February 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: omap-dmic: Clean up with devm_request_and_ioremap
by Sebastien Guiriec 13 Feb '13

13 Feb '13

Clean up dmic code with devm_request_and_ioremap function. Signed-off-by: Sebastien Guiriec <s-guiriec(a)ti.com> --- sound/soc/omap/omap-dmic.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index ba49ccd..77e9e7e 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -493,16 +493,9 @@ static int asoc_dmic_probe(struct platform_device *pdev) goto err_put_clk; } - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_err(dmic->dev, "memory region already claimed\n"); - ret = -ENODEV; - goto err_put_clk; - } - - dmic->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); + dmic->io_base = devm_request_and_ioremap(&pdev->dev, res); if (!dmic->io_base) { + dev_err(&pdev->dev, "cannot remap\n"); ret = -ENOMEM; goto err_put_clk; } -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: omap-mcpdm: Clean up with devm_* function
by Sebastien Guiriec 13 Feb '13

13 Feb '13

Clean up McPDM driver with devm_ function. Signed-off-by: Sebastien Guiriec <s-guiriec(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 5ca11bd..079f277 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -369,7 +369,7 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(mcpdm->dev); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); - ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler, 0, "McPDM", (void *)mcpdm); pm_runtime_put_sync(mcpdm->dev); @@ -389,7 +389,6 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); return 0; @@ -465,14 +464,11 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) if (res == NULL) return -ENOMEM; - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), "McPDM")) - return -EBUSY; - - mcpdm->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!mcpdm->io_base) + mcpdm->io_base = devm_request_and_ioremap(&pdev->dev, res); + if (!mcpdm->io_base) { + dev_err(&pdev->dev, "cannot remap\n"); return -ENOMEM; + } mcpdm->irq = platform_get_irq(pdev, 0); if (mcpdm->irq < 0) -- 1.7.10.4

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Re: [alsa-devel] [RFC]Connection list of hda widget
by Takashi Iwai 13 Feb '13

