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

[alsa-devel] [PATCH] ALSA: hda - resume codecs in parallel
by mengdong.lin＠intel.com 27 Nov '13

27 Nov '13

From: Mengdong Lin <mengdong.lin(a)intel.com> To reduce driver resume time, this patch resumes the codecs in parallel if there are multiple codecs on the bus. - The PM workqueue of bus is also used to parallel resuming multiple codecs. - The work item 'pm_work' is renamed to 'suspend_work' to parallel suspending codecs. - Add a work item 'resume_work' to parallel resuming codecs. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 60ff45a..1e04a20 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -104,7 +104,8 @@ static inline void hda_call_pm_notify(struct hda_bus *bus, bool power_up) if (bus->ops.pm_notify) bus->ops.pm_notify(bus, power_up); } -static void hda_pm_work(struct work_struct *work); +static void hda_suspend_work(struct work_struct *work); +static void hda_resume_work(struct work_struct *work); #else #define codec_in_pm(codec) 0 static inline void hda_keep_power_on(struct hda_codec *codec) {} @@ -1447,7 +1448,8 @@ int snd_hda_codec_new(struct hda_bus *bus, #ifdef CONFIG_PM spin_lock_init(&codec->power_lock); INIT_DELAYED_WORK(&codec->power_work, hda_power_work); - INIT_WORK(&codec->pm_work, hda_pm_work); + INIT_WORK(&codec->suspend_work, hda_suspend_work); + INIT_WORK(&codec->resume_work, hda_resume_work); /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. * the caller has to power down appropriatley after initialization * phase. @@ -5058,13 +5060,21 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power); -static void hda_pm_work(struct work_struct *work) +static void hda_suspend_work(struct work_struct *work) { struct hda_codec *codec = - container_of(work, struct hda_codec, pm_work); + container_of(work, struct hda_codec, suspend_work); hda_call_codec_suspend(codec, false); } + +static void hda_resume_work(struct work_struct *work) +{ + struct hda_codec *codec = + container_of(work, struct hda_codec, resume_work); + + hda_call_codec_resume(codec); +} #endif /* @@ -5633,7 +5643,7 @@ int snd_hda_suspend(struct hda_bus *bus) cancel_delayed_work_sync(&codec->jackpoll_work); if (hda_codec_is_power_on(codec)) { if (bus->num_codecs > 1) - queue_work(bus->pm_wq, &codec->pm_work); + queue_work(bus->pm_wq, &codec->suspend_work); else hda_call_codec_suspend(codec, false); } @@ -5659,9 +5669,15 @@ int snd_hda_resume(struct hda_bus *bus) printk("snd_hda_resume, pci device 0x%04x\n", bus->pci->device); list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (bus->num_codecs > 1) + queue_work(bus->pm_wq, &codec->resume_work); + else + hda_call_codec_resume(codec); } + if (bus->num_codecs > 1) + flush_workqueue(bus->pm_wq); + printk("snd_hda_resume ok, pci device 0x%04x\n", bus->pci->device); return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 82807ef..bedb436 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -920,7 +920,9 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; - struct work_struct pm_work; /* task to parallel multi-codec PM */ + /* tasks to parallel multi-codec suspend/resume */ + struct work_struct suspend_work; + struct work_struct resume_work; #endif /* filter the requested power state per nid */ -- 1.8.1.2

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[alsa-devel] [RFC PATCH] ALSA: hda - suspend codecs in parallel
by mengdong.lin＠intel.com 27 Nov '13

