From ab548d2dba63ba947287965e525cc02a15d9853d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Sep 2012 10:10:11 +0200 Subject: ALSA: hda - Fix missing Master volume for STAC9200/925x With the commit [2faa3bf: ALSA: hda - Rewrite the mute-LED hook with vmaster hook in patch_sigmatel.c], the former Master volume control was converted to PCM. This was supposed to be covered by the vmaster control. But due to the lack of "PCM" slave definition, this didn't happen properly. The patch fixes the missing entry. Reported-by: Andrew Shadura Cc: [v3.4+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6f806d3e56bb..3d4722f0a1ca 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1075,7 +1075,7 @@ static struct snd_kcontrol_new stac_smux_mixer = { static const char * const slave_pfxs[] = { "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Speaker", "IEC958", + "Headphone", "Speaker", "IEC958", "PCM", NULL }; -- cgit v1.2.3 From 1213a205f9ed27d97de3d5bed28fb085ef4853e2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Sep 2012 14:58:00 +0200 Subject: ALSA: usb-audio: Fix bogus error messages for delay accounting The recent fix for the missing fine delayed time adjustment gives strange error messages at each start of the playback stream, such as delay: estimated 0, actual 352 delay: estimated 353, actual 705 These come from the sanity check in retire_playback_urb(). Before the stream is activated via start_endpoints(), a few silent packets have been already sent. And at this point the delay account is still in the state as if the new packets are just queued, so the driver gets confused and spews the bogus error messages. For fixing the issue, we just need to check whether the received packet is valid, whether it's zero sized or not. Reported-by: Markus Trippelsdorf Cc: [v3.5+] Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index fd5e982fc98c..f782ce19bf5a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1140,6 +1140,12 @@ static void retire_playback_urb(struct snd_usb_substream *subs, int processed = urb->transfer_buffer_length / stride; int est_delay; + /* ignore the delay accounting when procssed=0 is given, i.e. + * silent payloads are procssed before handling the actual data + */ + if (!processed) + return; + spin_lock_irqsave(&subs->lock, flags); est_delay = snd_usb_pcm_delay(subs, runtime->rate); /* update delay with exact number of samples played */ -- cgit v1.2.3 From 07dc59f0988cb54fd87bd373b3b27eb2401dd811 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 10 Sep 2012 09:39:31 +0200 Subject: ALSA: hda - Fix Oops at codec reset/reconfig snd_hda_codec_reset() calls restore_pincfgs() where the codec is powered up again, which eventually tries to resume and initialize via the callbacks of the codec. However, it's the place just after codec free callback, thus no codec callbacks should be called after that. On a codec like CS4206, it results in Oops due to the access in init callback. This patch fixes the issue by clearing the codec callbacks properly after freeing codec. Reported-by: Daniel J Blueman Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f25c24c743f9..1c65cc5e3a31 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2353,6 +2353,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) } if (codec->patch_ops.free) codec->patch_ops.free(codec); + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); snd_hda_jack_tbl_clear(codec); codec->proc_widget_hook = NULL; codec->spec = NULL; @@ -2368,7 +2369,6 @@ int snd_hda_codec_reset(struct hda_codec *codec) codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; - memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); codec->slave_dig_outs = NULL; codec->spdif_status_reset = 0; module_put(codec->owner); -- cgit v1.2.3 From 81cb324675eec592ab8f3038f980c074fbf7fb9b Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 11 Sep 2012 14:12:43 +0300 Subject: ALSA: compress_core: fix open flags test in snd_compr_open() O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always false and it will never do compress capture. The test for O_WRONLY is also slightly off. The original test would consider "->flags = (O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid. I've also removed the pr_err() because that could flood dmesg. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ec2118d0e27a..eb60cb8dbb8a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -80,14 +80,12 @@ static int snd_compr_open(struct inode *inode, struct file *f) int maj = imajor(inode); int ret; - if (f->f_flags & O_WRONLY) + if ((f->f_flags & O_ACCMODE) == O_WRONLY) dirn = SND_COMPRESS_PLAYBACK; - else if (f->f_flags & O_RDONLY) + else if ((f->f_flags & O_ACCMODE) == O_RDONLY) dirn = SND_COMPRESS_CAPTURE; - else { - pr_err("invalid direction\n"); + else return -EINVAL; - } if (maj == snd_major) compr = snd_lookup_minor_data(iminor(inode), -- cgit v1.2.3 From c302d6133c094bda7a7ce94eac5b50c018a7ca7b Mon Sep 17 00:00:00 2001 From: Catalin Iacob Date: Sun, 9 Sep 2012 21:41:11 +0000 Subject: ALSA: hda_intel: add position_fix quirk for Asus K53E Commit c20c5a841cbe47f5b7812b57bd25397497e5fbc0 changed some chipsets to default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including repeated sounds on my Asus laptop. My hardware is Cougar Point which the commit log of c20c5a841cbe47f5b7812b57bd25397497e5fbc0 mentions as tested so POS_FIX_COMBO probably works in general but apparently it doesn't on Asus K53E therefore the need for the quirk. Signed-off-by: Catalin Iacob Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 60882c62f180..228cdf93fa29 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2701,6 +2701,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), -- cgit v1.2.3 From 3737e2be505d872bf2b3c1cd4151b2d2b413d7b5 Mon Sep 17 00:00:00 2001 From: Matteo Frigo Date: Wed, 12 Sep 2012 10:12:06 -0400 Subject: ALSA: ice1724: Use linear scale for AK4396 volume control. The AK4396 DAC has a linear-scale attentuator, but sound/pci/ice1712/prodigy_hifi.c used a log scale instead, which is not quite right. This patch restores the correct scale, borrowing from the ak4396 code in sound/pci/oxygen/oxygen.c. Signed-off-by: Matteo Frigo Cc: Signed-off-by: Takashi Iwai --- sound/pci/ice1712/prodigy_hifi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 764cc93dbca4..075d5aa1fee0 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -297,6 +297,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem } static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); +static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { { @@ -307,7 +308,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { .info = ak4396_dac_vol_info, .get = ak4396_dac_vol_get, .put = ak4396_dac_vol_put, - .tlv = { .p = db_scale_wm_dac }, + .tlv = { .p = ak4396_db_scale }, }, }; -- cgit v1.2.3