June 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - power setting error check
by Wang Xingchao 07 Jun '12

07 Jun '12

codec may reject power state transition requests(reporting PS-ERROR set), in that case we re-issue a power state setting and check error bit again. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d0ca370..bb64eba 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3526,6 +3526,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { + int state, count = 0; if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3537,9 +3538,19 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); msleep(epss? 10:100); } +again: snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); snd_hda_codec_set_power_to_all(codec, fg, power_state, true); + state = snd_hda_codec_read(codec, fg, 0, + AC_VERB_GET_POWER_STATE, 0); + if (state & AC_PWRST_ERROR) { + /* repeat power states setting at most 10 times*/ + if (count++ > 10) + snd_printk(KERN_WARNING "Power states transition reject!\n"); + else + goto again; + } } /* modem wake on ring: transition from D2 to D0 in less than 2ms. For modems, -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - modem function group wake on ring support
by Wang Xingchao 07 Jun '12

07 Jun '12

D2 allows for the lowest possible power consuming state from which it can return to "fully on" state D0 within 2ms. For modems, it's the "wake on ring" power state. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2737fe7..d0ca370 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3542,6 +3542,18 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_set_power_to_all(codec, fg, power_state, true); } +/* modem wake on ring: transition from D2 to D0 in less than 2ms. For modems, + * this is the wake-on-ring power state */ +static void hda_modem_wake_on_ring(struct hda_codec *codec, bool enter) +{ + if (codec->mfg) { + if (enter) + hda_set_power_state(codec, codec->mfg, AC_PWRST_D2); + else + hda_set_power_state(codec, codec->mfg, AC_PWRST_D0); + } +} + #ifdef CONFIG_SND_HDA_HWDEP /* execute additional init verbs */ static void hda_exec_init_verbs(struct hda_codec *codec) -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ALSA: hda - HDMI Audio init all connectors
by Takashi Iwai 07 Jun '12

07 Jun '12

At Thu, 07 Jun 2012 10:00:24 +0100, Steven Newbury wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 07/06/12 09:47, Takashi Iwai wrote: > > At the next time you submit a patch, please add Cc to maintainer > > and subsystem ML. Otherwise it'd be easily overlooked. > > > I meant to, somehow I managed to send it off before adding the CC, > probably too early in the morning! > > > At Thu, 07 Jun 2012 06:52:59 +0100, Steven Newbury wrote: > >> > >> When VGA_SWITCHEROO support is enabled hda_intel initialises the > >> HDMI audio device on the current VGA device. When it's not > >> enabled it only initialises the HDMI device on the default VGA > >> adaptor, this means secondary cards get no audio support which is > >> very unhelpful for multi-seat! > >> > >> With this patch, when SUPPORT_VGA_SWITCHEROO is disabled > >> hda_intel initialises all HDMI audio devices, not just the > >> default VGA. > >> > >> Signed-off-by: Steven Newbury <steve(a)snewbury.org.uk> > > > > OK, it must be fixed indeed. The bug was introduced due to the > > change in VGA-switcheroo side -- originally check_hdmi_disabled() > > really checked the VGA-switcheroo state, so it returned the correct > > value in the earlier version. > > > > In anyway, your patch looks almost OK, but it'd be better to move > > ifdef and simplify like below. Could you check whether it works > > for you? > > > Looks better, saves the pointless function call. I'll give it spin on > my next boot. Thanks! Meanwhile I found that the current version is also buggy for some setups with VGA-switcheroo. The current version won't work, for example, when VGA-switcheroo is built-in for systems without switcheroo. I'll need to cook up the fix for 3.5-rc2 in addition to your fix. Takashi

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[alsa-devel] [PATCH] ALSA: hda - check proper return value
by Wang Xingchao 07 Jun '12

07 Jun '12

snd_hda_param_read() return value -1 means error, others are responses Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 79d0188..99316a7 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -451,7 +451,7 @@ static void print_power_state(struct snd_info_buffer *buffer, int sup = snd_hda_param_read(codec, nid, AC_PAR_POWER_STATE); int pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); - if (sup) + if (sup != -1) snd_iprintf(buffer, " Power states: %s\n", bits_names(sup, names, ARRAY_SIZE(names))); -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: hda - check supported power states
by Wang Xingchao 07 Jun '12

07 Jun '12

Add function to check whether power states supported by specific codec node. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 41ca803..b89c8ec 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3505,6 +3505,22 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all); /* + * supported power states check + */ +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); + + if (sup < 0) + return false; + if (sup & power_state) + return true; + else + return false; +} + +/* * set power state of the codec */ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, -- 1.7.9.5

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[alsa-devel] [PATCH V2] ALSA: hda - reduce msleep time if EPSS power states supported
by Wang Xingchao 07 Jun '12

07 Jun '12

if EPSS supported, transition from D3 state to D0 state in less than 10ms Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b89c8ec..2737fe7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3532,8 +3532,11 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, } /* this delay seems necessary to avoid click noise at power-down */ - if (power_state == AC_PWRST_D3) - msleep(100); + if (power_state == AC_PWRST_D3) { + /* transition time less than 10ms for power down */ + bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); + msleep(epss ? 10 : 100); + } snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); snd_hda_codec_set_power_to_all(codec, fg, power_state, true); -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ALSA: hda - HDMI Audio init all connectors
by Takashi Iwai 07 Jun '12

