown_rwsem); snd_usb_autosuspend(chip); if (ret < 0) { @@ -453,7 +453,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, err = snd_usb_autoresume(chip); if (err < 0) return -EIO; - mutex_lock(&chip->shutdown_mutex); + down_read(&chip->shutdown_rwsem); while (timeout-- > 0) { if (chip->shutdown) break; @@ -471,7 +471,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, err = -EINVAL; out: - mutex_unlock(&chip->shutdown_mutex); + up_read(&chip->shutdown_rwsem); snd_usb_autosuspend(chip); return err; } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 55e741c..37428f7 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -503,12 +503,12 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - mutex_lock(&subs->stream->chip->shutdown_mutex); + down_read(&subs->stream->chip->shutdown_rwsem); if (subs->stream->chip->shutdown) ret = -ENODEV; else ret = set_format(subs, fmt); - mutex_unlock(&subs->stream->chip->shutdown_mutex); + up_read(&subs->stream->chip->shutdown_rwsem); if (ret < 0) return ret; @@ -531,12 +531,12 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; - mutex_lock(&subs->stream->chip->shutdown_mutex); + down_read(&subs->stream->chip->shutdown_rwsem); if (!subs->stream->chip->shutdown) { stop_endpoints(subs, 0, 1, 1); deactivate_endpoints(subs); } - mutex_unlock(&subs->stream->chip->shutdown_mutex); + up_read(&subs->stream->chip->shutdown_rwsem); return snd_pcm_lib_free_vmalloc_buffer(substream); } @@ -558,7 +558,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) return -ENXIO; } - mutex_lock(&subs->stream->chip->shutdown_mutex); + down_read(&subs->stream->chip->shutdown_rwsem); if (subs->stream->chip->shutdown) { ret = -ENODEV; goto unlock; @@ -608,7 +608,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) ret = start_endpoints(subs, 1); unlock: - mutex_unlock(&subs->stream->chip->shutdown_mutex); + up_read(&subs->stream->chip->shutdown_rwsem); return ret; } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b8233eb..ef42797 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -37,7 +37,7 @@ struct snd_usb_audio { struct usb_interface *pm_intf; u32 usb_id; struct mutex mutex; - struct mutex shutdown_mutex; + struct rw_semaphore shutdown_rwsem; unsigned int shutdown:1; unsigned int probing:1; unsigned int autosuspended:1; -- cgit v1.1 From 888ea7d5ac6815ba16b3b3a20f665a92c7af6724 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Oct 2012 12:40:37 +0200 Subject: ALSA: usb-audio: Fix races at disconnection in mixer_quirks.c Similar like the previous commit, cover with chip->shutdown_rwsem and chip->shutdown checks. Reported-by: Matthieu CASTET Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 58 ++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 51 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 690000d..ae2b714 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -283,6 +283,11 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e if (value > 1) return -EINVAL; changed = value != mixer->audigy2nx_leds[index]; + down_read(&mixer->chip->shutdown_rwsem); + if (mixer->chip->shutdown) { + err = -ENODEV; + goto out; + } if (mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, @@ -299,6 +304,8 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, value, index + 2, NULL, 0); + out: + up_read(&mixer->chip->shutdown_rwsem); if (err < 0) return err; mixer->audigy2nx_leds[index] = value; @@ -392,11 +399,16 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, for (i = 0; jacks[i].name; ++i) { snd_iprintf(buffer, "%s: ", jacks[i].name); - err = snd_usb_ctl_msg(mixer->chip->dev, + down_read(&mixer->chip->shutdown_rwsem); + if (mixer->chip->shutdown) + err = 0; + else + err = snd_usb_ctl_msg(mixer->chip->dev, usb_rcvctrlpipe(mixer->chip->dev, 0), UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE, 0, jacks[i].unitid << 8, buf, 3); + up_read(&mixer->chip->shutdown_rwsem); if (err == 3 && (buf[0] == 3 || buf[0] == 6)) snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]); else @@ -426,10 +438,15 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, else new_status = old_status & ~0x02; changed = new_status != old_status; - err = snd_usb_ctl_msg(mixer->chip->dev, + down_read(&mixer->chip->shutdown_rwsem); + if (mixer->chip->shutdown) + err = -ENODEV; + else + err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x08, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, 50, 0, &new_status, 1); + up_read(&mixer->chip->shutdown_rwsem); if (err < 0) return err; mixer->xonar_u1_status = new_status; @@ -468,11 +485,17 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol, u8 bRequest = (kcontrol->private_value >> 16) & 0xff; u16 wIndex = kcontrol->private_value & 0xffff; u8 tmp; + int ret; - int ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, + down_read(&mixer->chip->shutdown_rwsem); + if (mixer->chip->shutdown) + ret = -ENODEV; + else + ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0, cpu_to_le16(wIndex), &tmp, sizeof(tmp), 1000); + up_read(&mixer->chip->shutdown_rwsem); if (ret < 0) { snd_printk(KERN_ERR @@ -493,11 +516,17 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol, u8 bRequest = (kcontrol->private_value >> 16) & 0xff; u16 wIndex = kcontrol->private_value & 0xffff; u16 wValue = ucontrol->value.integer.value[0]; + int ret; - int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest, + down_read(&mixer->chip->shutdown_rwsem); + if (mixer->chip->shutdown) + ret = -ENODEV; + else + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, cpu_to_le16(wValue), cpu_to_le16(wIndex), NULL, 0, 1000); + up_read(&mixer->chip->shutdown_rwsem); if (ret < 0) { snd_printk(KERN_ERR @@ -656,11 +685,16 @@ static int snd_ftu_eff_switch_get(struct snd_kcontrol *kctl, return -EINVAL; - err = snd_usb_ctl_msg(chip->dev, + down_read(&mixer->chip->shutdown_rwsem); + if (mixer->chip->shutdown) + err = -ENODEV; + else + err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), value, val_len); + up_read(&mixer->chip->shutdown_rwsem); if (err < 0) return err; @@ -703,11 +737,16 @@ static int snd_ftu_eff_switch_put(struct snd_kcontrol *kctl, if (!pval->is_cached) { /* Read current value */ - err = snd_usb_ctl_msg(chip->dev, + down_read(&mixer->chip->shutdown_rwsem); + if (mixer->chip->shutdown) + err = -ENODEV; + else + err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), value, val_len); + up_read(&mixer->chip->shutdown_rwsem); if (err < 0) return err; @@ -719,11 +758,16 @@ static int snd_ftu_eff_switch_put(struct snd_kcontrol *kctl, if (cur_val != new_val) { value[0] = new_val; value[1] = 0; - err = snd_usb_ctl_msg(chip->dev, + down_read(&mixer->chip->shutdown_rwsem); + if (mixer->chip->shutdown) + err = -ENODEV; + else + err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), value, val_len); + up_read(&mixer->chip->shutdown_rwsem); if (err < 0) return err; -- cgit v1.1 From a0830dbd4e42b38aefdf3fb61ba5019a1a99ea85 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Oct 2012 13:05:59 +0200 Subject: ALSA: Add a reference counter to card instance For more strict protection for wild disconnections, a refcount is introduced to the card instance, and let it up/down when an object is referred via snd_lookup_*() in the open ops. The free-after-last-close check is also changed to check this refcount instead of the empty list, too. Reported-by: Matthieu CASTET Cc: Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 9 ++++++-- sound/core/control.c | 3 +++ sound/core/hwdep.c | 5 ++++- sound/core/init.c | 50 ++++++++++++++++++++++++++----------------- sound/core/oss/mixer_oss.c | 10 +++++++-- sound/core/oss/pcm_oss.c | 2 ++ sound/core/pcm_native.c | 9 ++++++-- sound/core/rawmidi.c | 6 +++++- sound/core/sound.c | 11 ++++++++-- sound/core/sound_oss.c | 10 +++++++-- 10 files changed, 83 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c40ae57..ad11dc9 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -100,12 +100,15 @@ static int snd_compr_open(struct inode *inode, struct file *f) if (dirn != compr->direction) { pr_err("this device doesn't support this direction\n"); + snd_card_unref(compr->card); return -EINVAL; } data = kzalloc(sizeof(*data), GFP_KERNEL); - if (!data) + if (!data) { + snd_card_unref(compr->card); return -ENOMEM; + } data->stream.ops = compr->ops; data->stream.direction = dirn; data->stream.private_data = compr->private_data; @@ -113,6 +116,7 @@ static int snd_compr_open(struct inode *inode, struct file *f) runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); if (!runtime) { kfree(data); + snd_card_unref(compr->card); return -ENOMEM; } runtime->state = SNDRV_PCM_STATE_OPEN; @@ -126,7 +130,8 @@ static int snd_compr_open(struct inode *inode, struct file *f) kfree(runtime); kfree(data); } - return ret; + snd_card_unref(compr->card); + return 0; } static int snd_compr_free(struct inode *inode, struct file *f) diff --git a/sound/core/control.c b/sound/core/control.c index 7e86a5b..9768a39 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -86,6 +86,7 @@ static int snd_ctl_open(struct inode *inode, struct file *file) write_lock_irqsave(&card->ctl_files_rwlock, flags); list_add_tail(&ctl->list, &card->ctl_files); write_unlock_irqrestore(&card->ctl_files_rwlock, flags); + snd_card_unref(card); return 0; __error: @@ -93,6 +94,8 @@ static int snd_ctl_open(struct inode *inode, struct file *file) __error2: snd_card_file_remove(card, file); __error1: + if (card) + snd_card_unref(card); return err; } diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 75ea16f..53a6ba5 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -100,8 +100,10 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) if (hw == NULL) return -ENODEV; - if (!try_module_get(hw->card->module)) + if (!try_module_get(hw->card->module)) { + snd_card_unref(hw->card); return -EFAULT; + } init_waitqueue_entry(&wait, current); add_wait_queue(&hw->open_wait, &wait); @@ -148,6 +150,7 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) mutex_unlock(&hw->open_mutex); if (err < 0) module_put(hw->card->module); + snd_card_unref(hw->card); return err; } diff --git a/sound/core/init.c b/sound/core/init.c index d8ec849..7b012d1 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -213,6 +213,7 @@ int snd_card_create(int idx, const char *xid, spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); init_waitqueue_head(&card->shutdown_sleep); + atomic_set(&card->refcount, 0); #ifdef CONFIG_PM mutex_init(&card->power_lock); init_waitqueue_head(&card->power_sleep); @@ -446,21 +447,36 @@ static int snd_card_do_free(struct snd_card *card) return 0; } +/** + * snd_card_unref - release the reference counter + * @card: the card instance + * + * Decrements the reference counter. When it reaches to zero, wake up + * the sleeper and call the destructor if needed. + */ +void snd_card_unref(struct snd_card *card) +{ + if (atomic_dec_and_test(&card->refcount)) { + wake_up(&card->shutdown_sleep); + if (card->free_on_last_close) + snd_card_do_free(card); + } +} +EXPORT_SYMBOL(snd_card_unref); + int snd_card_free_when_closed(struct snd_card *card) { - int free_now = 0; - int ret = snd_card_disconnect(card); - if (ret) - return ret; + int ret; - spin_lock(&card->files_lock); - if (list_empty(&card->files_list)) - free_now = 1; - else - card->free_on_last_close = 1; - spin_unlock(&card->files_lock); + atomic_inc(&card->refcount); + ret = snd_card_disconnect(card); + if (ret) { + atomic_dec(&card->refcount); + return ret; + } - if (free_now) + card->free_on_last_close = 1; + if (atomic_dec_and_test(&card->refcount)) snd_card_do_free(card); return 0; } @@ -474,7 +490,7 @@ int snd_card_free(struct snd_card *card) return ret; /* wait, until all devices are ready for the free operation */ - wait_event(card->shutdown_sleep, list_empty(&card->files_list)); + wait_event(card->shutdown_sleep, !atomic_read(&card->refcount)); snd_card_do_free(card); return 0; } @@ -886,6 +902,7 @@ int snd_card_file_add(struct snd_card *card, struct file *file) return -ENODEV; } list_add(&mfile->list, &card->files_list); + atomic_inc(&card->refcount); spin_unlock(&card->files_lock); return 0; } @@ -908,7 +925,6 @@ EXPORT_SYMBOL(snd_card_file_add); int snd_card_file_remove(struct snd_card *card, struct file *file) { struct snd_monitor_file *mfile, *found = NULL; - int last_close = 0; spin_lock(&card->files_lock); list_for_each_entry(mfile, &card->files_list, list) { @@ -923,19 +939,13 @@ int snd_card_file_remove(struct snd_card *card, struct file *file) break; } } - if (list_empty(&card->files_list)) - last_close = 1; spin_unlock(&card->files_lock); - if (last_close) { - wake_up(&card->shutdown_sleep); - if (card->free_on_last_close) - snd_card_do_free(card); - } if (!found) { snd_printk(KERN_ERR "ALSA card file remove problem (%p)\n", file); return -ENOENT; } kfree(found); + snd_card_unref(card); return 0; } diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 29f6ded..a9a2e63 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -52,14 +52,19 @@ static int snd_mixer_oss_open(struct inode *inode, struct file *file) SNDRV_OSS_DEVICE_TYPE_MIXER); if (card == NULL) return -ENODEV; - if (card->mixer_oss == NULL) + if (card->mixer_oss == NULL) { + snd_card_unref(card); return -ENODEV; + } err = snd_card_file_add(card, file); - if (err < 0) + if (err < 0) { + snd_card_unref(card); return err; + } fmixer = kzalloc(sizeof(*fmixer), GFP_KERNEL); if (fmixer == NULL) { snd_card_file_remove(card, file); + snd_card_unref(card); return -ENOMEM; } fmixer->card = card; @@ -68,6 +73,7 @@ static int snd_mixer_oss_open(struct inode *inode, struct file *file) if (!try_module_get(card->module)) { kfree(fmixer); snd_card_file_remove(card, file); + snd_card_unref(card); return -EFAULT; } return 0; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 08fde00..2529e01 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2457,6 +2457,8 @@ static int snd_pcm_oss_open(struct inode *inode, struct file *file) __error2: snd_card_file_remove(pcm->card, file); __error1: + if (pcm) + snd_card_unref(pcm->card); return err; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 8753c89..48c6a70 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1642,6 +1642,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) write_unlock_irq(&snd_pcm_link_rwlock); up_write(&snd_pcm_link_rwsem); _nolock: + snd_card_unref(substream1->pcm->card); fput_light(file, fput_needed); if (res < 0) kfree(group); @@ -2116,7 +2117,9 @@ static int snd_pcm_playback_open(struct inode *inode, struct file *file) return err; pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_PLAYBACK); - return snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_PLAYBACK); + err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_PLAYBACK); + snd_card_unref(pcm->card); + return err; } static int snd_pcm_capture_open(struct inode *inode, struct file *file) @@ -2127,7 +2130,9 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file) return err; pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_CAPTURE); - return snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_CAPTURE); + err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_CAPTURE); + snd_card_unref(pcm->card); + return err; } static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index ebf6e49..7d4f62a 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -379,8 +379,10 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) if (rmidi == NULL) return -ENODEV; - if (!try_module_get(rmidi->card->module)) + if (!try_module_get(rmidi->card->module)) { + snd_card_unref(rmidi->card); return -ENXIO; + } mutex_lock(&rmidi->open_mutex); card = rmidi->card; @@ -440,6 +442,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) #endif file->private_data = rawmidi_file; mutex_unlock(&rmidi->open_mutex); + snd_card_unref(rmidi->card); return 0; __error: @@ -447,6 +450,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) __error_card: mutex_unlock(&rmidi->open_mutex); module_put(rmidi->card->module); + snd_card_unref(rmidi->card); return err; } diff --git a/sound/core/sound.c b/sound/core/sound.c index 6439760..89780c3 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -98,6 +98,10 @@ static void snd_request_other(int minor) * * Checks that a minor device with the specified type is registered, and returns * its user data pointer. + * + * This function increments the reference counter of the card instance + * if an associated instance with the given minor number and type is found. + * The caller must call snd_card_unref() appropriately later. */ void *snd_lookup_minor_data(unsigned int minor, int type) { @@ -108,9 +112,11 @@ void *snd_lookup_minor_data(unsigned int minor, int type) return NULL; mutex_lock(&sound_mutex); mreg = snd_minors[minor]; - if (mreg && mreg->type == type) + if (mreg && mreg->type == type) { private_data = mreg->private_data; - else + if (mreg->card_ptr) + atomic_inc(&mreg->card_ptr->refcount); + } else private_data = NULL; mutex_unlock(&sound_mutex); return private_data; @@ -275,6 +281,7 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev, preg->device = dev; preg->f_ops = f_ops; preg->private_data = private_data; + preg->card_ptr = card; mutex_lock(&sound_mutex); #ifdef