27 Nov '13

From: Mengdong Lin <mengdong.lin(a)intel.com> The time to suspend a single codec may be several hundreds of ms. When runtime power saving is not enabled, driver suspend time can be long especially there are more than one codec on the bus. To reduce driver suspend time, this patch creates a work queue for each codec and will suspend the codecs in parallel, if there are multiple codecs on bus. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index cb0c776..c9092a1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -97,6 +97,7 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset); #ifdef CONFIG_PM #define codec_in_pm(codec) ((codec)->in_pm) static void hda_power_work(struct work_struct *work); +static void hda_pm_work(struct work_struct *work); static void hda_keep_power_on(struct hda_codec *codec); #define hda_codec_is_power_on(codec) ((codec)->power_on) static inline void hda_call_pm_notify(struct hda_bus *bus, bool power_up) @@ -1355,6 +1356,8 @@ static void snd_hda_codec_free(struct hda_codec *codec) #ifdef CONFIG_PM if (!codec->pm_down_notified) /* cancel leftover refcounts */ hda_call_pm_notify(codec->bus, false); + if (codec->pm_wq) + destroy_workqueue(codec->pm_wq); #endif module_put(codec->owner); free_hda_cache(&codec->amp_cache); @@ -1434,6 +1437,13 @@ int snd_hda_codec_new(struct hda_bus *bus, */ hda_keep_power_on(codec); hda_call_pm_notify(bus, true); + + codec->pm_wq = create_workqueue("hda_codec_wq"); + if (!codec->pm_wq) { + snd_hda_codec_free(codec); + return -ENOMEM; + } + INIT_WORK(&codec->pm_work, hda_pm_work); #endif if (codec->bus->modelname) { @@ -1445,6 +1455,7 @@ int snd_hda_codec_new(struct hda_bus *bus, } list_add_tail(&codec->list, &bus->codec_list); + bus->num_codecs++; bus->caddr_tbl[codec_addr] = codec; codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT, @@ -5032,6 +5043,14 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power); + +static void hda_pm_work(struct work_struct *work) +{ + struct hda_codec *codec = + container_of(work, struct hda_codec, pm_work); + + hda_call_codec_suspend(codec, false); +} #endif /* @@ -5595,11 +5614,24 @@ EXPORT_SYMBOL_HDA(snd_hda_add_imux_item); int snd_hda_suspend(struct hda_bus *bus) { struct hda_codec *codec; + unsigned int mask = 0; list_for_each_entry(codec, &bus->codec_list, list) { cancel_delayed_work_sync(&codec->jackpoll_work); - if (hda_codec_is_power_on(codec)) - hda_call_codec_suspend(codec, false); + if (hda_codec_is_power_on(codec)) { + if (codec->epss || bus->num_codecs == 1) + hda_call_codec_suspend(codec, false); + else { + mask |= 1 << codec->addr; + queue_work(codec->pm_wq, &codec->pm_work); + } + } + } + + list_for_each_entry(codec, &bus->codec_list, list) { + if (mask & (1 << codec->addr)) + flush_work(&codec->pm_work); + } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 7aa9870..ffb1866 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -673,6 +673,7 @@ struct hda_bus { /* codec linked list */ struct list_head codec_list; + unsigned int num_codecs; /* link caddr -> codec */ struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; @@ -916,6 +917,8 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; + struct workqueue_struct *pm_wq; + struct work_struct pm_work; #endif /* filter the requested power state per nid */ -- 1.8.1.2

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[alsa-devel] [PATCH] ALSA: usb: use multiple packets per urb for Wireless USB inbound audio
by Thomas Pugliese 27 Nov '13

27 Nov '13

For Wireless USB audio devices, use multiple isoc packets per URB for inbound endpoints with a datainterval < 5. This allows the WUSB host controller to take advantage of bursting to service endpoints whose logical polling interval is less than the 4ms minimum polling interval limit in WUSB. Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> --- sound/usb/endpoint.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index b9ba0fc..83aabea2 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -636,8 +636,22 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, if (usb_pipein(ep->pipe) || snd_usb_endpoint_implicit_feedback_sink(ep)) { + urb_packs = packs_per_ms; + /* + * Wireless devices can poll at a max rate of once per 4ms. + * For dataintervals less than 5, increase the packet count to + * allow the host controller to use bursting to fill in the + * gaps. + */ + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) { + int interval = ep->datainterval; + while (interval < 5) { + urb_packs <<= 1; + ++interval; + } + } /* make capture URBs <= 1 ms and smaller than a period */ - urb_packs = min(max_packs_per_urb, packs_per_ms); + urb_packs = min(max_packs_per_urb, urb_packs); while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) urb_packs >>= 1; ep->nurbs = MAX_URBS; -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: hda - Enable mute/mic-mute LEDs for more Thinkpads with Conexant codec
by Hui Wang 27 Nov '13

27 Nov '13

Most Thinkpad Edge series laptops use conexant codec, so far although the codecs have different minor Vendor Id and minor Subsystem Id, but they all belong to the cxt_5066 family, this change can make the mute/mic-mute LEDs support more generic among cxt_5066 family. Cc: Alex Hung <alex.hung(a)canonical.com> Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_local.h | 22 ++++++++++++++++++++++ sound/pci/hda/patch_conexant.c | 3 +++ sound/pci/hda/patch_realtek.c | 20 -------------------- 3 files changed, 25 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index d398b64..1155958 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -777,4 +777,26 @@ void snd_hdmi_write_eld_info(struct hdmi_eld *eld, #define SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE 80 void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen); +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) +#include <acpi/acpi.h> + +static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, + void **rv) +{ + bool *found = context; + *found = true; + return AE_OK; +} + +static inline bool is_thinkpad(struct hda_codec *codec) +{ + bool found = false; + if (codec->subsystem_id >> 16 != 0x17aa) + return false; + if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) + return true; + found = false; + return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; +} +#endif #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c205bb1..8063ae9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3279,6 +3279,8 @@ static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + if (!is_thinkpad(codec)) + return; if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -3494,6 +3496,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), {} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5e42059..bedf35f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3673,29 +3673,9 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> -#include <acpi/acpi.h> static int (*led_set_func)(int, bool); -static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, - void **rv) -{ - bool *found = context; - *found = true; - return AE_OK; -} - -static bool is_thinkpad(struct hda_codec *codec) -{ - bool found = false; - if (codec->subsystem_id >> 16 != 0x17aa) - return false; - if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) - return true; - found = false; - return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; -} - static void update_tpacpi_mute_led(void *private_data, int enabled) { if (led_set_func) -- 1.8.1.2