07 Jun '12

At the next time you submit a patch, please add Cc to maintainer and subsystem ML. Otherwise it'd be easily overlooked. At Thu, 07 Jun 2012 06:52:59 +0100, Steven Newbury wrote: > > When VGA_SWITCHEROO support is enabled hda_intel initialises the HDMI audio device on the current VGA device. When it's not enabled it only initialises the HDMI device on the default VGA adaptor, this means secondary cards get no audio support which is very unhelpful for multi-seat! > > With this patch, when SUPPORT_VGA_SWITCHEROO is disabled hda_intel initialises all HDMI audio devices, not just the default VGA. > > Signed-off-by: Steven Newbury <steve(a)snewbury.org.uk> OK, it must be fixed indeed. The bug was introduced due to the change in VGA-switcheroo side -- originally check_hdmi_disabled() really checked the VGA-switcheroo state, so it returned the correct value in the earlier version. In anyway, your patch looks almost OK, but it'd be better to move ifdef and simplify like below. Could you check whether it works for you? Also, please give a patch that is applicable directly via git-am. Apparently you created it via git show. Instead, use git-format-patch or use --pretty=email for git-show. Or, send the patch directly via git-send-email. thanks, Takashi --- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2b6392b..2eb1b47 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2484,9 +2484,9 @@ static void azx_notifier_unregister(struct azx *chip) static int DELAYED_INIT_MARK azx_first_init(struct azx *chip); static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip); +#ifdef SUPPORT_VGA_SWITCHEROO static struct pci_dev __devinit *get_bound_vga(struct pci_dev *pci); -#ifdef SUPPORT_VGA_SWITCHEROO static void azx_vs_set_state(struct pci_dev *pci, enum vga_switcheroo_state state) { @@ -2578,6 +2578,7 @@ static int __devinit register_vga_switcheroo(struct azx *chip) #else #define init_vga_switcheroo(chip) /* NOP */ #define register_vga_switcheroo(chip) 0 +#define check_hdmi_disabled(pci) false #endif /* SUPPORT_VGA_SWITCHER */ /* @@ -2638,6 +2639,7 @@ static int azx_dev_free(struct snd_device *device) return azx_free(device->device_data); } +#ifdef SUPPORT_VGA_SWITCHEROO /* * Check of disabled HDMI controller by vga-switcheroo */ @@ -2676,6 +2678,7 @@ static bool __devinit check_hdmi_disabled(struct pci_dev *pci) } return vga_inactive; } +#endif /* SUPPORT_VGA_SWITCHEROO */ /* * white/black-listing for position_fix

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Re: [alsa-devel] 3.5-rc1 snd_usb_audio stream format breakage
by Takashi Iwai 07 Jun '12

07 Jun '12

At Thu, 7 Jun 2012 03:38:30 +0000, Jamie Heilman wrote: > > With the USB audio streaming logic changes, my HeadRoom Bithead > (08bb:2902 Burr-Brown PCM2902) no longer works. Symptoms include the > loss of most of the normally available formats leaving only: > > root@cucamonga:~# cat /proc/asound/bithead/stream0 > Burr-Brown from TI USB Audio CODEC at usb-0000:00:1a.7-1.1.4, full speed : USB Audio > > Playback: > Status: Stop > Interface 1 > Altset 6 > Format: U8 > Channels: 1 > Endpoint: 2 OUT (ADAPTIVE) > Rates: 32000, 44100, 48000 > > Capture: > Status: Stop > Interface 2 > Altset 18 > Format: S8 > Channels: 1 > Endpoint: 4 IN (SYNC) > Rates: 11025 > > (normally 2 channel S16_LE is available, with capture rates up to > 48kHz ... only the last altset's are being reported now) and a slew of > proc registration warnings logged: > > ------------[ cut here ]------------ > WARNING: at fs/proc/generic.c:586 proc_register+0x169/0x1a6() > Hardware name: Precision WorkStation T3400 > proc_dir_entry 'card1/pcm0p' already registered > Modules linked in: snd_usb_audio(+) snd_usbmidi_lib snd_hda_intel(+) snd_hda_codec snd_hwdep snd_seq_midi snd_seq_midi_event snd_pcm snd_page_alloc snd_rawmidi snd_seq snd_seq_device snd_timer snd soundcore dcdbas floppy sr_mod usb_storage cdrom kvm_intel psmouse kvm tg3 evdev xfs dm_mod raid1 md_mod > Pid: 1152, comm: modprobe Not tainted 3.5.0-rc1-00110-g71fae7e #2 > Call Trace: > [<ffffffff8102a400>] warn_slowpath_common+0x83/0x9c > [<ffffffff8102a4bc>] warn_slowpath_fmt+0x46/0x48 > [<ffffffff813829ad>] ? add_preempt_count+0x39/0x3b > [<ffffffff811292f0>] proc_register+0x169/0x1a6 > [<ffffffff8112962e>] create_proc_entry+0x74/0x8c > [<ffffffffa018eb63>] snd_info_register+0x3e/0xc3 [snd] > [<ffffffffa01fde2e>] snd_pcm_new_stream+0xb1/0x404 [snd_pcm] > [<ffffffffa024861f>] snd_usb_add_audio_stream+0xd2/0x230 [snd_usb_audio] > [<ffffffffa0241d33>] ? snd_usb_parse_audio_format+0x252/0x34f [snd_usb_audio] > [<ffffffff810d6b17>] ? kmem_cache_alloc_trace+0xab/0xbb > [<ffffffffa0248c29>] snd_usb_parse_audio_interface+0x4ac/0x567 [snd_usb_audio] > [<ffffffffa023f0ff>] snd_usb_create_stream+0xe9/0x125 [snd_usb_audio] > [<ffffffffa023f9b1>] usb_audio_probe+0x62a/0x72c [snd_usb_audio] > [<ffffffff812c49f5>] usb_probe_interface+0x136/0x1b1 > [<ffffffff81287e66>] ? driver_probe_device+0x1d3/0x1d3 > [<ffffffff81287d43>] driver_probe_device+0xb0/0x1d3 > [<ffffffff81287ec5>] __driver_attach+0x5f/0x83 > [<ffffffff812863a2>] bus_for_each_dev+0x54/0x86 > [<ffffffff81287730>] driver_attach+0x1e/0x20 > [<ffffffff8128747f>] bus_add_driver+0xfd/0x23a > [<ffffffff812882e8>] driver_register+0x93/0x10e > [<ffffffff812c383f>] usb_register_driver+0x7e/0x12a > [<ffffffffa0257000>] ? 0xffffffffa0256fff > [<ffffffffa0257000>] ? 0xffffffffa0256fff > [<ffffffffa0257040>] snd_usb_audio_init+0x40/0x1000 [snd_usb_audio] > [<ffffffff810001ff>] do_one_initcall+0x7f/0x134 > [<ffffffffa0257000>] ? 0xffffffffa0256fff > [<ffffffff810688d4>] sys_init_module+0x16cf/0x195e > [<ffffffff8106516c>] ? free_notes_attrs+0x4d/0x4d > [<ffffffff810c38cb>] ? mmap_region+0x1c7/0x469 > [<ffffffff813856d2>] system_call_fastpath+0x16/0x1b > ---[ end trace c1baea1a25a91564 ]--- Looks like the PCM substream assignment is buggy. Does the patch (untested at all) below help? thanks, Takashi --- diff --git a/sound/usb/card.h b/sound/usb/card.h index 0d37238..2b9ffff 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -119,6 +119,7 @@ struct snd_usb_substream { unsigned long unlink_mask; /* bitmask of unlinked urbs */ /* data and sync endpoints for this stream */ + unsigned int ep_num; /* the endpoint number */ struct snd_usb_endpoint *data_endpoint; struct snd_usb_endpoint *sync_endpoint; unsigned long flags; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 6b7d7a2..083ed81 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -97,6 +97,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->formats |= fp->formats; subs->num_formats++; subs->fmt_type = fp->fmt_type; + subs->ep_num = fp->endpoint; } /* @@ -119,9 +120,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; - if (!subs->data_endpoint) - continue; - if (subs->data_endpoint->ep_num == fp->endpoint) { + if (subs->ep_num == fp->endpoint) { list_add_tail(&fp->list, &subs->fmt_list); subs->num_formats++; subs->formats |= fp->formats; @@ -134,7 +133,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; - if (subs->data_endpoint) + if (subs->ep_num) continue; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0)

