From f9ef724d4896763479f3921afd1ee61552fc9836 Mon Sep 17 00:00:00 2001 From: Jeronimo Borque Date: Sun, 18 Aug 2019 22:35:38 -0300 Subject: ALSA: hda - Fixes inverted Conexant GPIO mic mute led "enabled" parameter historically referred to the device input or output, not to the led indicator. After the changes added with the led helper functions the mic mute led logic refers to the led and not to the mic input which caused led indicator to be negated. Fixing logic in cxt_update_gpio_led and updated cxt_fixup_gpio_mute_hook Also updated debug messages to ease further debugging if necessary. Fixes: 184e302b46c9 ("ALSA: hda/conexant - Use the mic-mute LED helper") Suggested-by: Takashi Iwai Signed-off-by: Jeronimo Borque Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 14298ef45b21..968d3caab6ac 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -611,18 +611,20 @@ static void cxt_fixup_hp_gate_mic_jack(struct hda_codec *codec, /* update LED status via GPIO */ static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask, - bool enabled) + bool led_on) { struct conexant_spec *spec = codec->spec; unsigned int oldval = spec->gpio_led; if (spec->mute_led_polarity) - enabled = !enabled; + led_on = !led_on; - if (enabled) - spec->gpio_led &= ~mask; - else + if (led_on) spec->gpio_led |= mask; + else + spec->gpio_led &= ~mask; + codec_dbg(codec, "mask:%d enabled:%d gpio_led:%d\n", + mask, led_on, spec->gpio_led); if (spec->gpio_led != oldval) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); @@ -633,8 +635,8 @@ static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; struct conexant_spec *spec = codec->spec; - - cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); + /* muted -> LED on */ + cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, !enabled); } /* turn on/off mic-mute LED via GPIO per capture hook */ @@ -656,7 +658,6 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 }, {} }; - codec_info(codec, "action: %d gpio_led: %d\n", action, spec->gpio_led); if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook; -- cgit From 1a15718b41df026cffd0e42cfdc38a1384ce19f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Aug 2019 08:58:12 +0200 Subject: ALSA: usb-audio: Add implicit fb quirk for Behringer UFX1604 Behringer UFX1604 requires the similar quirk to apply implicit fb like another Behringer model UFX1204 in order to fix the noisy playback. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204631 Cc: Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 75b96929f76c..e4bbf79de956 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -339,6 +339,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; + case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */ case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ ep = 0x81; ifnum = 1; -- cgit From 2ca371d847511f97ef991ef612a2ce805489840e Mon Sep 17 00:00:00 2001 From: Paweł Rekowski Date: Mon, 19 Aug 2019 22:40:07 +0200 Subject: ALSA: hda/ca0132 - Add new SBZ quirk MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch adds a new PCI subsys ID for the SBZ, as found and tested by me and some reddit users. Link: https://lore.kernel.org/lkml/20190819204008.14426-1-p.rekowski@gmail.com Signed-off-by: Paweł Rekowski Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0d51823d7270..6d1fb7c11f17 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1175,6 +1175,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0027, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0033, "Sound Blaster ZxR", QUIRK_SBZ), SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), -- cgit From 6de3c9e3f6b3eaf66859e1379b3f35dda781416b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Aug 2019 11:41:06 +0200 Subject: ALSA: usb-audio: Fix invalid NULL check in snd_emuusb_set_samplerate() The quirk function snd_emuusb_set_samplerate() has a NULL check for the mixer element, but this is useless in the current code. It used to be a check against mixer->id_elems[unitid] but it was changed later to the value after mixer_eleme_list_to_info() which is always non-NULL due to the container_of() usage. This patch fixes the check before the conversion. While we're at it, correct a typo in the comment in the function, too. Fixes: 8c558076c740 ("ALSA: usb-audio: Clean up mixer element list traverse") Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199fa157a411..27dcb3743690 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1155,17 +1155,17 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, { struct usb_mixer_interface *mixer; struct usb_mixer_elem_info *cval; - int unitid = 12; /* SamleRate ExtensionUnit ID */ + int unitid = 12; /* SampleRate ExtensionUnit ID */ list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); - if (cval) { + if (mixer->id_elems[unitid]) { + cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, cval->control << 8, samplerate_id); snd_usb_mixer_notify_id(mixer, unitid); + break; } - break; } } -- cgit From 1bc8d18c75fef3b478dbdfef722aae09e2a9fde7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 21 Aug 2019 20:00:02 +0200 Subject: ALSA: line6: Fix memory leak at line6_init_pcm() error path I forgot to release the allocated object at the early error path in line6_init_pcm(). For addressing it, slightly shuffle the code so that the PCM destructor (pcm->private_free) is assigned properly before all error paths. Fixes: 3450121997ce ("ALSA: line6: Fix write on zero-sized buffer") Cc: Signed-off-by: Takashi Iwai --- sound/usb/line6/pcm.