13 Feb '13

At Thu, 07 Feb 2013 16:09:35 +0100, David Henningsson wrote: > > On 02/07/2013 06:38 AM, Raymond Yau wrote: > > > > Treat vendor defined widget different from audio selector/audio input > > widget since hda_proc.c query the connection list of vendor defined widget > > > > ADI codecs have vendor defined widgets ( BIAS Power-Down and VREF > > Power-Down) which have connection list > > > > static void print_conn_list(struct snd_info_buffer *buffer, > > struct hda_codec *codec, hda_nid_t nid, > > unsigned int wid_type, hda_nid_t *conn, > > int conn_len) > > { > > int c, curr = -1; > > > > if (conn_len > 1 && > > wid_type != AC_WID_AUD_MIX && > > wid_type != AC_WID_VOL_KNB && > > + wide_type != AC_WID_VENDOR && > > wid_type != AC_WID_POWER) > > curr = snd_hda_codec_read(codec, nid, 0, > > AC_VERB_GET_CONNECT_SEL, 0); > > snd_iprintf(buffer, " Connection: %d\n", conn_len); > > > > I think there are some Sigmatel/IDT (or was it Analog Devices?) codecs > that actually have sensible connections for their vendor specific nodes, > so I would prefer not to change this. Right. The widget connection longer than 32 itself is OK. It's just ugly. That being said, it's basically a bug that we restrict the max connection size in the core code. So I applied the patch below. This removes the limit while we still keep the efficiency (except for proc output, which isn't too critical). The patch became bigger than I wanted originally, but it's life. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Remove limit of widget connections Currently we set the max number of connections to be 32, but there seems codec that gives longer connection lists like AD1988, and we see errors in proc output and else. (Though, in the case of AD1988, it's a list of all codecs connected to a single vendor widget, so this must be something fishy, but it's still valid from the h/w design POV.) This patch tries to remove this restriction. For efficiency, we still use the fixed size array in the parser, but takes a dynamic array when the size is reported to be greater than that. Now the fixed array size is found only in patch_hdmi.c, but it should be fine, as the codec itself can't support so many pins. Reported-by: Raymond Yau <superquad.vortex2(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 62 ++++++++++++++++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 4 +-- sound/pci/hda/hda_proc.c | 19 +++++++++++--- sound/pci/hda/patch_hdmi.c | 3 +++ 4 files changed, 60 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f82a64d..3c92514 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -382,13 +382,23 @@ static void remove_conn_list(struct hda_codec *codec) /* read the connection and add to the cache */ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) { - hda_nid_t list[HDA_MAX_CONNECTIONS]; + hda_nid_t list[32]; + hda_nid_t *result = list; int len; len = snd_hda_get_raw_connections(codec, nid, list, ARRAY_SIZE(list)); - if (len < 0) - return len; - return snd_hda_override_conn_list(codec, nid, len, list); + if (len == -ENOSPC) { + len = snd_hda_get_num_raw_conns(codec, nid); + result = kmalloc(sizeof(hda_nid_t) * len, GFP_KERNEL); + if (!result) + return -ENOMEM; + len = snd_hda_get_raw_connections(codec, nid, result, len); + } + if (len >= 0) + len = snd_hda_override_conn_list(codec, nid, len, result); + if (result != list) + kfree(result); + return len; } /** @@ -466,6 +476,27 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_get_connections); +/* return CONNLIST_LEN parameter of the given widget */ +static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!(wcaps & AC_WCAP_CONN_LIST) && + get_wcaps_type(wcaps) != AC_WID_VOL_KNB) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); + if (parm == -1) + parm = 0; + return parm; +} + +int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid) +{ + return get_num_conns(codec, nid) & AC_CLIST_LENGTH; +} + /** * snd_hda_get_raw_connections - copy connection list without cache * @codec: the HDA codec @@ -483,19 +514,16 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, unsigned int parm; int i, conn_len, conns; unsigned int shift, num_elems, mask; - unsigned int wcaps; hda_nid_t prev_nid; int null_count = 0; if (snd_BUG_ON(!conn_list || max_conns <= 0)) return -EINVAL; - wcaps = get_wcaps(codec, nid); - if (!(wcaps & AC_WCAP_CONN_LIST) && - get_wcaps_type(wcaps) != AC_WID_VOL_KNB) + parm = get_num_conns(codec, nid); + if (!parm) return 0; - parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); if (parm & AC_CLIST_LONG) { /* long form */ shift = 16; @@ -552,21 +580,13 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, continue; } for (n = prev_nid + 1; n <= val; n++) { - if (conns >= max_conns) { - snd_printk(KERN_ERR "hda_codec: " - "Too many connections %d for NID 0x%x\n", - conns, nid); - return -EINVAL; - } + if (conns >= max_conns) + return -ENOSPC; conn_list[conns++] = n; } } else { - if (conns >= max_conns) { - snd_printk(KERN_ERR "hda_codec: " - "Too many connections %d for NID 0x%x\n", - conns, nid); - return -EINVAL; - } + if (conns >= max_conns) + return -ENOSPC; conn_list[conns++] = val; } prev_nid = val; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6c59272..0be1826 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -551,9 +551,6 @@ enum { AC_JACK_PORT_BOTH, }; -/* max. connections to a widget */ -#define HDA_MAX_CONNECTIONS 32 - /* max. codec address */ #define HDA_MAX_CODEC_ADDRESS 0x0f @@ -958,6 +955,7 @@ snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) { return snd_hda_get_connections(codec, nid, NULL, 0); } +int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid); int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 5e02f26..97cdbef 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -623,7 +623,7 @@ static void print_codec_info(struct snd_info_entry *entry, snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); unsigned int wid_type = get_wcaps_type(wid_caps); - hda_nid_t conn[HDA_MAX_CONNECTIONS]; + hda_nid_t *conn = NULL; int conn_len = 0; snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, @@ -660,9 +660,18 @@ static void print_codec_info(struct snd_info_entry *entry, if (wid_type == AC_WID_VOL_KNB) wid_caps |= AC_WCAP_CONN_LIST; - if (wid_caps & AC_WCAP_CONN_LIST) - conn_len = snd_hda_get_raw_connections(codec, nid, conn, - HDA_MAX_CONNECTIONS); + if (wid_caps & AC_WCAP_CONN_LIST) { + conn_len = snd_hda_get_num_raw_conns(codec, nid); + if (conn_len > 0) { + conn = kmalloc(sizeof(hda_nid_t) * conn_len, + GFP_KERNEL); + if (!conn) + return; + if (snd_hda_get_raw_connections(codec, nid, conn, + conn_len) < 0) + conn_len = 0; + } + } if (wid_caps & AC_WCAP_IN_AMP) { snd_iprintf(buffer, " Amp-In caps: "); @@ -735,6 +744,8 @@ static void print_codec_info(struct snd_info_entry *entry, if (codec->proc_widget_hook) codec->proc_widget_hook(buffer, codec, nid); + + kfree(conn); } snd_hda_power_down(codec); } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 85236da..899c4fb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -64,6 +64,9 @@ struct hdmi_spec_per_cvt { unsigned int maxbps; }; +/* max. connections to a widget */ +#define HDA_MAX_CONNECTIONS 32 + struct hdmi_spec_per_pin { hda_nid_t pin_nid; int num_mux_nids; -- 1.8.1.2