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[alsa-devel] [PATCH V2] ALSA: hda - add power states information in proc
by Wang Xingchao 06 Jun '12

06 Jun '12

add more power states information: - reset status - clock stop ok - power states error Output like: Power: setting=D0, actual=D0, Error, Clock-stop-OK, Setting-reset Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_proc.c | 9 ++++++++- 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 4fc3960..71864cdd 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -323,6 +323,9 @@ enum { #define AC_PWRST_D1 0x01 #define AC_PWRST_D2 0x02 #define AC_PWRST_D3 0x03 +#define AC_PWRST_ERROR (1<<8) +#define AC_PWRST_CLK_STOP_OK (1<<9) +#define AC_PWRST_SETTING_RESET (1<<10) /* Processing capabilies */ #define AC_PCAP_BENIGN (1<<0) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index e59e2f0..79d0188 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -455,10 +455,17 @@ static void print_power_state(struct snd_info_buffer *buffer, snd_iprintf(buffer, " Power states: %s\n", bits_names(sup, names, ARRAY_SIZE(names))); - snd_iprintf(buffer, " Power: setting=%s, actual=%s\n", + snd_iprintf(buffer, " Power: setting=%s, actual=%s", get_pwr_state(pwr & AC_PWRST_SETTING), get_pwr_state((pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT)); + if (pwr & AC_PWRST_ERROR) + snd_iprintf(buffer, " , Error"); + if (pwr & AC_PWRST_CLK_STOP_OK) + snd_iprintf(buffer, " , Clock-stop-OK"); + if (pwr & AC_PWRST_SETTING_RESET) + snd_iprintf(buffer, " , Setting-reset"); + snd_iprintf(buffer, "\n"); } static void print_unsol_cap(struct snd_info_buffer *buffer, -- 1.7.9.5

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Re: [alsa-devel] [PATCH v2 2/2] ASoC: codecs: Add DA732x codec driver
by Adam Thomson 06 Jun '12

06 Jun '12

On Wed, June 4, 2012 at 15:31, Mark Brown wrote: > A few small things below, very minor and easy to correct though. > > > + { DA732X_REG_STATUS_EXT , 0x00 }, > > + { DA732X_REG_STATUS , 0x00 }, > > > + { DA732X_REG_MBOX0 , 0x00 }, > > + { DA732X_REG_MBOX1 , 0x00 }, > > + { DA732X_REG_MBOX2 , 0x00 }, > > + { DA732X_REG_MBOX_STATUS , 0x00 }, > > > + { DA732X_REG_ID , 0x00 }, > > > + { DA732X_REG_DMA_STATUS , 0x00 }, > > + { DA732X_REG_BROWNOUT , 0x00 }, > > I would be very surprised if some or all of the above regsters aren't > volatile and therefore shouldn't have default specified. Correct. Will remove those from the cache. > > +static int da732x_hpf_set(struct snd_kcontrol *kcontrol, > > + struct snd_ctl_elem_value *ucontrol) > > Why doesn't a standard enum work for this? This is because the two bits that are required for this are not adjacent in the register, so we use our own get/set methods to control them. > > + /* High Pass Filters */ > > + SOC_ENUM_EXT("DAC1 High Pass Filter Mode Switch", > > + da732x_dac1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), > > This shouldn't be a Switch, it's an enumeration. Switches are plain > boolean controls. Ok, fair point. Will rename accordingly. > > + SOC_SINGLE("ADC1 Equalizer Switch", DA732X_REG_ADC1_EQ5, > > + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), > > + SOC_SINGLE_TLV("ADC1 EQ Band 1 Volume", DA732X_REG_ADC1_EQ12, > > + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, > > + DA732X_INVERT, eq_band_pga_tlv), > > I'd suggest calling the Switch "ADC1 EQ Switch" for consistency with the > gains for the bands (or expanding EQ in the gains but that'll make for a > lot of really long control names). Again, makes sense. Will update all the ADC & DAC instances to align. > > + case SND_SOC_DAPM_POST_PMU: > > + if (w->reg == DA732X_REG_ADC1_PD) > > + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, > > + DA732X_ADCA_BB_CLK_EN, > > + DA732X_ADCA_BB_CLK_EN); > > + else > > + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, > > + DA732X_ADCC_BB_CLK_EN, > > + DA732X_ADCC_BB_CLK_EN); > > This reads like it should be a switch statement. The effect will be the > same but it's a bit clearer. Ok, don't mind changing it if you think it's clearer. > > + da732x->regmap = devm_regmap_init_i2c(i2c, &da732x_regmap); > > + if (IS_ERR(da732x->regmap)) { > > + ret = PTR_ERR(da732x->regmap); > > + dev_err(&i2c->dev, "Failed to initialize regmap\n"); > > + goto err; > > + } > > + > > + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x, > > + da732x_dai, ARRAY_SIZE(da732x_dai)); > > The register cache defaults make it look like there's a device ID > (or possibly chip revision?) register - it'd be good practice to read it > here and either verify that it has the expected value or log it > depending on which it is. Is revision information. Will log this. > > +static const struct i2c_device_id da732x_i2c_id[] = { > > + { "da732x", 0}, > > + { } > > +}; > > +MODULE_DEVICE_TABLE(i2c, da732x_i2c_id); > > Good practice would be to enumerate all the device IDs (so users can > just register devices) but it doesn't really matter. For now will stick with this.

