August 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Trouble about Roland UA-11 UA-33 UA-55 and snd-usb-audio module
by FirebUntU 07 Mar '16

07 Mar '16

Hi guys, I write you about support of snd-usb-audio module and a relatively-new gamma of devices from Roland/Edirol: DUO CAPTURE, TRI CAPTURE and QUAD CAPTURE respectively: UA-11, UA-33, UA-55 These aren't not listed in the Alsa db. I have searched a solution for a week and I haven't found a trick to make UA-11 to work under Alsa 1.25.0 then I downloaded the driver for try to patch it and I found these devices is not listed in the tricks table In despair I have played with the switch button called "EXT * **" in the bottom of the unit of UA-11 and magically the device ID is transformed from * 0582:012b to ** 0582:0137 Image: https://dl.dropbox.com/u/32759898/output.jpg (Rear of the UA-11 unit, from the manual) *Before* (Recognized from udev - Unknown from alsa): http://www.alsa-project.org/db/?f=502ee68cf31882ba9ef23525fd66452c34d329e1 *After* (Recognized from udev - Recognized from alsa): http://www.alsa-project.org/db/?f=b2ae3e4331fb3fe613ba71e7aba1ad2e0e6dcd29 In the tricks table header of ALSA the 0582:0137 stay for PM-1000 in /TODO/ statement. I understand the EXT switch, undiscovered in the manual of the DUO CAPTURE is - as the fact - the Advanced Mode. Now the peripherical running both in playback and recording in 16 and 24 bit and without any mixer options in alsamixer. Can you update future releases with this gamma of devices? Awaiting for your positive response :) Thanks. Best regards, Leonardo

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[alsa-devel] writing an alsa driver
by Michele Curti 04 Mar '16

04 Mar '16

Hi all, I'm going to *try* to "write" a driver for the sound card on my new laptop, an asus x205ta. To be fair I have almost no experience but I'm hoping that all the code is already present and it only needs some adaptations. The cpu is a intel bay trail z3735f, the sound card is a realtek rt5648 that has the same product Id of the rt5645 (0x6308). I see in kernel sources: - the codec driver sound/soc/codecs/rt5645.c - a baytrail board that uses the rt5640 sound/soc/intel/boards/byt-rt5640.c - a braswell board that uses the rt5645 sound/soc/intel/boards/cht_bsw_rt5645.c What do you think? Is it feasible, given a good amount of free-time? :p Thanks, Michele

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[alsa-devel] Usb audio - unsuported sampling rates
by Gordon Dilan 12 Feb '16

12 Feb '16

Hello, I'm using an usb mic which(according to it's producer should support up to 48kHz sampling rate). The problem is ALSA allows me to use only 16 kHz for this device. I do not want to resample the data, I want to sample it at 48kHz in order to be able to record some high pitch sounds 18-20kHz. I would want to try to do any modifications necessary in order to try support 48kHz, but I'm not sure where to start. Some information: cat /proc/asound/modules ... 1 snd_usb_audio arecord --list-devices **** List of CAPTURE Hardware Devices **** .... card 1: CameraB409241 [USB Camera-B4.09.24.1], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 when I try to use: arecord -v -D hw:1,0 -f S16_LE -c 4 -r 48000 gigi.wav I get this: Recording WAVE 'gigi.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channels 4 Warning: rate is not accurate (requested = 48000Hz, got = 16000Hz) please, try the plug plugin Hardware PCM card 1 'USB Camera-B4.09.24.1' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 4 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 8000 period_size : 2000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 2000 period_event : 0 start_threshold : 1 stop_threshold : 8000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 appl_ptr : 0 hw_ptr : 0 Any sampling rate I would try higher or lower than 16kHz, I will always get 16000 kHz. Some more information: lsusb Bus 001 Device 007: ID 1415:2000 Nam Tai E&E Products Ltd. or OmniVision Technologies, Inc. Sony Playstation Eye lsusb -t /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/2p, 480M |__ Port 1: Dev 2, If 0, Class=hub, Driver=hub/6p, 480M |__ Port 2: Dev 7, If 0, Class=vend., Driver=ov534, 480M |__ Port 2: Dev 7, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 7, If 2, Class=audio, Driver=snd-usb-audio, 480M At the moment I'm looking inside ALSA's source code to try to see from where this limitation of 16000kHz for my device appears, but I have to admit that I'm kind of lost. Any pointers on what direction should I go to try make my desired rate supported would be greatly appreciated! Thank you and best regards, Gordon

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[alsa-devel] [PATCH v2 1/3] ARM: shmobile: armadillo800eva: Properly specify HDMI audio link format
by Lars-Peter Clausen 29 Jan '16