CONFIG_SND_DYNAMIC_MINORS minor = snd_find_free_minor(type); diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index e952833..e1d79ee 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -40,6 +40,9 @@ static struct snd_minor *snd_oss_minors[SNDRV_OSS_MINORS]; static DEFINE_MUTEX(sound_oss_mutex); +/* NOTE: This function increments the refcount of the associated card like + * snd_lookup_minor_data(); the caller must call snd_card_unref() appropriately + */ void *snd_lookup_oss_minor_data(unsigned int minor, int type) { struct snd_minor *mreg; @@ -49,9 +52,11 @@ void *snd_lookup_oss_minor_data(unsigned int minor, int type) return NULL; mutex_lock(&sound_oss_mutex); mreg = snd_oss_minors[minor]; - if (mreg && mreg->type == type) + if (mreg && mreg->type == type) { private_data = mreg->private_data; - else + if (mreg->card_ptr) + atomic_inc(&mreg->card_ptr->refcount); + } else private_data = NULL; mutex_unlock(&sound_oss_mutex); return private_data; @@ -123,6 +128,7 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev, preg->device = dev; preg->f_ops = f_ops; preg->private_data = private_data; + preg->card_ptr = card; mutex_lock(&sound_oss_mutex); snd_oss_minors[minor] = preg; minor_unit = SNDRV_MINOR_OSS_DEVICE(minor); -- cgit v1.1 From 0914f7961babbf28aaa2f19b453951fb4841c03f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Oct 2012 16:43:39 +0200 Subject: ALSA: Avoid endless sleep after disconnect When disconnect callback is called, each component should wake up sleepers and check card->shutdown flag for avoiding the endless sleep blocking the proper resource release. Cc: Signed-off-by: Takashi Iwai --- sound/core/control.c | 2 ++ sound/core/hwdep.c | 7 +++++++ sound/core/oss/pcm_oss.c | 4 ++++ sound/core/pcm.c | 6 +++++- sound/core/pcm_native.c | 8 ++++++++ sound/core/rawmidi.c | 20 ++++++++++++++++++++ 6 files changed, 46 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 9768a39..8c7c2c9 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1437,6 +1437,8 @@ static ssize_t snd_ctl_read(struct file *file, char __user *buffer, spin_unlock_irq(&ctl->read_lock); schedule(); remove_wait_queue(&ctl->change_sleep, &wait); + if (ctl->card->shutdown) + return -ENODEV; if (signal_pending(current)) return -ERESTARTSYS; spin_lock_irq(&ctl->read_lock); diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 53a6ba5..3f7f662 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -131,6 +131,10 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) mutex_unlock(&hw->open_mutex); schedule(); mutex_lock(&hw->open_mutex); + if (hw->card->shutdown) { + err = -ENODEV; + break; + } if (signal_pending(current)) { err = -ERESTARTSYS; break; @@ -462,12 +466,15 @@ static int snd_hwdep_dev_disconnect(struct snd_device *device) mutex_unlock(®ister_mutex); return -EINVAL; } + mutex_lock(&hwdep->open_mutex); + wake_up(&hwdep->open_wait); #ifdef CONFIG_SND_OSSEMUL if (hwdep->ossreg) snd_unregister_oss_device(hwdep->oss_type, hwdep->card, hwdep->device); #endif snd_unregister_device(SNDRV_DEVICE_TYPE_HWDEP, hwdep->card, hwdep->device); list_del_init(&hwdep->list); + mutex_unlock(&hwdep->open_mutex); mutex_unlock(®ister_mutex); return 0; } diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 2529e01..f337b66 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2441,6 +2441,10 @@ static int snd_pcm_oss_open(struct inode *inode, struct file *file) mutex_unlock(&pcm->open_mutex); schedule(); mutex_lock(&pcm->open_mutex); + if (pcm->card->shutdown) { + err = -ENODEV; + break; + } if (signal_pending(current)) { err = -ERESTARTSYS; break; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 993b240..030102c 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1087,12 +1087,16 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) goto unlock; mutex_lock(&pcm->open_mutex); + wake_up(&pcm->open_wait); list_del_init(&pcm->list); for (cidx = 0; cidx < 2; cidx++) for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) { snd_pcm_stream_lock_irq(substream); - if (substream->runtime) + if (substream->runtime) { substream->runtime->status->state = SNDRV_PCM_STATE_DISCONNECTED; + wake_up(&substream->runtime->sleep); + wake_up(&substream->runtime->tsleep); + } snd_pcm_stream_unlock_irq(substream); } list_for_each_entry(notify, &snd_pcm_notify_list, list) { diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 48c6a70..6e8872d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1518,6 +1518,10 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, down_read(&snd_pcm_link_rwsem); snd_pcm_stream_lock_irq(substream); remove_wait_queue(&to_check->sleep, &wait); + if (card->shutdown) { + result = -ENODEV; + break; + } if (tout == 0) { if (substream->runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) result = -ESTRPIPE; @@ -2169,6 +2173,10 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) mutex_unlock(&pcm->open_mutex); schedule(); mutex_lock(&pcm->open_mutex); + if (pcm->card->shutdown) { + err = -ENODEV; + break; + } if (signal_pending(current)) { err = -ERESTARTSYS; break; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 7d4f62a..1bb95ae 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -424,6 +424,10 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) mutex_unlock(&rmidi->open_mutex); schedule(); mutex_lock(&rmidi->open_mutex); + if (rmidi->card->shutdown) { + err = -ENODEV; + break; + } if (signal_pending(current)) { err = -ERESTARTSYS; break; @@ -995,6 +999,8 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun spin_unlock_irq(&runtime->lock); schedule(); remove_wait_queue(&runtime->sleep, &wait); + if (rfile->rmidi->card->shutdown) + return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; if (!runtime->avail) @@ -1238,6 +1244,8 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, spin_unlock_irq(&runtime->lock); timeout = schedule_timeout(30 * HZ); remove_wait_queue(&runtime->sleep, &wait); + if (rfile->rmidi->card->shutdown) + return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; if (!runtime->avail && !timeout) @@ -1613,9 +1621,20 @@ static int snd_rawmidi_dev_register(struct snd_device *device) static int snd_rawmidi_dev_disconnect(struct snd_device *device) { struct snd_rawmidi *rmidi = device->device_data; + int dir; mutex_lock(®ister_mutex); + mutex_lock(&rmidi->open_mutex); + wake_up(&rmidi->open_wait); list_del_init(&rmidi->list); + for (dir = 0; dir < 2; dir++) { + struct snd_rawmidi_substream *s; + list_for_each_entry(s, &rmidi->streams[dir].substreams, list) { + if (s->runtime) + wake_up(&s->runtime->sleep); + } + } + #ifdef CONFIG_SND_OSSEMUL if (rmidi->ossreg) { if ((int)rmidi->device == midi_map[rmidi->card->number]) { @@ -1630,6 +1649,7 @@ static int snd_rawmidi_dev_disconnect(struct snd_device *device) } #endif /* CONFIG_SND_OSSEMUL */ snd_unregister_device(SNDRV_DEVICE_TYPE_RAWMIDI, rmidi->card, rmidi->device); + mutex_unlock(&rmidi->open_mutex); mutex_unlock(®ister_mutex); return 0; } -- cgit v1.1 From 16c2e1fae8d60a9d6d16e009a76ba3472568e094 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 31 Oct 2012 07:41:42 +0100 Subject: ALSA: ice1724: Fix rate setup after resume The rate isn't restored properly after resume since it's only set up in hw_params, and not in prepare callback. For fixing it, put the corresponding call to resume callback as well. Reported-and-tested-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 3050a52..245d874 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2859,7 +2859,12 @@ static int snd_vt1724_resume(struct device *dev) ice->set_spdif_clock(ice, 0); } else { /* internal on-card clock */ - snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 1); + int rate; + if (ice->cur_rate) + rate = ice->cur_rate; + else + rate = ice->pro_rate_default; + snd_vt1724_set_pro_rate(ice, rate, 1); } update_spdif_bits(ice, ice->pm_saved_spdif_ctrl); -- cgit v1.1 From 9f4c3f1cde541d477633479a0203ef8a834ee5f9 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 31 Oct 2012 01:20:05 -0200 Subject: ASoC: mxs-saif: Add MODULE_ALIAS Add MODULE_ALIAS information. Signed-off-by: Fabio Estevam Acked-by: Dong Aisheng Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aa037b2..93380cc 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -812,3 +812,4 @@ module_platform_driver(mxs_saif_driver); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("MXS ASoC SAIF driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mxs-saif"); -- cgit v1.1 From 213a79656462176b553c6f9cdf96e14313e43bcf Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 2 Nov 2012 13:02:53 +0000 Subject: ASoC: bells: Add missing select of WM0010 Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index e7b8317..fa166bd 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -207,6 +207,7 @@ config SND_SOC_BELLS select SND_SOC_WM5102 select SND_SOC_WM5110 select SND_SOC_WM9081 + select SND_SOC_WM0010 config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" -- cgit v1.1 From 4868ce57bfe1810262231dd8fe83fbba0ab59f13 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 2 Nov 2012 13:02:54 +0000 Subject: ASoC: bells: Select WM1250-EV1 Springbank audio I/O module Ensure we select the WM1250-EV1 as the current software system configuration demands it. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fa166bd..3c7c3a5 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -208,6 +208,7 @@ config SND_SOC_BELLS select SND_SOC_WM5110 select SND_SOC_WM9081 select SND_SOC_WM0010 + select SND_SOC_WM1250_EV1 config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" -- cgit v1.1 From f55f14752ecaccf7d6a52fd13929b73fcb191f19 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 1 Nov 2012 15:57:11 -0200 Subject: ASoC: mxs-saif: Fix channel swap for 24-bit format Playing 24-bit format file leads to channel swap on mx28 and the reason is that the current driver performs one write/read to/from the SAIF_DATA register to trigger the transfer. This approach works fine for S16_LE case because SAIF_DATA is a 32-bit register and thus is capable of storing the 16-bit left and right channels, but for the S24_LE case it can only store one channel, so in order to not lose the FIFO sync an extra read/write is needed. Reported-by: Dan Winner Signed-off-by: Fabio Estevam Tested-by: Dan Winner Acked-by: Dong Aisheng Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 93380cc..c294fbb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -523,16 +523,24 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* - * write a data to saif data register to trigger - * the transfer + * write data to saif data register to trigger + * the transfer. + * For 24-bit format the 32-bit FIFO register stores + * only one channel, so we need to write twice. + * This is also safe for the other non 24-bit formats. */ __raw_writel(0, saif->base + SAIF_DATA); + __raw_writel(0, saif->base + SAIF_DATA); } else { /* - * read a data from saif data register to trigger - * the receive + * read data from saif data register to trigger + * the receive. + * For 24-bit format the 32-bit FIFO register stores + * only one channel, so we need to read twice. + * This is also safe for the other non 24-bit formats. */ __raw_readl(saif->base + SAIF_DATA); + __raw_readl(saif->base + SAIF_DATA); } master_saif->ongoing = 1; -- cgit v1.1 From ec8f53fb693dda095ad3342b927a074e7c4dddfa Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Fri, 2 Nov 2012 00:28:50 +0900 Subject: ALSA: Fix typo in drivers sound Correct spelling typo in debug messages within drivers/sound Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai --- sound/i2c/other/ak4113.c | 2 +- sound/i2c/other/ak4114.c | 2 +- sound/i2c/other/ak4117.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index ef68d71..e04e750 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -426,7 +426,7 @@ static struct snd_kcontrol_new snd_ak4113_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4113_spdif_pinfo, diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 816e7d2..5bf4fca 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -401,7 +401,7 @@ static struct snd_kcontrol_new snd_ak4114_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4114_spdif_pinfo, .get = snd_ak4114_spdif_pget, diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index b4b2a51..40e33c9 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -380,7 +380,7 @@ static struct snd_kcontrol_new snd_ak4117_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4117_spdif_pinfo, .get = snd_ak4117_spdif_pget, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index f1cd1e3..9a8d5ce 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4899,7 +4899,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, insel = "Coaxial"; break; default: - insel = "Unkown"; + insel = "Unknown"; } snd_iprintf(buffer, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 6159929..4d8db36 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -763,7 +763,7 @@ static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai, if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) { cs42l52->sysclk = freq; } else { - dev_err(codec->dev, "Invalid freq paramter\n"); + dev_err(codec->dev, "Invalid freq parameter\n"); return -EINVAL; } return 0; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3fddc7a..b2b2b37 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3722,7 +3722,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) } while (count--); if (count == 0) - dev_warn(codec->dev, "No impedence range reported for jack\n"); + dev_warn(codec->dev, "No impedance range reported for jack\n"); #ifndef CONFIG_SND_SOC_WM8994_MODULE trace_snd_soc_jack_irq(dev_name(codec->dev)); -- cgit v1.1 From f0b3da98434589a5665d70041f8e1a5600b84fe8 Mon Sep 17 00:00:00 2001 From: "Lars R. Damerow" Date: Fri, 2 Nov 2012 13:10:39 -0700 Subject: ALSA: hda - support Teradici 2200 host card audio The audio chipset used in Teradici's Tera2 host cards is the same as that in the 1200 host cards. This patch allows ALSA to recognize the Tera2 cards. Signed-off-by: Lars R. Damerow Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 72b085a..cd2dbaf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3563,6 +3563,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT }, + { PCI_DEVICE(0x6549, 0x2200), + .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT }, /* Creative X-Fi (CA0110-IBG) */ /* CTHDA chips */ { PCI_DEVICE(0x1102, 0x0010), -- cgit v1.1 From 5a83b4b5a391f07141b157ac9daa51c409e71ab5 Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Thu, 1 Nov 2012 13:42:37 +0100 Subject: ALSA: hda: Cirrus: Fix coefficient index for beep configuration Signed-off-by: Alexander Stein Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 61a7113..3b7d67a 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1107,7 +1107,7 @@ static const struct hda_verb cs_coef_init_verbs[] = { | 0x0400 /* Disable Coefficient Auto increment */ )}, /* Beep */ - {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, + {0x11, AC_VERB_SET_COEF_INDEX, IDX_BEEP_CFG}, {0x11, AC_VERB_SET_PROC_COEF, 0x0007}, /* Enable Beep thru DAC1/2/3 */ {} /* terminator */ -- cgit v1.1 From 16337e028a6dae9fbdd718c0d42161540a668ff3 Mon Sep 17 00:00:00 2001 From: Daniel J Blueman Date: Sun, 4 Nov 2012 13:19:03 +0800 Subject: ALSA: HDA: Fix digital microphone on CS420x Correctly enable the digital microphones with the right bits in the right coeffecient registers on Cirrus CS4206/7 codecs. It also prevents misconfiguring ADC1/2. This fixes the digital mic on the Macbook Pro 10,1/Retina. Based-on-patch-by: Alexander Stein Signed-off-by: Daniel J Blueman Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 3b7d67a..859a119 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -101,8 +101,8 @@ enum { #define CS420X_VENDOR_NID 0x11 #define CS_DIG_OUT1_PIN_NID 0x10 #define CS_DIG_OUT2_PIN_NID 0x15 -#define CS_DMIC1_PIN_NID 0x12 -#define CS_DMIC2_PIN_NID 0x0e +#define CS_DMIC1_PIN_NID 0x0e +#define CS_DMIC2_PIN_NID 0x12 /* coef indices */ #define IDX_SPDIF_STAT 0x0000 @@ -1079,14 +1079,18 @@ static void init_input(struct hda_codec *codec) cs_automic(codec, NULL); coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ + cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + + coef = cs_vendor_coef_get(codec, IDX_BEEP_CFG); if (is_active_pin(codec, CS_DMIC2_PIN_NID)) - coef |= 0x0500; /* DMIC2 2 chan on, GPIO1 off */ + coef |= 1 << 4; /* DMIC2 2 chan on, GPIO1 off */ if (is_active_pin(codec, CS_DMIC1_PIN_NID)) - coef |= 0x1800; /* DMIC1 2 chan on, GPIO0 off + coef |= 1 << 3; /* DMIC1 2 chan on, GPIO0 off * No effect if SPDIF_OUT2 is * selected in IDX_SPDIF_CTL. */ - cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + + cs_vendor_coef_set(codec, IDX_BEEP_CFG, coef); } else { if (spec->mic_detect) cs_automic(codec, NULL); -- cgit v1.1 From 00e17f767e3e8d42b83a12af3ed16e3129e4feb0 Mon Sep 17 00:00:00 2001 From: Daniel J Blueman Date: Sun, 4 Nov 2012 13:19:04 +0800 Subject: ALSA: HDA: Mark CS260x immutable structures const Mark structures that won't change const. Signed-off-by: Daniel J Blueman Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 859a119..d5f3a26 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1732,8 +1732,7 @@ static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, } -static struct snd_kcontrol_new cs421x_capture_source = { - +static const struct snd_kcontrol_new cs421x_capture_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -1950,7 +1949,7 @@ static int cs421x_suspend(struct hda_codec *codec) } #endif -static struct hda_codec_ops cs421x_patch_ops = { +static const struct hda_codec_ops cs421x_patch_ops = { .build_controls = cs421x_build_controls, .build_pcms = cs_build_pcms, .init = cs421x_init, -- cgit v1.1 From 5c0ee9497b33cde3e57460efe4f73313dc0b57a3 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sun, 4 Nov 2012 23:34:58 +0100 Subject: ALSA: es1968: Add ESS vendor ID to pm_whitelist Add generic ESS vendor ID to pm_whitelist. This should fix suspend on all Maestro-2 and Maestro-2E based PCI cards. Tested on Terratec DMX and SF64-PCE2. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 5d0e568..50169bc 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2655,6 +2655,8 @@ static struct ess_device_list pm_whitelist[] __devinitdata = { { TYPE_MAESTRO2E, 0x1179 }, { TYPE_MAESTRO2E, 0x14c0 }, /* HP omnibook 4150 */ { TYPE_MAESTRO2E, 0x1558 }, + { TYPE_MAESTRO2E, 0x125d }, /* a PCI card, e.g. Terratec DMX */ + { TYPE_MAESTRO2, 0x125d }, /* a PCI card, e.g. SF64-PCE2 */ }; static struct ess_device_list mpu_blacklist[] __devinitdata = { -- cgit v1.1 From ae24c3191ba2ab03ec6b4be323e730e00404b4b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Nov 2012 12:32:46 +0100 Subject: ALSA: hda - Force to reset IEC958 status bits for AD codecs Several bug reports suggest that the forcibly resetting IEC958 status bits is required for AD codecs to get the SPDIF output working properly after changing streams. Original fix credit to Javeed Shaikh. BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/359361 Reported-by: Robin Kreis Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cdd43ea..1eeba73 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -545,6 +545,7 @@ static int ad198x_build_pcms(struct hda_codec *codec) if (spec->multiout.dig_out_nid) { info++; codec->num_pcms++; + codec->spdif_status_reset = 1; info->name = "AD198x Digital"; info->pcm_type = HDA_PCM_TYPE_SPDIF; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback; -- cgit v1.1 From 55c6f4cb6ef49afbb86222c6a3ff85329199c729 Mon Sep 17 00:00:00 2001 From: Eric Millbrandt Date: Fri, 2 Nov 2012 17:05:44 -0400 Subject: ASoC: wm8978: pll incorrectly configured when codec is master When MCLK is supplied externally and BCLK and LRC are configured as outputs (codec is master), the PLL values are only calculated correctly on the first transmission. On subsequent transmissions, at differenct sample rates, the wrong PLL values are used. Test for f_opclk instead of f_pllout to determine if the PLL values are needed. Signed-off-by: Eric Millbrandt Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8978.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 5421fd9..4c0a8e4 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -782,7 +782,7 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, wm8978->mclk_idx = -1; f_sel = wm8978->f_mclk; } else { - if (!wm8978->f_pllout) { + if (!wm8978->f_opclk) { /* We only enter here, if OPCLK is not used */ int ret = wm8978_configure_pll(codec); if (ret < 0) -- cgit v1.1 From 5b3761954dac2d1393beef8210eb8cee81d16b8d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:32:47 +0100 Subject: ALSA: hda - Fix empty DAC filling in patch_via.c In via_auto_fill_adc_nids(), the parser tries to fill dac_nids[] at the point of the current line-out (i). When no valid path is found for this output, this results in dac = 0, thus it creates a hole in dac_nids[]. This confuses is_empty_dac() and trims the detected DAC in later reference. This patch fixes the bug by appending DAC properly to dac_nids[] in via_auto_fill_adc_nids(). Reported-by: Massimo Del Fedele Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 72a2f60..bf57fa6 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1809,11 +1809,11 @@ static int via_auto_fill_dac_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; const struct auto_pin_cfg *cfg = &spec->autocfg; - int i, dac_num; + int i; hda_nid_t nid; + spec->multiout.num_dacs = 0; spec->multiout.dac_nids = spec->private_dac_nids; - dac_num = 0; for (i = 0; i < cfg->line_outs; i++) { hda_nid_t dac = 0; nid = cfg->line_out_pins[i]; @@ -1824,16 +1824,13 @@ static int via_auto_fill_dac_nids(struct hda_codec *codec) if (!i && parse_output_path(codec, nid, dac, 1, &spec->out_mix_path)) dac = spec->out_mix_path.path[0]; - if (dac) { - spec->private_dac_nids[i] = dac; - dac_num++; - } + if (dac) + spec->private_dac_nids[spec->multiout.num_dacs++] = dac; } if (!spec->out_path[0].depth && spec->out_mix_path.depth) { spec->out_path[0] = spec->out_mix_path; spec->out_mix_path.depth = 0; } - spec->multiout.num_dacs = dac_num; return 0; } -- cgit v1.1 From ef4da45828603df57e5e21b8aa21a66ce309f79b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:37:48 +0100 Subject: ALSA: hda - Fix invalid connections in VT1802 codec VT1802 codec provides the invalid connection lists of NID 0x24 and 0x33 containing the routes to a non-exist widget 0x3e. This confuses the auto-parser. Fix it up in the driver by overriding these connections. Reported-by: Massimo Del Fedele Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index bf57fa6..c2eef5c 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3646,6 +3646,18 @@ static const struct snd_pci_quirk vt2002p_fixups[] = { {} }; +/* NIDs 0x24 and 0x33 on VT1802 have connections to non-existing NID 0x3e + * Replace this with mixer NID 0x1c + */ +static void fix_vt1802_connections(struct hda_codec *codec) +{ + static hda_nid_t conn_24[] = { 0x14, 0x1c }; + static hda_nid_t conn_33[] = { 0x1c }; + + snd_hda_override_conn_list(codec, 0x24, ARRAY_SIZE(conn_24), conn_24); + snd_hda_override_conn_list(codec, 0x33, ARRAY_SIZE(conn_33), conn_33); +} + /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) { @@ -3660,6 +3672,8 @@ static int patch_vt2002P(struct hda_codec *codec) spec->aa_mix_nid = 0x21; override_mic_boost(codec, 0x2b, 0, 3, 40); override_mic_boost(codec, 0x29, 0, 3, 40); + if (spec->codec_type == VT1802) + fix_vt1802_connections(codec); add_secret_dac_path(codec); snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); -- cgit v1.1 From d5266125fb439a5dfa4edd548d888fda47414ac5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:40:36 +0100 Subject: ALSA: hda - Add pin fixups for ASUS G75 To parse properly the subwoofer outputs on ASUS G75 laptop with VT1802 codec, correct the default configurations of speaker pins 0x24 and 0x33. Reported-by: Massimo Del Fedele Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c2eef5c..019e1a0 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3625,6 +3625,7 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) */ enum { VIA_FIXUP_INTMIC_BOOST, + VIA_FIXUP_ASUS_G75, }; static void via_fixup_intmic_boost(struct hda_codec *codec, @@ -3639,9 +3640,19 @@ static const struct hda_fixup via_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = via_fixup_intmic_boost, }, + [VIA_FIXUP_ASUS_G75] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* set 0x24 and 0x33 as speakers */ + { 0x24, 0x991301f0 }, + { 0x33, 0x991301f1 }, /* subwoofer */ + { } + } + }, }; static const struct snd_pci_quirk vt2002p_fixups[] = { + SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75), SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), {} }; -- cgit v1.1 From 6268f74990c7fab6727bcb2dc82b3c4d4b302317 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 6 Nov 2012 16:33:18 +0000 Subject: ASoC: bells: Correct type in sub speaker DAI name for WM5102 Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/samsung/bells.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index b0d46d6..b56b9a3 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -247,7 +247,7 @@ static struct snd_soc_dai_link bells_dai_wm5110[] = { { .name = "Sub", .stream_name = "Sub", - .cpu_dai_name = "wm5110-aif3", + .cpu_dai_name = "wm5102-aif3", .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF -- cgit v1.1 From 5c855c8e2be67f2d5a989ef1190098f924f9f820 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Wed, 7 Nov 2012 20:38:35 +0800 Subject: ASoC: cs42l52: fix the return value of cs42l52_set_fmt() Fix the return value of cs42l52_set_fmt() when clock inversion is not allowed and also remove the useless variable ret. dpatch engine is used to auto generate this patch. (https://github.com/weiyj/dpatch) [We had been assigning to ret but then ignoring the value we assgined -- broonie] Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs42l52.