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[alsa-devel] Channel map for Logitech Web Camera
by Raymond Yau 13 Feb '13

13 Feb '13

It seem that channel map is still missing for Logitech Web Camera since I can find the corresponding entries of hda-intel and the patched au8830 in asound.state except the Logitech Web Camera state.U0x46d0x992 { control.1 { iface MIXER name 'Mic Capture Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.2 { iface MIXER name 'Mic Capture Volume' value 3041 comment { access 'read write' type INTEGER count 1 range '0 - 3072' dbmin 1800 dbmax 3000 dbvalue.0 2987 } } } Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 2 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 2 Streaming bFunctionProtocol 0 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 38 bInCollection 1 baInterfaceNr( 0) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 1 wChannelConfig 0x0000 iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0101 USB Streaming bAssocTerminal 1 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 1 bControlSize 1 bmaControls( 0) 0x03 Mute Control Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 16000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0044 1x 68 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined

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[alsa-devel] [RFC] Fix for conflict of HDMI and SPDIF IEC958 controls
by Takashi Iwai 12 Feb '13

12 Feb '13

Hi, there is a long-standing problem in HD-audio regarding IEC958 controls. When both an SPDIF and an HDMI are created on the same card (e.g. one from analog codec and one from graphics chip), the driver assigns the IEC958 controls just with new indices, 0, 1, 2... The problem is that there is no way to connect between this index and the actual PCM device. Currently, alsa-lib HDA-Intel.conf has a fixed configuration: spdif -> IEC958 xxx index=0, PCM dev=1 hdmi,0 -> IEC958 xxx index=0, PCM dev=3 hdmi,1 -> IEC958 xxx index=1, PCM dev=7 hdmi,2 -> IEC958 xxx index=2, PCM dev=8 hdmi,3 -> IEC958 xxx index=3, PCM dev=9 So obviously spdif and the first hdmi conflict. Basically this can be fixed by reassigning each IEC958 control with the same device number corresponding to the PCM device. That is, for SPDIF, assign a control element with device=1, for HDMI, device=3,7... However, this obviously breaks the old configuration unless user upgrades the alsa-lib configuration. Thus this is no-go. Now here is a compromise: the IEC958 control for SPDIF is reassigned to device=1 *only* when SPDIF and HDMI coexist. Since the configuration is anyway broken as is now in such a case, it's no big deal to fix one side in an incompatible way. (The reason why SPDIF is re-assigned is that I guess majority of user require more HDMI than SPDIF in such a configuration.) In addition, we need a fix in alsa-lib. Also not for breaking the compatibility with older kernel, we need some fallback check. I did a quick hack to alsa-lib conf code and added "skip_rest" option to the hook element, so that only the first matching element is taken. Long story short, I cooked up three patches. One patch is for kernel, to add the workaround above, and the two are for alsa-lib, one clean up and one to introduce the skip_rest option (and application to HDA-Intel.conf). My plan is to merge this to 3.8 tree, so it's no urgent issue. But it's of course always good to fix something. thanks, Takashi

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Re: [alsa-devel] Xonar DG pci sound card driver incomplete
by Clemens Ladisch 12 Feb '13

12 Feb '13

kernel kernel wrote: > Requesting guidance on how to implement the missing mic input support > for this Asus Xonar card. I've downloaded the relevant datasheets but > am unsure how to proceed. Rip out the card, and look (or eletrically trace) how the CMI8788's GPIOs and the CS4245's inputs/outputs are connected. Test how to configure and set the GPIOs so that the mic works. Also check the register configuration of the CS4245. Regards, Clemens

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Re: [alsa-devel] [PATCH 2/3] ALSA: hda - no-primary-hp is a quirk for model ALC889 not ALC882
by Takashi Iwai 12 Feb '13