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[alsa-devel] [PATCH] ASoC:isabelle:Remove regmap_exit()
by MR.Swami.Reddy＠ti.com 06 Jun '12

06 Jun '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> ASoC:isabelle:With devm_ APIs regmap_exit() not needed, so remove regmap_exit(). Signed-off-by: Vishwas A Deshpande <vishwas.a.deshpande(a)ti.com> Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> --- sound/soc/codecs/isabelle.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 69228ec..bcc77ef 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1141,7 +1141,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, ARRAY_SIZE(isabelle_dai)); if (ret < 0) { dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); - regmap_exit(dev_get_regmap(&i2c->dev, NULL)); return ret; } @@ -1151,7 +1150,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, static int __devexit isabelle_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - regmap_exit(dev_get_regmap(&client->dev, NULL)); return 0; }

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[alsa-devel] [PATCH v3] ASoC: omap-abe-twl6040: Add device tree support
by Peter Ujfalusi 06 Jun '12

06 Jun '12

When the board boots with device tree the driver will receive the name of the card, DAPM routing map, phandle for the audio components described in the dts file, mclk speed, and the possibility of detecting the jack detection. The card will be set up based on this information. Since the routing is provided via DT we can mark the card fully routed so core can take care of disconnecting the unused pins. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reviwed-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Hi, Changes since v2 (resend): - Do not override the return value from snd_soc_of_parse_audio_routing in case of error. Regards, Peter .../devicetree/bindings/sound/omap-abe-twl6040.txt | 91 ++++++++++++ sound/soc/omap/omap-abe-twl6040.c | 145 ++++++++++++++++---- 2 files changed, 206 insertions(+), 30 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt diff --git a/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt new file mode 100644 index 0000000..65dec87 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt @@ -0,0 +1,91 @@ +* Texas Instruments OMAP4+ and twl6040 based audio setups + +Required properties: +- compatible: "ti,abe-twl6040" +- ti,model: Name of the sound card ( for example "SDP4430") +- ti,mclk-freq: MCLK frequency for HPPLL operation +- ti,mcpdm: phandle for the McPDM node +- ti,twl6040: phandle for the twl6040 core node +- ti,audio-routing: List of connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. + +Optional properties: +- ti,dmic: phandle for the OMAP dmic node if the machine have it connected +- ti,jack_detection: Need to be set to <1> if the board capable to detect jack + insertion, removal. + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Earphone Spk + * Ext Spk + * Line Out + * Vibrator + * Headset Mic + * Main Handset Mic + * Sub Handset Mic + * Line In + * Digital Mic + +twl6040 pins: + * HSOL + * HSOR + * EP + * HFL + * HFR + * AUXL + * AUXR + * VIBRAL + * VIBRAR + * HSMIC + * MAINMIC + * SUBMIC + * AFML + * AFMR + + * Headset Mic Bias + * Main Mic Bias + * Digital Mic1 Bias + * Digital Mic2 Bias + +Digital mic pins: + * DMic + +Example: + +sound { + compatible = "ti,abe-twl6040"; + ti,model = "SDP4430"; + + ti,jack-detection = <1>; + ti,mclk-freq = <38400000>; + + ti,mcpdm = <&mcpdm>; + ti,dmic = <&dmic>; + + ti,twl6040 = <&twl6040>; + + /* Audio routing */ + ti,audio-routing = + "Headset Stereophone", "HSOL", + "Headset Stereophone", "HSOR", + "Earphone Spk", "EP", + "Ext Spk", "HFL", + "Ext Spk", "HFR", + "Line Out", "AUXL", + "Line Out", "AUXR", + "Vibrator", "VIBRAL", + "Vibrator", "VIBRAR", + "HSMIC", "Headset Mic", + "Headset Mic", "Headset Mic Bias", + "MAINMIC", "Main Handset Mic", + "Main Handset Mic", "Main Mic Bias", + "SUBMIC", "Sub Handset Mic", + "Sub Handset Mic", "Main Mic Bias", + "AFML", "Line In", + "AFMR", "Line In", + "DMic", "Digital Mic", + "Digital Mic", "Digital Mic1 Bias"; +}; diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 9d93793..1343147 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -25,6 +25,7 @@ #include <linux/mfd/twl6040.h> #include <linux/platform_data/omap-abe-twl6040.h> #include <linux/module.h> +#include <linux/of.h> #include <sound/core.h> #include <sound/pcm.h> @@ -43,6 +44,8 @@ struct abe_twl6040 { int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ + + struct platform_device *dmic_codec_dev; }; static int omap_abe_hw_params(struct snd_pcm_substream *substream, @@ -185,17 +188,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) int hs_trim; int ret = 0; - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); - twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); - twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); - twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - /* * Configure McPDM offset cancellation based on the HSOTRIM value from * twl6040. @@ -216,6 +208,24 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } + /* + * NULL pdata means we booted with DT. In this case the routing is + * provided and the card is fully routed, no need to mark pins. + */ + if (!pdata) + return ret; + + /* Disable not connected paths if not used */ + twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); + twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); + twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); + twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); + twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); + twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); + return ret; } @@ -270,52 +280,116 @@ static struct snd_soc_card omap_abe_card = { static __devinit int omap_abe_probe(struct platform_device *pdev) { struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); + struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; struct abe_twl6040 *priv; int num_links = 0; - int ret; + int ret = 0; card->dev = &pdev->dev; - if (!pdata) { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; - } - priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); if (priv == NULL) return -ENOMEM; - if (pdata->card_name) { - card->name = pdata->card_name; + priv->dmic_codec_dev = ERR_PTR(-EINVAL); + + if (node) { + struct device_node *dai_node; + + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + ret = snd_soc_of_parse_audio_routing(card, + "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, + "Error while parsing DAPM routing\n"); + return ret; + } + + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_dai_of_node = dai_node; + + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_dai_of_node = dai_node; + + priv->dmic_codec_dev = platform_device_register_simple( + "dmic-codec", -1, NULL, 0); + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, + "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); + } + } else { + num_links = 1; + } + + of_property_read_u32(node, "ti,jack-detection", + &priv->jack_detection); + of_property_read_u32(node, "ti,mclk-freq", + &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; + } + + omap_abe_card.fully_routed = 1; + } else if (pdata) { + if (pdata->card_name) { + card->name = pdata->card_name; + } else { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + if (pdata->has_dmic) + num_links = 2; + else + num_links = 1; + + priv->jack_detection = pdata->jack_detection; + priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Card name is not provided\n"); + dev_err(&pdev->dev, "Missing pdata\n"); return -ENODEV; } - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; - if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); - return -ENODEV; + ret = -ENODEV; + goto err_unregister; } - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - card->dai_link = abe_twl6040_dai_links; card->num_links = num_links; snd_soc_card_set_drvdata(card, priv); ret = snd_soc_register_card(card); - if (ret) + if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + goto err_unregister; + } + + return 0; + +err_unregister: + if (!IS_ERR(priv->dmic_codec_dev)) + platform_device_unregister(priv->dmic_codec_dev); return ret; } @@ -323,17 +397,28 @@ static __devinit int omap_abe_probe(struct platform_device *pdev) static int __devexit omap_abe_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); + struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); snd_soc_unregister_card(card); + if (!IS_ERR(priv->dmic_codec_dev)) + platform_device_unregister(priv->dmic_codec_dev); + return 0; } +static const struct of_device_id omap_abe_of_match[] = { + {.compatible = "ti,abe-twl6040", }, + { }, +}; +MODULE_DEVICE_TABLE(of, omap_abe_of_match); + static struct platform_driver omap_abe_driver = { .driver = { .name = "omap-abe-twl6040", .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, + .of_match_table = omap_abe_of_match, }, .probe = omap_abe_probe, .remove = __devexit_p(omap_abe_remove), -- 1.7.8.6