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 6159929..f91136c 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -773,7 +773,6 @@ static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); - int ret = 0; u8 iface = 0; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -822,7 +821,7 @@ static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_IF: break; default: - ret = -EINVAL; + return -EINVAL; } cs42l52->config.format = iface; snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format); -- cgit v1.1 From d1a3c98d50731c627909029bb653a0557946f0f5 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Wed, 7 Nov 2012 18:00:09 +0100 Subject: ALSA: hdspm - Fix sync check reporting on RME RayDAT The RayDAT reports the sync status of its inputs in consecutive bit positions, so all we do in hdspm_s1_sync_check is to iterate over idx: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); lock = (status & (0x1<private_value: HDSPM_SYNC_CHECK("WC SyncCheck", 0), HDSPM_SYNC_CHECK("AES SyncCheck", 1), HDSPM_SYNC_CHECK("SPDIF SyncCheck", 2), HDSPM_SYNC_CHECK("ADAT1 SyncCheck", 3), HDSPM_SYNC_CHECK("ADAT2 SyncCheck", 4), HDSPM_SYNC_CHECK("ADAT3 SyncCheck", 5), HDSPM_SYNC_CHECK("ADAT4 SyncCheck", 6), HDSPM_SYNC_CHECK("TCO SyncCheck", 7), HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 8), The patch corrects the indicated sync flags by passing the proper index value to hdspm_s1_sync_check(). Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 9a8d5ce..748e36c 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3979,7 +3979,8 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, case 8: /* SYNC IN */ val = hdspm_sync_in_sync_check(hdspm); break; default: - val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1); + val = hdspm_s1_sync_check(hdspm, + kcontrol->private_value-1); } break; -- cgit v1.1 From f58161ba1b05a968e5136824b5a16b714b6a5317 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Nov 2012 08:52:45 +0100 Subject: ALSA: usb-audio: Fix crash at re-preparing the PCM stream There are bug reports of a crash with USB-audio devices when PCM prepare is performed immediately after the stream is stopped via trigger callback. It turned out that the problem is that we don't wait until all URBs are killed. This patch adds a new function to synchronize the pending stop operation on an endpoint, and calls in the prepare callback for avoiding the crash above. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181 Reported-and-tested-by: Artem S. Tashkinov Cc: [v3.6] Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 13 +++++++++++++ sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 3 +++ 3 files changed, 17 insertions(+) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7f78c6d..34de6f2 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -35,6 +35,7 @@ #define EP_FLAG_ACTIVATED 0 #define EP_FLAG_RUNNING 1 +#define EP_FLAG_STOPPING 2 /* * snd_usb_endpoint is a model that abstracts everything related to an @@ -502,10 +503,20 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep) if (alive) snd_printk(KERN_ERR "timeout: still %d active urbs on EP #%x\n", alive, ep->ep_num); + clear_bit(EP_FLAG_STOPPING, &ep->flags); return 0; } +/* sync the pending stop operation; + * this function itself doesn't trigger the stop operation + */ +void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep) +{ + if (ep && test_bit(EP_FLAG_STOPPING, &ep->flags)) + wait_clear_urbs(ep); +} + /* * unlink active urbs. */ @@ -918,6 +929,8 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, if (wait) wait_clear_urbs(ep); + else + set_bit(EP_FLAG_STOPPING, &ep->flags); } } diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6376ccf..3d4c970 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -19,6 +19,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep); void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, int force, int can_sleep, int wait); +void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct list_head *head); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 37428f7..5c12a3f 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -568,6 +568,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint); + snd_usb_endpoint_sync_pending_stop(subs->data_endpoint); + ret = set_format(subs, subs->cur_audiofmt); if (ret < 0) goto unlock; -- cgit v1.1 From 1387e2d12799e554df2f60e7ae7fe01384bcb96f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 8 Nov 2012 10:23:18 +0100 Subject: ALSA: hda - Improve HP depop when system enter to S3 alc269_toggle_power_output() was only use in ALC269VB. I rename it to alc269vb_toggle_power_output(). Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 +++++++++++------------- 1 file changed, 11 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f7397ad..b25e9b2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5840,7 +5840,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } -static void alc269_toggle_power_output(struct hda_codec *codec, int power_up) +static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { int val = alc_read_coef_idx(codec, 0x04); if (power_up) @@ -5857,10 +5857,10 @@ static void alc269_shutup(struct hda_codec *codec) if (spec->codec_variant != ALC269_TYPE_ALC269VB) return; - if ((alc_get_coef0(codec) & 0x00ff) == 0x017) - alc269_toggle_power_output(codec, 0); - if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { - alc269_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && + (alc_get_coef0(codec) & 0x00ff) == 0x018) { msleep(150); } } @@ -5870,24 +5870,22 @@ static int alc269_resume(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (spec->codec_variant == ALC269_TYPE_ALC269VB || + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x018) { - alc269_toggle_power_output(codec, 0); msleep(150); } codec->patch_ops.init(codec); - if (spec->codec_variant == ALC269_TYPE_ALC269VB || + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 1); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x017) { - alc269_toggle_power_output(codec, 1); msleep(200); } - if (spec->codec_variant == ALC269_TYPE_ALC269VB || - (alc_get_coef0(codec) & 0x00ff) == 0x018) - alc269_toggle_power_output(codec, 1); - snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); hda_call_check_power_status(codec, 0x01); -- cgit v1.1 From 19a62823eae453619604636082085812c14ee391 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 8 Nov 2012 10:25:37 +0100 Subject: ALSA: hda - Add new codec ALC668 and ALC900 (default name ALC1150) Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b25e9b2..c0ce3b1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7077,6 +7077,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, + { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, @@ -7094,6 +7095,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 }, { .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 }, { .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 }, + { .id = 0x10ec0900, .name = "ALC1150", .patch = patch_alc882 }, {} /* terminator */ }; -- cgit v1.1 From 8bb4d9ce08b0a92ca174e41d92c180328f86173f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Nov 2012 14:36:18 +0100 Subject: ALSA: Fix card refcount unbalance There are uncovered cases whether the card refcount introduced by the commit a0830dbd isn't properly increased or decreased: - OSS PCM and mixer success paths - When lookup function gets NULL This patch fixes these places. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50251 Cc: Signed-off-by: Takashi Iwai --- sound/core/oss/mixer_oss.c | 1 + sound/core/oss/pcm_oss.c | 1 + sound/core/pcm_native.c | 6 ++++-- sound/core/sound.c | 2 +- sound/core/sound_oss.c | 2 +- 5 files changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index a9a2e63..e8a1d18 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -76,6 +76,7 @@ static int snd_mixer_oss_open(struct inode *inode, struct file *file) snd_card_unref(card); return -EFAULT; } + snd_card_unref(card); return 0; } diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f337b66..4c1cc51 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2454,6 +2454,7 @@ static int snd_pcm_oss_open(struct inode *inode, struct file *file) mutex_unlock(&pcm->open_mutex); if (err < 0) goto __error; + snd_card_unref(pcm->card); return err; __error: diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 6e8872d..f9ddecf 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2122,7 +2122,8 @@ static int snd_pcm_playback_open(struct inode *inode, struct file *file) pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_PLAYBACK); err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_PLAYBACK); - snd_card_unref(pcm->card); + if (pcm) + snd_card_unref(pcm->card); return err; } @@ -2135,7 +2136,8 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file) pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_CAPTURE); err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_CAPTURE); - snd_card_unref(pcm->card); + if (pcm) + snd_card_unref(pcm->card); return err; } diff --git a/sound/core/sound.c b/sound/core/sound.c index 89780c3..70ccdab 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -114,7 +114,7 @@ void *snd_lookup_minor_data(unsigned int minor, int type) mreg = snd_minors[minor]; if (mreg && mreg->type == type) { private_data = mreg->private_data; - if (mreg->card_ptr) + if (private_data && mreg->card_ptr) atomic_inc(&mreg->card_ptr->refcount); } else private_data = NULL; diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index e1d79ee..726a49a 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -54,7 +54,7 @@ void *snd_lookup_oss_minor_data(unsigned int minor, int type) mreg = snd_oss_minors[minor]; if (mreg && mreg->type == type) { private_data = mreg->private_data; - if (mreg->card_ptr) + if (private_data && mreg->card_ptr) atomic_inc(&mreg->card_ptr->refcount); } else private_data = NULL; -- cgit v1.1 From 445632ad6dda42f4d3f9df2569a852ca0d4ea608 Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Thu, 8 Nov 2012 12:03:12 -0600 Subject: ASoC: dapm: Use card_list during DAPM shutdown DAPM shutdown incorrectly uses "list" field of codec struct while iterating over probed components (codec_dev_list). "list" field refers to codecs registered in the system, "card_list" field is used for probed components. Signed-off-by: Misael Lopez Cruz Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d0a4be3..6e35bca 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3745,7 +3745,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) { struct snd_soc_codec *codec; - list_for_each_entry(codec, &card->codec_dev_list, list) { + list_for_each_entry(codec, &card->codec_dev_list, card_list) { soc_dapm_shutdown_codec(&codec->dapm); if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) snd_soc_dapm_set_bias_level(&codec->dapm, -- cgit v1.1 From d055852ee86703d48b0c571e94bd2eb33aa9b91d Mon Sep 17 00:00:00 2001 From: Mukund Navada Date: Fri, 9 Nov 2012 11:53:40 +0530 Subject: ASoC: core: Double control update err for snd_soc_put_volsw_sx snd_soc_put_volsw_sx function fails to update second control if first control is updated by snd_soc_update_bits_locked. Signed-off-by: Mukund Navada Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d119862..10d21be 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2786,8 +2786,9 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, val = (ucontrol->value.integer.value[0] + min) & mask; val = val << shift; - if (snd_soc_update_bits_locked(codec, reg, val_mask, val)) - return err; + err = snd_soc_update_bits_locked(codec, reg, val_mask, val); + if (err < 0) + return err; if (snd_soc_volsw_is_stereo(mc)) { val_mask = mask << rshift; -- cgit v1.1 From 05193639ca977cc889668718adb38db6d585045b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Nov 2012 10:07:36 +0100 Subject: ALSA: hda - Add a missing quirk entry for iMac 9,1 This is another variant of iMac 9,1 with a different codec SSID. Reported-and-tested-by: Everaldo Canuto Cc: [v3.3+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c0ce3b1..68fd492 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5407,6 +5407,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), -- cgit v1.1 From 5574f7745436d2014fcba1163f820d132e816c85 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sat, 10 Nov 2012 19:52:50 +0100 Subject: ASoC: cs4271: free allocated GPIO In case of probe deferral, the allocated GPIO line is not freed, which prevents it from being claimed and properly asserted in later attempts. Fix this by using devm_gpio_request(). Signed-off-by: Daniel Mack Reported-by: Michael Hirsch Cc: Alexander Sverdlin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f994af3..e3f0a7f 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -485,7 +485,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) gpio_nreset = cs4271plat->gpio_nreset; if (gpio_nreset >= 0) - if (gpio_request(gpio_nreset, "CS4271 Reset")) + if (devm_gpio_request(codec->dev, gpio_nreset, "CS4271 Reset")) gpio_nreset = -EINVAL; if (gpio_nreset >= 0) { /* Reset codec */ @@ -535,15 +535,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) static int cs4271_remove(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - int gpio_nreset; - gpio_nreset = cs4271->gpio_nreset; - - if (gpio_is_valid(gpio_nreset)) { + if (gpio_is_valid(cs4271->gpio_nreset)) /* Set codec to the reset state */ - gpio_set_value(gpio_nreset, 0); - gpio_free(gpio_nreset); - } + gpio_set_value(cs4271->gpio_nreset, 0); return 0; }; -- cgit v1.1 From d2153a1595ee8235ecf9f9e2d1ac18eee373cbb5 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 13 Nov 2012 10:44:54 +0300 Subject: ALSA: es1968: precedence bug in snd_es1968_tea575x_get_pins() I don't think this works as intended. '|' higher precedence than ?: so the bitwize OR "0 | (val & STR_MOST)" is a no-op. I have re-written it to be more clear. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 50169bc..7266020 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2581,9 +2581,14 @@ static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; u16 val = inw(io); - - return (val & STR_DATA) ? TEA575X_DATA : 0 | - (val & STR_MOST) ? TEA575X_MOST : 0; + u8 ret; + + ret = 0; + if (val & STR_DATA) + ret |= TEA575X_DATA; + if (val & STR_MOST) + ret |= TEA575X_MOST; + return ret; } static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool output) -- cgit v1.1 From effded75e24c7941961d473e4f4babed4c52af3c Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 14 Nov 2012 11:23:54 +0300 Subject: ALSA: fm801: precedence bug in snd_fm801_tea575x_get_pins() There is a precedence bug because | has higher precedence than ?:. This code was cut and pasted and I fixed a similar bug a few days ago. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/fm801.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index cc2e91d..c5806f8 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -767,9 +767,14 @@ static u8 snd_fm801_tea575x_get_pins(struct snd_tea575x *tea) struct fm801 *chip = tea->private_data; unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip); - - return (reg & FM801_GPIO_GP(gpio.data)) ? TEA575X_DATA : 0 | - (reg & FM801_GPIO_GP(gpio.most)) ? TEA575X_MOST : 0; + u8 ret; + + ret = 0; + if (reg & FM801_GPIO_GP(gpio.data)) + ret |= TEA575X_DATA; + if (reg & FM801_GPIO_GP(gpio.most)) + ret |= TEA575X_MOST; + return ret; } static void snd_fm801_tea575x_set_direction(struct snd_tea575x *tea, bool output) -- cgit v1.1 From 10e44239f67d0b6fb74006e61a7e883b8075247a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Nov 2012 11:22:48 +0100 Subject: ALSA: usb-audio: Fix mutex deadlock at disconnection The recent change for USB-audio disconnection race fixes introduced a mutex deadlock again. There is a circular dependency between chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a device is opened during the disconnection operation: A. snd_usb_audio_disconnect() -> card.c::register_mutex -> chip->shutdown_rwsem (write) -> snd_card_disconnect() -> pcm.c::register_mutex -> pcm->open_mutex B. snd_pcm_open() -> pcm->open_mutex -> snd_usb_pcm_open() -> chip->shutdown_rwsem (read) Since the chip->shutdown_rwsem protection in the case A is required only for turning on the chip->shutdown flag and it doesn't have to be taken for the whole operation, we can reduce its window in snd_usb_audio_disconnect(). Reported-by: Jiri Slaby Cc: Signed-off-by: Takashi Iwai --- sound/usb/card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 282f0fc..dbf7999 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -559,9 +559,11 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, return; card = chip->card; - mutex_lock(®ister_mutex); down_write(&chip->shutdown_rwsem); chip->shutdown = 1; + up_write(&chip->shutdown_rwsem); + + mutex_lock(®ister_mutex); chip->num_interfaces--; if (chip->num_interfaces <= 0) { snd_card_disconnect(card); @@ -582,11 +584,9 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, snd_usb_mixer_disconnect(p); } usb_chip[chip->index] = NULL; - up_write(&chip->shutdown_rwsem); mutex_unlock(®ister_mutex); snd_card_free_when_closed(card); } else { - up_write(&chip->shutdown_rwsem); mutex_unlock(®ister_mutex); } } -- cgit v1.1 From c3c9b370ea4fa2566dfb0c3d88d9f02be0533e7a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Nov 2012 09:26:41 +0900 Subject: ASoC: bells: Fix up git patch application failure It seems git has been getting confused by the very similar contexts for the speaker DAIs and has been applying patches to the wrong places causing all sorts of confusion. Fix this up by hand. Reported-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/samsung/bells.