12 Feb '13

At Tue, 12 Feb 2013 16:54:16 +0900, Fernando Luis Vázquez Cao wrote: > > Substitute ALC889 for ALC882 in macro and function names. > > Cc: stable(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Fernando Luis Vazquez Cao <fernando(a)oss.ntt.co.jp> This fixup can work generically to all ALC882 variants, including ALC889. Unless you are really sure that this must be used exclusively only for ALC889, no reason to change the prefix. thanks, Takashi > --- > > diff -urNp linux-3.7.6-orig/sound/pci/hda/patch_realtek.c linux-3.7.6/sound/pci/hda/patch_realtek.c > --- linux-3.7.6-orig/sound/pci/hda/patch_realtek.c 2013-02-09 22:52:40.301209823 +0900 > +++ linux-3.7.6/sound/pci/hda/patch_realtek.c 2013-02-09 22:58:54.695063675 +0900 > @@ -5030,7 +5030,7 @@ enum { > ALC889_FIXUP_MBP_VREF, > ALC889_FIXUP_IMAC91_VREF, > ALC882_FIXUP_INV_DMIC, > - ALC882_FIXUP_NO_PRIMARY_HP, > + ALC889_FIXUP_NO_PRIMARY_HP, > }; > > static void alc889_fixup_coef(struct hda_codec *codec, > @@ -5156,7 +5156,7 @@ static void alc889_fixup_imac91_vref(str > * Strangely, the speaker output doesn't work on Vaio Z and some Vaio > * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 > */ > -static void alc882_fixup_no_primary_hp(struct hda_codec *codec, > +static void alc889_fixup_no_primary_hp(struct hda_codec *codec, > const struct alc_fixup *fix, int action) > { > struct alc_spec *spec = codec->spec; > @@ -5350,9 +5350,9 @@ static const struct alc_fixup alc882_fix > .type = ALC_FIXUP_FUNC, > .v.func = alc_fixup_inv_dmic_0x12, > }, > - [ALC882_FIXUP_NO_PRIMARY_HP] = { > + [ALC889_FIXUP_NO_PRIMARY_HP] = { > .type = ALC_FIXUP_FUNC, > - .v.func = alc882_fixup_no_primary_hp, > + .v.func = alc889_fixup_no_primary_hp, > }, > }; > > @@ -5388,8 +5388,8 @@ static const struct snd_pci_quirk alc882 > SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), > SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), > SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), > - SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), > - SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), > + SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC889_FIXUP_NO_PRIMARY_HP), > + SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC889_FIXUP_NO_PRIMARY_HP), > > /* All Apple entries are in codec SSIDs */ > SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), > @@ -5432,7 +5432,7 @@ static const struct alc_model_fixup alc8 > {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, > {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, > {.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"}, > - {.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, > + {.id = ALC889_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, > {} > }; > > >

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Re: [alsa-devel] [PATCH 2/3] ALSA: hda - update documentation for no-primary-hp fixup
by Takashi Iwai 12 Feb '13

12 Feb '13

At Tue, 12 Feb 2013 16:49:46 +0900, Fernando Luis Vázquez Cao wrote: > > The problem addressed by this fixup is not specific to Vaio Z, affecting > some Vaio all-in-one desktop PCs too. Update the code comments accordingly. > > Cc: stable(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Fernando Luis Vazquez Cao <fernando(a)oss.ntt.co.jp> I applied this now to sound git tree but dropped Cc to stable because the patch is no longer cleanly applicable due to the recent code change. And this is just some minor documentation update, thus we can still live peacefully without this fix. thanks, Takashi > --- > > diff -urNp linux-3.7.6-orig/Documentation/sound/alsa/HD-Audio-Models.txt linux-3.7.6/Documentation/sound/alsa/HD-Audio-Models.txt > --- linux-3.7.6-orig/Documentation/sound/alsa/HD-Audio-Models.txt 2012-12-11 12:30:57.000000000 +0900 > +++ linux-3.7.6/Documentation/sound/alsa/HD-Audio-Models.txt 2013-02-09 22:44:28.614772054 +0900 > @@ -53,7 +53,7 @@ ALC882/883/885/888/889 > acer-aspire-8930g Acer Aspire 8330G/6935G > acer-aspire Acer Aspire others > inv-dmic Inverted internal mic workaround > - no-primary-hp VAIO Z workaround (for fixed speaker DAC) > + no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) > > ALC861/660 > ========== > diff -urNp linux-3.7.6-orig/sound/pci/hda/patch_realtek.c linux-3.7.6/sound/pci/hda/patch_realtek.c > --- linux-3.7.6-orig/sound/pci/hda/patch_realtek.c 2013-02-09 22:39:11.745202168 +0900 > +++ linux-3.7.6/sound/pci/hda/patch_realtek.c 2013-02-09 22:50:10.924470402 +0900 > @@ -5153,7 +5153,8 @@ static void alc889_fixup_imac91_vref(str > } > > /* Don't take HP output as primary > - * strangely, the speaker output doesn't work on VAIO Z through DAC 0x05 > + * Strangely, the speaker output doesn't work on Vaio Z and some Vaio > + * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 > */ > static void alc882_fixup_no_primary_hp(struct hda_codec *codec, > const struct alc_fixup *fix, int action) > >