1 0

[alsa-devel] [PATCH 9/8] printk: Only look for prefix levels in kernel messages
by Joe Perches 06 Jun '12

06 Jun '12

vprintk_emit prefix parsing should only be done for internal kernel messages. This allows existing behavior to be kept in all cases. Signed-off-by: Joe Perches <joe(a)perches.com> --- kernel/printk.c | 32 +++++++++++++++++--------------- 1 files changed, 17 insertions(+), 15 deletions(-) diff --git a/kernel/printk.c b/kernel/printk.c index 5cd73f7..4e72c07 100644 --- a/kernel/printk.c +++ b/kernel/printk.c @@ -1267,7 +1267,6 @@ asmlinkage int vprintk_emit(int facility, int level, static char cont_buf[LOG_LINE_MAX]; static size_t cont_len; static int cont_level; - int kern_level; static struct task_struct *cont_task; static char textbuf[LOG_LINE_MAX]; char *text = textbuf; @@ -1329,21 +1328,24 @@ asmlinkage int vprintk_emit(int facility, int level, newline = true; } - /* strip syslog prefix and extract log level or control flags */ - kern_level = printk_get_level(text); - if (kern_level) { - const char *end_of_header = printk_skip_level(text); - switch (kern_level) { - case '0' ... '7': - if (level == -1) - level = kern_level - '0'; - case 'd': /* KERN_DEFAULT */ - prefix = true; - case 'c': /* KERN_CONT */ - break; + /* strip kernel syslog prefix and extract log level or control flags */ + if (facility == 0) { + int kern_level = printk_get_level(text); + + if (kern_level) { + const char *end_of_header = printk_skip_level(text); + switch (kern_level) { + case '0' ... '7': + if (level == -1) + level = kern_level - '0'; + case 'd': /* KERN_DEFAULT */ + prefix = true; + case 'c': /* KERN_CONT */ + break; + } + text_len -= end_of_header - text; + text = (char *)end_of_header; } - text_len -= end_of_header - text; - text = (char *)end_of_header; } if (level == -1)

1 0

Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

On Wed, 2012-06-06 at 03:10 +0200, Kay Sievers wrote: > On Wed, Jun 6, 2012 at 2:46 AM, Andrew Morton <akpm(a)linux-foundation.org> wrote: > > On Tue, 05 Jun 2012 17:40:05 -0700 Joe Perches <joe(a)perches.com> wrote: > > > >> On Tue, 2012-06-05 at 17:37 -0700, Andrew Morton wrote: > >> > On Tue, 05 Jun 2012 17:07:27 -0700 Joe Perches <joe(a)perches.com> wrote: > >> > > >> > > On Tue, 2012-06-05 at 16:58 -0700, Andrew Morton wrote: > >> > > > echo "\0014Hello Joe" > /dev/kmsg > >> > > > >> > > # echo -e "\x014Hello Me" > /dev/kmsg > >> > > gives: > >> > > 12,778,4057982669;Hello Me > >> > > >> > That's changed behavior. > >> > >> Which is an improvement too. > > > > No it isn't. It exposes internal kernel implementation details in > > random weird inexplicable ways. It doesn't seem at all important > > though. > > > >> I very much doubt a single app will change > >> because of this. > > > > I doubt it as well. > > Yeah, the value of injecting such binary data is kind of questionable. :) > > Joe, maybe you can change printk_emit() to skip the prefix > detection/stripping if a prefix is already passed to the function? Sure, it's pretty trivial. Perhaps all binary data data should be elided. Maybe print . for all non-printable ascii chars.

1 0

Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Andrew Morton 06 Jun '12

06 Jun '12

On Tue, 05 Jun 2012 17:07:27 -0700 Joe Perches <joe(a)perches.com> wrote: > On Tue, 2012-06-05 at 16:58 -0700, Andrew Morton wrote: > > echo "\0014Hello Joe" > /dev/kmsg > > # echo -e "\x014Hello Me" > /dev/kmsg > gives: > 12,778,4057982669;Hello Me That's changed behavior. On Wed, 6 Jun 2012 02:28:39 +0200 Kay Sievers <kay(a)vrfy.org> wrote: > Yeah, but printk_emit() will not try to parse it? I did not check, but > with your change, the prefix parsing in printk_emit() is still skipped > if a level is given as a parameter to printk_emit(), right? printk_emit() does parse the leading \0014, and then skips over it, removing it from the output stream. printk_emit() then throws away the resulting level because devkmsg_writev() did not pass in level==-1.