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Re: [alsa-devel] Commit:ce6cfaf1 query for shared control
by Lars-Peter Clausen 27 Nov '13

27 Nov '13

On 11/25/2013 11:57 PM, noman pouigt wrote: > On Sun, Nov 24, 2013 at 11:45 PM, Lars-Peter Clausen <lars(a)metafoo.de> wrote: >> On 11/25/2013 08:12 AM, noman pouigt wrote: >>> Hello Lars, >>> >>> I was wondering if you can enlighten me >>> on below query. >>> >>> Commit:ce6cfaf1 >>> >>> From the commit text "The input and >>> output paths for each widgets are only >>> updated though during the respective run >>> for that widget". > Thanks for replying.Probably my query was > not clear.I wanted to understand the commit > text which i quoted above.And my understanding > of that commit text is below: The text just describes the behavior as it was before the commit, which was separate power-up/power-down sequence for each widget. >>> Does this commit text mean: when mux >>> or mixer_power_update is called then only >>> the widget is powered and it's peer- widget >>> source(path source) and widget sink(path sink)? >> >> Hi, >> >> What the commit does is to make sure that if a control is shared between >> multiple mixers/muxes to apply all the changes in the same update sequence. >> Instead of, as it was done before, to run a update sequence for each mixer/mux. > I understood this as you have clearly mentioned it with a nice > diagram. >> >>> >>> +------+ >>> A1 ------| | >>> | MUX1 |----- C1 >>> B1 ------| | >>> +------+ >>> | >>> control ---+ >>> | >>> +------+ >>> A2 ------| | >>> | MUX2 |----- C2 >>> B2 ------| | >>> +------+ >>> >>> Can I represent the above diagram as below: >>> >>> char *input_text_1[] = { >>> "A1", "B1" >>> }; >>> >>> struct soc_enum input_enum_1 = >>> SOC_ENUM_SINGLE(some_register, some_bit, >>> 2, input_text_1); >>> >>> char *input_text_2[] = { >>> "A2", "B2" >>> }; >>> struct soc_enum input_enum_2 = >>> SOC_ENUM_SINGLE(some_register, some_bit, >>> 2, input_text_2); >>> >>> >>> struct snd_kcontrol_new input_mux = >>> SOC_DAPM_ENUM("C1", input_enum_1); >>> struct snd_kcontrol_new input_mux = >>> SOC_DAPM_ENUM("C2", input_enum_2); >>> >>> dapm_route[] { >>> "A1", "control","C1" >>> "B1", "control","C1" >>> "A2", "control","C2" >>> "B2", "control","C2" >>> }; >> >> The control part needs to have the name of one of the enum items. So this >> would be { "A1", "A1", "C1" } and so on. > Yes makes sense as then only it will know > which widget to connect to as it has multiple > widgets. >> >> - Lars >> >> >> >> >>

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[alsa-devel] [PATCH 0/5] A few HD-audio things for 3.14
by Takashi Iwai 26 Nov '13

26 Nov '13

Hi, here are a few patches that are targeted for 3.14. So they are not urgently needed, mostly clean ups, but also includes some runtime PM changes as we're discussing now. The patches are found in sound-unstable git tree test/hda branch, too. Takashi

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Re: [alsa-devel] [alsa-cvslog] [ALSA GIT]ALSA utilities repository branch master updated. v1.0.27.2-14-gd2d9b6f
by Clemens Ladisch 26 Nov '13

26 Nov '13

GIT server wrote: > - Log ----------------------------------------------------------------- > commit d2d9b6fc8d20fca1e06fc261ac20fd6cfd8a8d1b > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Tue Nov 26 11:44:40 2013 +0100 > > alsactl: Add monitor command > mask = snd_ctl_event_elem_get_mask(event); > if (mask & SND_CTL_EVENT_MASK_VALUE) > printf(" VALUE"); > if (mask & SND_CTL_EVENT_MASK_INFO) > printf(" INFO"); > if (mask & SND_CTL_EVENT_MASK_ADD) > printf(" ADD"); > if (mask & SND_CTL_EVENT_MASK_TLV) > printf(" TLV"); > if (mask == SND_CTL_EVENT_MASK_REMOVE) > printf(" REMOVE"); The documentation of SND_CTL_EVENT_MASK_REMOVE says: | Warning: test this first and if set don't test the other masks Regards, Clemens