29 Jan '16

The DAI link format should be specified for the whole link rather than just one component on the link. So move the format specification for the HDMI audio link from the CPU component to the link itself. Since the sh-mobile-hdmi DAI driver doesn't implement the set_fmt() callback in this case there is no functional difference between only specifying the the format for the CPU side or for the whole link, but the later it will allow us to remove support for just specifying the format for one component. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes since v1. --- arch/arm/mach-shmobile/board-armadillo800eva.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 6d949f1..75de26c 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -1040,9 +1040,9 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", + .daifmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "sh_mobile_hdmi-hifi", -- 1.8.0

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[alsa-devel] [PATCH 0/4] ASoC: fix saif record unwork issue
by Dong Aisheng 16 Jan '16

16 Jan '16

This patch series mainly fixes the saif record unwork issue. The root cause consists of: 1) clk_prepare and clk_enable un-balanced 2) missed setting a proper initial clock rate for saif in dt 3) sgtl5000 dapm widget kcontrol does not work So can not test record on mx28evk. 4) missed setting clkmux mode for saif in dt With this patch series it can fix sgtl5000 widget kcontrol unwork and saif record unwork issue. BTW, just let people konw, i still met a few overruns during record, that's another issue and will be fixed in separte patches later. Patch based on sound-2.6/for-3.6. Dong Aisheng (4): ASoC: mxs-saif: fix clock prepare and enable unbalance issue ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work ASoC: sgtl5000: remove unneeded snd_soc_dapm_new_widgets in probe ARM: mx28evk: add missed clkmux select for saif arch/arm/mach-mxs/mach-mxs.c | 3 +++ sound/soc/codecs/sgtl5000.c | 2 -- sound/soc/mxs/mxs-saif.c | 24 ++++++++++++++++++++++++ 3 files changed, 27 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH 00/12] dw-hdmi development
by Russell King - ARM Linux 07 Jan '16

07 Jan '16

This sub-series is a mixture of development: * Removing the incorrect pixel repetition configuration code * Preventing pixel-doubled modes from being used * Adding interlaced video support * Implementing the sink_is_hdmi/sink_has_audio flags I suggested a few months ago * Only enabling audio support if the sink indicates it has audio * Avoiding double-enabling the HDMI interface * Fixing the mis-leading name of "dw_hdmi_phy_enable_power" * Adding connector mode forcing (important if your monitor bounces RXSENSE and HPD signals while in low-power mode.) * Improving the HDMI enable/disabling on sink status For review (and testing if people feel like it). Acks/tested-bys etc welcome. It applies on top of my drm-dwhdmi-devel branch, which is waiting for David Airlie to pull (see pull request on dri-devel, 15th July.) drivers/gpu/drm/bridge/dw_hdmi.c | 275 ++++++++++++++++++++++++++++++--------- drivers/gpu/ipu-v3/ipu-dc.c | 18 ++- drivers/gpu/ipu-v3/ipu-di.c | 129 +++++++++--------- 3 files changed, 291 insertions(+), 131 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

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[alsa-devel] [PATCH 1/2] ALSA: usb-audio: Add a volume scale quirk for AudioQuest DragonFly
by Anssi Hannula 14 Dec '15