2 2

Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

On Wed, 2012-06-06 at 02:28 +0200, Kay Sievers wrote: > On Wed, Jun 6, 2012 at 2:19 AM, Joe Perches <joe(a)perches.com> wrote: > > On Wed, 2012-06-06 at 02:13 +0200, Kay Sievers wrote: > >> The question is what happens if you inject your new binary two-byte > >> prefix, like: > >> echo -e "\x01\x02Hello" > /dev/kmsg > > > > It's not a 2 byte binary. > > It's a leading ascii SOH and a standard ascii char > > '0' ... '7' or 'd'. > > > > #define KERN_EMERG KERN_SOH "0" /* system is unusable */ > > #define KERN_ALERT KERN_SOH "1" /* action must be taken immediately */ > > etc... > > Ok. > > >> And if that changes the log-level to "2" instead of the default "4"? > > > > No it doesn't. > > So: > echo -e "\x012Hello" > /dev/kmsg > is still level 4? Sounds all fine then. Yes. # echo -e "\x012Hello again Kay" > /dev/kmsg gives: 12,780,6031964979;Hello again Kay > > It's not triggering that because devkmsg_writev does > > prefix parsing only on the old "<n>" form. > > Yeah, but printk_emit() will not try to parse it? I did not check, but > with your change, the prefix parsing in printk_emit() is still skipped > if a level is given as a parameter to printk_emit(), right? If level is not -1, then whatever prefix level the string has is ignored by vprintk_emit. from vprintk_emit: /* strip syslog prefix and extract log level or control flags */ kern_level = printk_get_level(text); if (kern_level) { const char *end_of_header = printk_skip_level(text); switch (kern_level) { case '0' ... '7': if (level == -1) level = kern_level - '0'; case 'd': /* Strip d KERN_DEFAULT, start new line */ plen = 0; default: if (!new_text_line) { log_buf_emit_char('\n'); new_text_line = 1; } } text_len -= end_of_header - text; text = (char *)end_of_header; } Only level == -1 will use the prefix level. devkmsg_writev always passes a non -1 level. cheers, Joe

1 0

Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

On Wed, 2012-06-06 at 02:13 +0200, Kay Sievers wrote: > The question is what happens if you inject your new binary two-byte > prefix, like: > echo -e "\x01\x02Hello" > /dev/kmsg It's not a 2 byte binary. It's a leading ascii SOH and a standard ascii char '0' ... '7' or 'd'. #define KERN_EMERG KERN_SOH "0" /* system is unusable */ #define KERN_ALERT KERN_SOH "1" /* action must be taken immediately */ etc... > And if that changes the log-level to "2" instead of the default "4"? No it doesn't. It's not triggering that because devkmsg_writev does prefix parsing only on the old "<n>" form.

1 0

Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

On Wed, 2012-06-06 at 01:48 +0200, Kay Sievers wrote: > On Wed, Jun 6, 2012 at 1:43 AM, Joe Perches <joe(a)perches.com> wrote: > > On Wed, 2012-06-06 at 01:39 +0200, Kay Sievers wrote: > > >> > # echo "\001Hello Andrew" > /dev/kmsg > >> > /dev/kmsg has > >> > 12,774,2462339252;\001Hello Andrew > >> > >> Try "echo -e"? The stuff is copied verbatim otherwise. > > > > # echo -e "\001Hello Kay" > /dev/kmsg > > gives > > 12,776,3046752764;\x01Hello Kay > > Don't you need two bytes to trigger the logic? Yes. Angle brackets fore and aft. If you use a "<n>" prefix for write to /dev/kmsg, then it's supported just like before. Otherwise, it's emitted at KERN_DEFAULT. cheers, Joe

2 2

Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

On Wed, 2012-06-06 at 01:39 +0200, Kay Sievers wrote: > Try "echo -e"? The stuff is copied verbatim otherwise. btw: I didn't change the devkmsg_writev function which does the decoding of any "<n>" prefix for printk_emit.

1 0

Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

On Wed, 2012-06-06 at 01:39 +0200, Kay Sievers wrote: > On Wed, Jun 6, 2012 at 1:35 AM, Joe Perches <joe(a)perches.com> wrote: > > On Tue, 2012-06-05 at 16:29 -0700, Andrew Morton wrote: > >> What about writes starting with \001n? AFACIT, that will be stripped > >> away and the printk level will be altered. This is new behavior. > > > > Nope. > > > > # echo "\001Hello Andrew" > /dev/kmsg > > /dev/kmsg has > > 12,774,2462339252;\001Hello Andrew > > Try "echo -e"? The stuff is copied verbatim otherwise. # echo -e "\001Hello Kay" > /dev/kmsg gives 12,776,3046752764;\x01Hello Kay

1 0

[alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

KERN_<LEVEL> currently takes up 3 bytes. Shrink the kernel size by using an ASCII SOH and then the level byte. Remove the need for KERN_CONT. Convert directly embedded uses of <.> to KERN_<LEVEL> Joe Perches (8): printk: Add generic functions to find KERN_<LEVEL> headers printk: Add kern_levels.h to make KERN_<LEVEL> available for asm use arch: Remove direct definitions of KERN_<LEVEL> uses btrfs: Use printk_get_level and printk_skip_level, add __printf, fix fallout sound: Use printk_get_level and printk_skip_level staging: wlags49_h2: Remove direct declarations of KERN_<LEVEL> prefixes printk: Convert the format for KERN_<LEVEL> to a 2 byte pattern printk: Remove the now unnecessary "C" annotation for KERN_CONT arch/arm/lib/io-acorn.S | 3 +- arch/arm/vfp/vfphw.S | 7 +++-- arch/frv/kernel/kernel_thread.S | 2 +- drivers/staging/wlags49_h2/hcf.c | 8 +++--- drivers/staging/wlags49_h2/wl_main.c | 4 +- fs/btrfs/ctree.h | 13 ++++++++++ fs/btrfs/disk-io.c | 2 +- fs/btrfs/relocation.c | 2 +- fs/btrfs/super.c | 41 +++++++++++++++++++++++++++++----- include/linux/kern_levels.h | 25 ++++++++++++++++++++ include/linux/printk.h | 41 +++++++++++++++++++-------------- kernel/printk.c | 14 +++++++---- sound/core/misc.c | 13 +++++++--- 13 files changed, 130 insertions(+), 45 deletions(-) create mode 100644 include/linux/kern_levels.h -- 1.7.8.111.gad25c.dirty