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Re: [alsa-devel] [PATCH 1/3] ALSA: hda - Workaround for silent output on Sony Vaio VGC-LN51JGB with ALC889
by Takashi Iwai 12 Feb '13

12 Feb '13

At Tue, 12 Feb 2013 16:47:44 +0900, Fernando Luis Vázquez Cao wrote: > > Some Vaio all-in-one desktop PCs (for example VGC-LN51JGB) are affected by > the same issue that caused Vaio Z laptops to become silent: the speaker pin > must be connected to the first DAC even though the codec itself advertises > flexible routing through any of the DACs. > > Use the no-primary-hp fixup for choosing the speaker pin as the primary so > that the right DAC is assigned on this device. > > Cc: stable(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Fernando Luis Vazquez Cao <fernando(a)oss.ntt.co.jp> Thanks, applied now. Takashi > --- > > diff -urNp linux-3.7.6-orig/sound/pci/hda/patch_realtek.c linux-3.7.6/sound/pci/hda/patch_realtek.c > --- linux-3.7.6-orig/sound/pci/hda/patch_realtek.c 2013-02-08 23:27:05.949484427 +0900 > +++ linux-3.7.6/sound/pci/hda/patch_realtek.c 2013-02-08 23:41:04.897644545 +0900 > @@ -5388,6 +5388,7 @@ static const struct snd_pci_quirk alc882 > SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), > SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), > SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), > + SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), > > /* All Apple entries are in codec SSIDs */ > SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), > >

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[alsa-devel] [patch] ALSA: hda/ca0132 - fix error handling for dsp_chip_to_dsp_addx()
by Dan Carpenter 12 Feb '13

12 Feb '13

The error handling here won't work on 64 bit systems. INVALID_CHIP_ADDRESS is an unsigned long defined like this: #define INVALID_CHIP_ADDRESS (~0UL) But we truncate the high bits away before returning and then we save it in an unsigned 32bit and then we compare it against the unsigned long version. I've changed everything to unsigned long. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 639a282..42114e9 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1581,7 +1581,7 @@ static int dsp_reset(struct hda_codec *codec) /* * Convert chip address to DSP address */ -static unsigned int dsp_chip_to_dsp_addx(unsigned int chip_addx, +static unsigned long dsp_chip_to_dsp_addx(unsigned int chip_addx, bool *code, bool *yram) { *code = *yram = false; @@ -1596,7 +1596,7 @@ static unsigned int dsp_chip_to_dsp_addx(unsigned int chip_addx, return Y_OFF(chip_addx); } - return (unsigned int)INVALID_CHIP_ADDRESS; + return INVALID_CHIP_ADDRESS; } /* @@ -1620,7 +1620,7 @@ static int dsp_dma_setup_common(struct hda_codec *codec, { int status = 0; unsigned int chnl_prop; - unsigned int dsp_addx; + unsigned long dsp_addx; unsigned int active; bool code, yram; @@ -1712,7 +1712,7 @@ static int dsp_dma_setup_common(struct hda_codec *codec, snd_printdd(KERN_INFO " dsp_dma_setup_common() Write IRQCNT"); snd_printdd( - "ChipA=0x%x,DspA=0x%x,dmaCh=%u, " + "ChipA=0x%x,DspA=0x%lx,dmaCh=%u, " "CHSEL=0x%x,CHPROP=0x%x,Active=0x%x\n", chip_addx, dsp_addx, dma_chan, port_map_mask, chnl_prop, active); @@ -1732,7 +1732,7 @@ static int dsp_dma_setup(struct hda_codec *codec, { int status = 0; bool code, yram; - unsigned int dsp_addx; + unsigned long dsp_addx; unsigned int addr_field; unsigned int incr_field; unsigned int base_cnt;

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