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[alsa-devel] HD-Audio: Is it okay to delay resuming codecs when resuming from S3?
by Lin, Mengdong 26 Nov '13

26 Nov '13

Hi Takashi, Is it okay to delay resuming codecs when resuming from S3? i.e. skip snd_hda_resume() in azx_resume()? This would reduce driver resume time from S3. And the codecs will be resumed when being used for the 1st time after S3. Resuming a codec may cost more than 100ms. So even if we parallel resuming codecs in snd_hda_resume(), driver resume time on some platforms is still a bit long. Regards Mengdong

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[alsa-devel] [PATCH] ALSA: hda - Drop bus->avoid_link_reset flag
by Takashi Iwai 26 Nov '13

26 Nov '13

Use bus->power_keep_link_on instead. The controller shouldn't go to D3 when the link isn't reset, so essentially avoiding the link reset means avoiding the runtime PM. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.h | 1 - sound/pci/hda/hda_intel.c | 3 +-- sound/pci/hda/patch_sigmatel.c | 3 ++- 3 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 77db69480c19..7aa9870040c1 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -698,7 +698,6 @@ struct hda_bus { unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ - unsigned int avoid_link_reset:1; /* don't reset link at runtime PM */ int primary_dig_out_type; /* primary digital out PCM type */ }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7a09404579a7..c6d230193da6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2994,8 +2994,7 @@ static int azx_runtime_suspend(struct device *dev) STATESTS_INT_MASK); azx_stop_chip(chip); - if (!chip->bus->avoid_link_reset) - azx_enter_link_reset(chip); + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d2cc0041d9d3..088a5afbd1b9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2094,7 +2094,8 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ - codec->bus->avoid_link_reset = 1; + /* resetting controller clears GPIO, so we need to keep on */ + codec->bus->power_keep_link_on = 1; } } -- 1.8.4.3

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[alsa-devel] [PATCH v2 0/9] ASoC: fsl-ssi: offline/online configuration and cleanups
by Markus Pargmann 26 Nov '13

26 Nov '13

Hi, This patch series fixes some fsl-ssi code that does not act exactly as it is described in the reference manuals. The reference manuals before imx50 do mention that some register bits, including TDMAE/RDMAE, should not be changed while the SSI unit is enabled (SSIEN). At the same time the SDMA unit has a undefined request handling if there are DMA requests before the SDMA engine and its channel is configured. The reference manual states that SSI DMA requests are sent as soon as the FIFO and the DMAE bits are enabled. It does not mention a dependency to SCR's TE/RE bit. To avoid sending DMA requests before a channel is configured, we have to seperate the fsl-ssi behavior into offline and online configuration. SoCs before imx50 have to use offline configuration while later SoC versions can use online reconfiguration. This series adds support for online configuration and cleans up the configuration register code especially in fsl_ssi_trigger. At the end we have a seperation between enable/disable logic and the actual configuration register values used to enable/disable TX/RX. I tested this series on mx53. The series is based on Mark Brown's topic/fsl branch. Regards, Markus Changes in v2: - Enable fsl-ssi interrupts independent of CONFIG_DEBUG_FS. - Move irq mapping fix into seperate patch (was in "Move sysfs stats to debugfs") - Rebased on v3.13-rc1 topic/fsl branch. Changes in v1: - Not removing imx21-ssi from the compatible array in the DTS patch. - Dropped "ASoC: fsl-ssi: Drop AC97 debug register usage" for the moment as I don't have time to setup AC97 hardware for some tests Markus Pargmann (9): ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Fix interrupt mapping and release ASoC: fsl-ssi: Add imx50-ssi and of_device_id matching ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ARM: DTS: imx5* imx6*, use imx50-ssi ASoC: fsl-ssi: Drop AC97 debug register usage arch/arm/boot/dts/imx51.dtsi | 10 +- arch/arm/boot/dts/imx53.dtsi | 10 +- arch/arm/boot/dts/imx6qdl.dtsi | 12 +- arch/arm/boot/dts/imx6sl.dtsi | 12 +- sound/soc/fsl/fsl_ssi.c | 521 +++++++++++++++++++++++++++-------------- 5 files changed, 371 insertions(+), 194 deletions(-) -- 1.8.4.2

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