2 7

Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Andrew Morton 06 Jun '12

06 Jun '12

On Wed, 06 Jun 2012 00:55:00 +0200 Kay Sievers <kay(a)vrfy.org> wrote: > On Tue, 2012-06-05 at 14:28 -0700, Andrew Morton wrote: > > > devkmsg_writev() does weird and wonderful things with > > facilities/levels. That function incorrectly returns "success" when > > copy_from_user() faults, btw. > > Oh. Better? > > Thanks, > Kay > > > From: Kay Sievers <kay(a)vrfy.org> > Subject: kmsg: /dev/kmsg - properly return possible copy_from_user() failure > > Reported-By: Andrew Morton <akpm(a)linux-foundation.org> > Signed-off-by: Kay Sievers <kay(a)vrfy.org > --- > printk.c | 4 +++- > 1 file changed, 3 insertions(+), 1 deletion(-) > > diff --git a/kernel/printk.c b/kernel/printk.c > index 32462d2..6bdacab 100644 > --- a/kernel/printk.c > +++ b/kernel/printk.c > @@ -365,8 +365,10 @@ static ssize_t devkmsg_writev(struct kiocb *iocb, const struct iovec *iv, > > line = buf; > for (i = 0; i < count; i++) { > - if (copy_from_user(line, iv[i].iov_base, iv[i].iov_len)) > + if (copy_from_user(line, iv[i].iov_base, iv[i].iov_len)) { > + ret = -EFAULT; > goto out; > + } > line += iv[i].iov_len; > } Strictly speaking, when write() encounters an error it should return number-of-bytes-written, or an errno if it wrote zero bytes. So something like --- a/kernel/printk.c~a +++ a/kernel/printk.c @@ -355,7 +355,7 @@ static ssize_t devkmsg_writev(struct kio int level = default_message_loglevel; int facility = 1; /* LOG_USER */ size_t len = iov_length(iv, count); - ssize_t ret = len; + ssize_t ret; if (len > LOG_LINE_MAX) return -EINVAL; @@ -365,8 +365,12 @@ static ssize_t devkmsg_writev(struct kio line = buf; for (i = 0; i < count; i++) { - if (copy_from_user(line, iv[i].iov_base, iv[i].iov_len)) + if (copy_from_user(line, iv[i].iov_base, iv[i].iov_len)) { + ret = line - buf; + if (!ret) + ret = -EFAULT; goto out; + } line += iv[i].iov_len; } @@ -396,6 +400,7 @@ static ssize_t devkmsg_writev(struct kio line[len] = '\0'; printk_emit(facility, level, NULL, 0, "%s", line); + ret = 0; out: kfree(buf); return ret; _

1 0

[alsa-devel] [PATCH] ALSA: hda - add power states information in proc
by Wang Xingchao 06 Jun '12

06 Jun '12

add more power states information: - reset status - clock stop ok - power states error Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_proc.c | 6 ++++++ 2 files changed, 9 insertions(+) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 4fc3960..71864cdd 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -323,6 +323,9 @@ enum { #define AC_PWRST_D1 0x01 #define AC_PWRST_D2 0x02 #define AC_PWRST_D3 0x03 +#define AC_PWRST_ERROR (1<<8) +#define AC_PWRST_CLK_STOP_OK (1<<9) +#define AC_PWRST_SETTING_RESET (1<<10) /* Processing capabilies */ #define AC_PCAP_BENIGN (1<<0) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index e59e2f0..5511de1 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -459,6 +459,12 @@ static void print_power_state(struct snd_info_buffer *buffer, get_pwr_state(pwr & AC_PWRST_SETTING), get_pwr_state((pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT)); + if (pwr & AC_PWRST_ERROR) + snd_iprintf(buffer, " Power state error\n"); + if (pwr & AC_PWRST_CLK_STOP_OK) + snd_iprintf(buffer, " PS: clock stop ok\n"); + if (pwr & AC_PWRST_SETTING_RESET) + snd_iprintf(buffer, " Power state setting reset\n"); } static void print_unsol_cap(struct snd_info_buffer *buffer, -- 1.7.9.5

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[alsa-devel] Confusing about Playback/Capture, CODEC/CODEC links, and snd_soc_dapm_link_dai_widgets()
by Stephen Warren 06 Jun '12