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 2c03e0f6bf72..f70211e6b174 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -550,6 +550,15 @@ int line6_init_pcm(struct usb_line6 *line6, line6pcm->volume_monitor = 255; line6pcm->line6 = line6; + spin_lock_init(&line6pcm->out.lock); + spin_lock_init(&line6pcm->in.lock); + line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD; + + line6->line6pcm = line6pcm; + + pcm->private_data = line6pcm; + pcm->private_free = line6_cleanup_pcm; + line6pcm->max_packet_size_in = usb_maxpacket(line6->usbdev, usb_rcvisocpipe(line6->usbdev, ep_read), 0); @@ -562,15 +571,6 @@ int line6_init_pcm(struct usb_line6 *line6, return -EINVAL; } - spin_lock_init(&line6pcm->out.lock); - spin_lock_init(&line6pcm->in.lock); - line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD; - - line6->line6pcm = line6pcm; - - pcm->private_data = line6pcm; - pcm->private_free = line6_cleanup_pcm; - err = line6_create_audio_out_urbs(line6pcm); if (err < 0) return err; -- cgit From f9f0e9ed350e15d51ad07364b4cf910de50c472a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Aug 2019 21:43:42 +0200 Subject: ALSA: usb-audio: Check mixer unit bitmap yet more strictly The bmControls (for UAC1) or bmMixerControls (for UAC2/3) bitmap has a variable size depending on both input and output pins. Its size is to fit with input * output bits. The problem is that the input size can't be determined simply from the unit descriptor itself but it needs to parse the whole connected sources. Although the uac_mixer_unit_get_channels() tries to check some possible overflow of this bitmap, it's incomplete due to the lack of the evaluation of input pins. For covering possible overflows, this patch adds the bitmap overflow check in the loop of input pins in parse_audio_mixer_unit(). Fixes: 0bfe5e434e66 ("ALSA: usb-audio: Check mixer unit descriptors more strictly") Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 36 ++++++++++++++++++++++++++++-------- 1 file changed, 28 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b5927c3d5bc0..eceab19766db 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -739,7 +739,6 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc) { int mu_channels; - void *c; if (desc->bLength < sizeof(*desc)) return -EINVAL; @@ -762,13 +761,6 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, break; } - if (!mu_channels) - return 0; - - c = uac_mixer_unit_bmControls(desc, state->mixer->protocol); - if (c - (void *)desc + (mu_channels - 1) / 8 >= desc->bLength) - return 0; /* no bmControls -> skip */ - return mu_channels; } @@ -2009,6 +2001,31 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, * Mixer Unit */ +/* check whether the given in/out overflows bmMixerControls matrix */ +static bool mixer_bitmap_overflow(struct uac_mixer_unit_descriptor *desc, + int protocol, int num_ins, int num_outs) +{ + u8 *hdr = (u8 *)desc; + u8 *c = uac_mixer_unit_bmControls(desc, protocol); + size_t rest; /* remaining bytes after bmMixerControls */ + + switch (protocol) { + case UAC_VERSION_1: + default: + rest = 1; /* iMixer */ + break; + case UAC_VERSION_2: + rest = 2; /* bmControls + iMixer */ + break; + case UAC_VERSION_3: + rest = 6; /* bmControls + wMixerDescrStr */ + break; + } + + /* overflow? */ + return c + (num_ins * num_outs + 7) / 8 + rest > hdr + hdr[0]; +} + /* * build a mixer unit control * @@ -2137,6 +2154,9 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, if (err < 0) return err; num_ins += iterm.channels; + if (mixer_bitmap_overflow(desc, state->mixer->protocol, + num_ins, num_outs)) + break; for (; ich < num_ins; ich++) { int och, ich_has_controls = 0; -- cgit From 75545304eba6a3d282f923b96a466dc25a81e359 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 25 Aug 2019 09:21:44 +0200 Subject: ALSA: seq: Fix potential concurrent access to the deleted pool The input pool of a client might be deleted via the resize ioctl, the the access to it should be covered by the proper locks. Currently the only missing place is the call in snd_seq_ioctl_get_client_pool(), and this patch papers over it. Reported-by: syzbot+4a75454b9ca2777f35c7@syzkaller.appspotmail.com Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 3 +-- sound/core/seq/seq_fifo.c | 17 +++++++++++++++++ sound/core/seq/seq_fifo.h | 2 ++ 3 files changed, 20 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 7737b2670064..6d9592f0ae1d 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1835,8 +1835,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client, if (cptr->type == USER_CLIENT) { info->input_pool = cptr->data.user.fifo_pool_size; info->input_free = info->input_pool; - if (cptr->data.user.fifo) - info->input_free = snd_seq_unused_cells(cptr->data.user.fifo->pool); + info->input_free = snd_seq_fifo_unused_cells(cptr->data.user.fifo); } else { info->input_pool = 0; info->input_free = 0; diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index ea69261f269a..eaaa8b5830bb 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -263,3 +263,20 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize) return 0; } + +/* get the number of unused cells safely */ +int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f) +{ + unsigned long flags; + int cells; + + if (!f) + return 0; + + snd_use_lock_use(&f->use_lock); + spin_lock_irqsave(&f->lock, flags); + cells = snd_seq_unused_cells(f->pool); + spin_unlock_irqrestore(&f->lock, flags); + snd_use_lock_free(&f->use_lock); + return cells; +} diff --git a/sound/core/seq/seq_fifo.h b/sound/core/seq/seq_fifo.h index edc68743943d..b56a7b897c9c 100644 --- a/sound/core/seq/seq_fifo.h +++ b/sound/core/seq/seq_fifo.h @@ -53,5 +53,7 @@ int snd_seq_fifo_poll_wait(struct snd_seq_fifo *f, struct file *file, poll_table /* resize pool in fifo */ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize); +/* get the number of unused cells safely */ +int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f); #endif -- cgit From 2fd2329393658514db074abd4f7dea8da1c20f81 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 26 Aug 2019 22:55:15 +0900 Subject: ALSA: oxfw: fix to handle correct stream for PCM playback When userspace application calls ioctl(2) to configure hardware for PCM playback substream, ALSA OXFW driver handles incoming AMDTP stream. In this case, outgoing AMDTP stream should be handled. This commit fixes the bug for v5.3-rc kernel. Fixes: 4f380d007052 ("ALSA: oxfw: configure packet format in pcm.hw_params callback") Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 9ea39348cdf5..7c6d1c277d4d 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -248,7 +248,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(hw_params); mutex_lock(&oxfw->mutex); - err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream, + err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream, rate, channels); if (err >= 0) ++oxfw->substreams_count; -- cgit