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index b56b9a3..a2ca156 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -212,7 +212,7 @@ static struct snd_soc_dai_link bells_dai_wm5102[] = { { .name = "Sub", .stream_name = "Sub", - .cpu_dai_name = "wm5110-aif3", + .cpu_dai_name = "wm5102-aif3", .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF @@ -247,7 +247,7 @@ static struct snd_soc_dai_link bells_dai_wm5110[] = { { .name = "Sub", .stream_name = "Sub", - .cpu_dai_name = "wm5102-aif3", + .cpu_dai_name = "wm5110-aif3", .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF -- cgit v1.1 From ae6a5d37725853325a2b3460165fbc5613ce2916 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 20 Nov 2012 12:17:51 +0000 Subject: ASoC: kirkwood-dma: fix use of virt_to_phys() This is part of a patch found in Rabeeh Khoury's git tree for the cubox. You can not use virt_to_phys() on the address returned from dma_alloc_coherent(); it may not be part of the kernel direct-mapped memory. Fix this to use the DMA address instead. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index b9f1659..afe1930 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -178,7 +178,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) } dram = mv_mbus_dram_info(); - addr = virt_to_phys(substream->dma_buffer.area); + addr = substream->dma_buffer.addr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { prdata->play_stream = substream; kirkwood_dma_conf_mbus_windows(priv->io, -- cgit v1.1 From 25ec6bbb63e7eec905d94ccb59cdd54cf22ee618 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 20 Nov 2012 12:18:11 +0000 Subject: ASoC: kirkwood-dma: don't ignore other irq causes on error Ignoring the real cause of the interrupt is not a good idea; this behaviour has been observed to bring Dove platforms to silently lockup. Instead, on error fall through to the normal interrupt processing. This is especially important on Dove platforms as errors are handled separately, and allows us to clear down the real cause of the interrupt. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index afe1930..2ba0814 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -71,7 +71,6 @@ static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) printk(KERN_WARNING "%s: got err interrupt 0x%lx\n", __func__, cause); writel(cause, priv->io + KIRKWOOD_ERR_CAUSE); - return IRQ_HANDLED; } /* we've enabled only bytes interrupts ... */ -- cgit v1.1 From 2424d458108e275ca736dabc792ee9b6733994c5 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 20 Nov 2012 12:18:32 +0000 Subject: ASoC: kirkwood-i2s: fix DCO lock detection This is part of a patch found in Rabeeh Khoury's git tree for the cubox, which is further attributed to Sebastian Hesselbrath. Rather than masking the KIRKWOOD_DCO_SPCR_STATUS register contents against the registers virtual address, let's actually use the bit definition for the locked status, as required in the documentation. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 542538d..485af80 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -95,7 +95,7 @@ static inline void kirkwood_set_dco(void __iomem *io, unsigned long rate) do { cpu_relax(); value = readl(io + KIRKWOOD_DCO_SPCR_STATUS); - value &= KIRKWOOD_DCO_SPCR_STATUS; + value &= KIRKWOOD_DCO_SPCR_STATUS_DCO_LOCK; } while (value == 0); } -- cgit v1.1 From 982b604bc56a3da874e489051fc7adb49b1eba65 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 20 Nov 2012 12:18:52 +0000 Subject: ASoC: kirkwood-i2s: fix DMA underruns Stress testing the driver with multiple start/stop events causes kirkwood-dma to report underrun errors (which used to cause the kernel to lock up solidly). This is because kirkwood-i2s is not respecting the restrictions imposed on clearing the 'pause' bit. Follow what the spec says; the busy bit must be read as being clear twice before the pause bit can be released. This solves the underruns. However, it has been noticed that the busy bit occasionally does not clear itself, hence the waiting is bounded to 5ms maximum to avoid a new reason for the kernel to lockup. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 67 ++++++++++++++++++++++----------------- 1 file changed, 38 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 485af80..826306d 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -180,67 +180,76 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); - unsigned long value; - - /* - * specs says KIRKWOOD_PLAYCTL must be read 2 times before - * changing it. So read 1 time here and 1 later. - */ - value = readl(priv->io + KIRKWOOD_PLAYCTL); + uint32_t ctl, value; + + ctl = readl(priv->io + KIRKWOOD_PLAYCTL); + if (ctl & KIRKWOOD_PLAYCTL_PAUSE) { + unsigned timeout = 5000; + /* + * The Armada510 spec says that if we enter pause mode, the + * busy bit must be read back as clear _twice_. Make sure + * we respect that otherwise we get DMA underruns. + */ + do { + value = ctl; + ctl = readl(priv->io + KIRKWOOD_PLAYCTL); + if (!((ctl | value) & KIRKWOOD_PLAYCTL_PLAY_BUSY)) + break; + udelay(1); + } while (timeout--); + + if ((ctl | value) & KIRKWOOD_PLAYCTL_PLAY_BUSY) + dev_notice(dai->dev, "timed out waiting for busy to deassert: %08x\n", + ctl); + } switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* stop audio, enable interrupts */ - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value |= KIRKWOOD_PLAYCTL_PAUSE; - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl |= KIRKWOOD_PLAYCTL_PAUSE; + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; writel(value, priv->io + KIRKWOOD_INT_MASK); /* configure audio & enable i2s playback */ - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value &= ~KIRKWOOD_PLAYCTL_BURST_MASK; - value &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE + ctl &= ~KIRKWOOD_PLAYCTL_BURST_MASK; + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | KIRKWOOD_PLAYCTL_SPDIF_EN); if (priv->burst == 32) - value |= KIRKWOOD_PLAYCTL_BURST_32; + ctl |= KIRKWOOD_PLAYCTL_BURST_32; else - value |= KIRKWOOD_PLAYCTL_BURST_128; - value |= KIRKWOOD_PLAYCTL_I2S_EN; - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl |= KIRKWOOD_PLAYCTL_BURST_128; + ctl |= KIRKWOOD_PLAYCTL_I2S_EN; + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); value &= ~KIRKWOOD_INT_CAUSE_PLAY_BYTES; writel(value, priv->io + KIRKWOOD_INT_MASK); /* disable all playbacks */ - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value &= ~(KIRKWOOD_PLAYCTL_I2S_EN | KIRKWOOD_PLAYCTL_SPDIF_EN); - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl &= ~(KIRKWOOD_PLAYCTL_I2S_EN | KIRKWOOD_PLAYCTL_SPDIF_EN); + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; default: -- cgit v1.1 From 3ccdf5bbdf5f2488e4a36692d055ba9c43ae6717 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 20 Nov 2012 12:19:13 +0000 Subject: ASoC: kirkwood-i2s: more pause-mode fixes Don't even momentarily set the pause status when starting the channel; if we do, we should check the busy bit to ensure that we comply with the spec. In any case, it isn't necessary; we will not active on a START event so there is no need to pause the DMA. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 826306d..1d5db48 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -205,10 +205,6 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - /* stop audio, enable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE; - writel(ctl, priv->io + KIRKWOOD_PLAYCTL); - value = readl(priv->io + KIRKWOOD_INT_MASK); value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; writel(value, priv->io + KIRKWOOD_INT_MASK); @@ -269,11 +265,6 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - /* stop audio, enable interrupts */ - value = readl(priv->io + KIRKWOOD_RECCTL); - value |= KIRKWOOD_RECCTL_PAUSE; - writel(value, priv->io + KIRKWOOD_RECCTL); - value = readl(priv->io + KIRKWOOD_INT_MASK); value |= KIRKWOOD_INT_CAUSE_REC_BYTES; writel(value, priv->io + KIRKWOOD_INT_MASK); -- cgit v1.1