06 Jun '12

Mark, Liam, recall the Tegra AHUB structure: > +-------------+ +-----+ +--------------+ +-----+ +------+ > | FIFO pair 0 |<->| CIF |<->| Cross-bar |<->| CIF |<->| I2S0 |<-> External IO > +-------------+ +-----+ | (the "AHUB") | +-----+ +------+ > . . . | | . . . > +-------------+ +-----+ | | +-----+ +------+ > | FIFO pair 3 |<->| CIF |<->| |<->| CIF |<->| I2S4 |<-> External IO > +-------------+ +-----+ | | +-----+ +------+ > | | > | | +-----+ +-------+ > | |<->| CIF |<->| SPDIF |<-> External IO > | | +-----+ +-------+ > | | > | | +-----+ +-------+ > | |<->| CIF |<->| DAM | > | | +-----+ +-------+ > +--------------+ (CIF is AHUB XBAR Client InterFace; basically the DAI or DAI link) I have created a driver for the 4 FIFOs on the left, which exposes 4 (CPU) DAIs. I have created a driver for the AHUB XBAR, which is an ASoC CODEC, and has a DAI for each of the CIFs. There's a CODEC/CODEC link between each FIFO DAI and the relevant AHUB CIF DAI. In order to connect the I2S devices to the AHUB XBAR, I had to make that a CODEC too. This exposes two DAIs; the CIF DAI to connect to the AHUB, and the DAP (Digital Audio Port) DAI to represent the I/O pins on the Tegra package that are connected to the external codec. I have an issue with the paths that snd_soc_dapm_link_dai_widgets() sets up inside both the AHUB XBAR and the I2S. I'll use the I2S to explain since it's internally much simpler. The two DAIs in I2S are roughly: > static const struct snd_soc_dai_driver tegra30_i2s_dais[] = { > { > .playback = { > .stream_name = "DAP Playback", > }, > .capture = { > .stream_name = "DAP Capture", > }, > }, > { > .playback = { > .stream_name = "CIF Playback", > }, > .capture = { > .stream_name = "CIF Capture", > }, > }, > }; For the CIF-side DAI, I instantiated widgets for the AIFs: > static const struct snd_soc_dapm_widget tegra30_i2s_widgets[] = { > SND_SOC_DAPM_AIF_IN("CIF RX", "CIF Playback", 0, SND_SOC_NOPM, 0, 0), > SND_SOC_DAPM_AIF_OUT("CIF TX", "CIF Capture", 0, SND_SOC_NOPM, 0, 0), This works fine; snd_soc_dapm_link_dai_widgets() create a route from the DAI widget "DAP Playback" that feeds into the AIF_IN widget "CIF RX". "works fine" means that the ASoC debugfs files in "asoc/$card/tegra30-i2s.1/dapm/CIF {Playback,RX}" show the expected connections. The DAPM routes inside the I2S are: > static const struct snd_soc_dapm_route tegra30_i2s_routes[] = { > { "DAP TX", NULL, "CIF RX" }, > { "CIF TX", NULL, "DAP RX" }, > }; This also works fine. Now, I instantiated two AIF widgets to interface with the DAP-side DAI: > SND_SOC_DAPM_AIF_IN("DAP RX", "DAP Capture", 0, SND_SOC_NOPM, 0, 0), > SND_SOC_DAPM_AIF_OUT("DAP TX", "DAP Playback", 0, SND_SOC_NOPM, 0, 0), However, this causes snd_soc_dapm_link_dai_widgets() to set up some unexpected paths; "DAP Playback" ends up feeding *into* "DAP TX", rather than feeding from/out of it... > # cat CIF\ Playback > CIF Playback: Off in 0 out 0 > stream CIF Playback inactive > out "static" "CIF RX" > out "static" "CIF RX" > # cat CIF\ RX > CIF RX: Off in 0 out 0 > stream CIF Playback inactive > in "static" "CIF Playback" > in "static" "CIF Playback" > out "static" "DAP TX" > # cat DAP\ TX > DAP TX: Off in 0 out 0 > stream DAP Playback inactive > in "static" "DAP Playback" <<<<<<<<<< > in "static" "DAP Playback" <<<<<<<<<< > in "static" "CIF RX" > # cat DAP\ Playback > DAP Playback: Off in 0 out 0 > stream DAP Playback inactive > out "static" "DAP TX" > out "static" "DAP TX" Now, I can fake this out by swapping the stream names on the DAP-side AIF widgets: > SND_SOC_DAPM_AIF_IN("DAP RX", "DAP Playback", 0, SND_SOC_NOPM, 0, 0), > SND_SOC_DAPM_AIF_OUT("DAP TX", "DAP Capture", 0, SND_SOC_NOPM, 0, 0), Which yields the expected paths in debugfs: > # cat CIF\ Playback > CIF Playback: Off in 0 out 0 > stream CIF Playback inactive > out "static" "CIF RX" > out "static" "CIF RX" > # cat CIF\ RX > CIF RX: Off in 0 out 0 > stream CIF Playback inactive > in "static" "CIF Playback" > in "static" "CIF Playback" > out "static" "DAP TX" > # cat DAP\ TX > DAP TX: Off in 0 out 0 > stream DAP Capture inactive > in "static" "CIF RX" > out "static" "DAP Capture" > out "static" "DAP Capture" > # cat DAP\ Capture > DAP Capture: Off in 0 out 0 > stream DAP Capture inactive > in "static" "DAP TX" > in "static" "DAP TX" But I'm not sure if that's quite correct. In the DAI definitions, is the .playback sub-structure always meant to represent: 1) Playback from the CPU's perspective, in which case the first set of DAP AIF definitions above would be correct or: 2) Is it more that Playback==RX_into_codec, Capture==TX_from_codec? This appears to be supported by the fact that the paths get set up correctly with the second set of AIF definitions above. But if (2) is correct, I wonder why soc_dapm_stream_event()'s first if statement appears to consider the playback_widget of both sides of the DAI to be coupled; wouldn't one side's playback widget be coupled to the other side's capture widget? As an aside, I'm not sure if it's conceptually correct to talk about playback or capture any more (beyond the initial CPU DAI) with arbitrary CODEC/CODEC links, since who knows what kind of routing/CPU->CPU loopbacks/external CODEC->CODEC loopbacks/... might exist in the CODECs? Either way though, something is still not working correctly even when the expected paths show up in debugfs. When I start playback on a PCM exposed by the DMA FIFO driver, I see the relevant AHUB XBAR's stream turned on, and a route exists to the relevant AIF_IN widget, but that widget is not considered to have any outward connections by is_connected_output_ep(), which I think is what is causing it and none of the rest of the path to turn on: (from asoc/$card/asoc/tegra30-ahub-xbar/dapm) > # cat APBIF0\ Playback > APBIF0 Playback: On in 1 out 1 > stream APBIF0 Playback active > out "static" "APBIF0 RX" > out "static" "APBIF0 RX" > # cat APBIF0\ RX > APBIF0 RX: Off in 2 out 0 <<<<< last value shouldn't be 0? > stream APBIF0 Playback inactive <<<<< still inactive? > in "static" "APBIF0 Playback" > in "static" "APBIF0 Playback" > out "APBIF0 RX" "I2S0 TX Mux" even though when I look at all the debugfs files, all the expected paths are there, at least within the AHUB XBAR and I2S drivers, and the external DAI links in the machine driver all probed as expected and bound all the components together. Probably related, the I2S and WM8903 (DAI) drivers aren't being called by the ASoC core to initialize themselves for playback either; pretty much all that happens is that DMA is started. Is the machine driver responsible for this? At this point, I'm not sure whether I have a gross mis-understanding of how this is supposed to work, or whether there are simply bits missing from the DAPM code to fully support CODEC/CODEC links, rather than CPU/CODEC links. Probably slightly related to all of this, but is the following code correct: > int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, > struct snd_soc_dapm_widget_list **list) ... > if (stream == SNDRV_PCM_STREAM_PLAYBACK) > paths = is_connected_output_ep(dai->playback_widget, list); > else > paths = is_connected_input_ep(dai->playback_widget, list); I would have expected this to use capture_widget on the final line, but I haven't thought about this in detail, just noticed the lack of symmetry by very brief inspection. Thanks for any help. Sorry for the long email.

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[alsa-devel] dapm_widget_power_read_file() symmetry issue?
by Stephen Warren 05 Jun '12

05 Jun '12

In dapm_widget_power_read_file(), there is first: > list_for_each_entry(p, &w->sources, list_sink) { > if (p->connected && !p->connected(w, p->sink)) > continue; then right after that: > list_for_each_entry(p, &w->sinks, list_source) { > if (p->connected && !p->connected(w, p->sink)) > continue; Should that last call to p->connected() be p->connected(p->source, w), since the function's parameters are (source, sink)?

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