14 Dec '15

AudioQuest DragonFly DAC reports a volume control range of 0..50 (0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which is obviously incorrect and causes software using the dB information in e.g. volume sliders to have a massive volume difference in 100..102% range. The actual volume mapping seems to be neither linear volume nor linear dB scale, but instead quite close to the cubic mapping e.g. alsamixer uses, with a range of -53...0 dB. Add a quirk for DragonFly to use a custom dB mapping, based on my measurements, using a 10-item range TLV (so it still fits in alsa-lib MAX_TLV_RANGE_SIZE). Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the range is 0..50, so if this gets fixed/changed in later HW revisions it will no longer be applied. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> --- sound/usb/mixer_quirks.c | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 337c317ead6f..39d7f34e44e6 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -37,6 +37,7 @@ #include <sound/control.h> #include <sound/hwdep.h> #include <sound/info.h> +#include <sound/tlv.h> #include "usbaudio.h" #include "mixer.h" @@ -1733,6 +1734,38 @@ static int snd_microii_controls_create(struct usb_mixer_interface *mixer) return 0; } +static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer) +{ + struct usb_mixer_elem_list *list; + struct usb_mixer_elem_info *cval; + static const int unit_id = 7; + + /* approximation using 10 ranges based on output measurement on hw v1.2 */ + static const DECLARE_TLV_DB_RANGE(scale, + 0, 1, TLV_DB_MINMAX_ITEM(-5300, -4970), + 2, 5, TLV_DB_MINMAX_ITEM(-4710, -4160), + 6, 7, TLV_DB_MINMAX_ITEM(-3884, -3710), + 8, 14, TLV_DB_MINMAX_ITEM(-3443, -2560), + 15, 16, TLV_DB_MINMAX_ITEM(-2475, -2324), + 17, 19, TLV_DB_MINMAX_ITEM(-2228, -2031), + 20, 26, TLV_DB_MINMAX_ITEM(-1910, -1393), + 27, 31, TLV_DB_MINMAX_ITEM(-1322, -1032), + 32, 40, TLV_DB_MINMAX_ITEM(-968, -490), + 41, 50, TLV_DB_MINMAX_ITEM(-441, 0), + ); + + for (list = mixer->id_elems[unit_id]; list; list = list->next_id_elem) { + cval = (struct usb_mixer_elem_info *)list; + if (cval->control == UAC_FU_VOLUME && + cval->min == 0 && cval->max == 50) { + usb_audio_info(mixer->chip, "applying DragonFly dB scale quirk\n"); + list->kctl->tlv.p = scale; + list->kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + list->kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } + } +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1810,6 +1843,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x800c): /* Focusrite Scarlett 18i20 */ err = snd_scarlett_controls_create(mixer); break; + + case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ + snd_dragonfly_quirk_db_scale(mixer); + break; } return err; -- 2.3.8

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[alsa-devel] [PATCH 0/4 v5] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 27 Nov '15

27 Nov '15

Hopefully last version? :-) * Added commit text about duplicate events (patch 4/4) * Added locks in bind/unbind on i915 side (patch 2/4) * Fixed docbook comments in i915 struct (patch 1/4) * Removed port_mst_streaming parameter (patch 1/4) * Added Reviewed-by - 1 & 2 by Jani, 3 & 4 by Takashi. Hopefully this was correct, otherwise feel free to adjust yourself before committing. Both Jani and Takashi seem okay with this going into 4.3. Nobody has said which tree you prefer to take it through, so how about Takashi merging it? David Henningsson (4): drm/i915: Add audio pin sense / ELD callback drm/i915: Call audio pin/ELD notify function ALSA: hda - allow codecs to access the i915 pin/ELD callback ALSA: hda - Wake the codec up on pin/ELD notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 27 ++++++++++++++++++++++++--- include/drm/i915_component.h | 16 ++++++++++++++++ include/sound/hda_i915.h | 7 +++++++ sound/hda/hdac_i915.c | 10 ++++++++++ sound/pci/hda/patch_hdmi.c | 22 +++++++++++++++++++++- 6 files changed, 79 insertions(+), 4 deletions(-) -- 1.9.1

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[alsa-devel] [RFC PATCH 02/10] ASoC: Define soc_init_dai_link() to wrap link intialization.
by mengdong.lin＠intel.com 25 Nov '15

25 Nov '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Define soc_init_dai_link() to wrap link initialization, to reuse it later by snd_soc_instantiate_card() when adding new DAI links from topology in component probing phase. Move static func snd_soc_init_multicodec(), so that it can be reused by soc_init_dai_link(). This saves adding a function declaration for snd_soc_init_multicodec(). Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fb78674..e9e8956 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1141,6 +1141,100 @@ static void soc_remove_dai_links(struct snd_soc_card *card) } } +static int snd_soc_init_multicodec(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + /* Legacy codec/codec_dai link is a single entry in multicodec */ + if (dai_link->codec_name || dai_link->codec_of_node || + dai_link->codec_dai_name) { + dai_link->num_codecs = 1; + + dai_link->codecs = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!dai_link->codecs) + return -ENOMEM; + + dai_link->codecs[0].name = dai_link->codec_name; + dai_link->codecs[0].of_node = dai_link->codec_of_node; + dai_link->codecs[0].dai_name = dai_link->codec_dai_name; + } + + if (!dai_link->codecs) { + dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); + return -EINVAL; + } + + return 0; +} + +static int soc_init_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link) +{ + int i, ret; + + ret = snd_soc_init_multicodec(card, link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init multicodec\n"); + return ret; + } + + for (i = 0; i < link->num_codecs; i++) { + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codecs[i].name == + !!link->codecs[i].of_node) { + dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* Codec DAI name must be specified */ + if (!link->codecs[i].dai_name) { + dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", + link->name); + return -EINVAL; + } + } + + /* + * Platform may be specified by either name or OF node, but + * can be left unspecified, and a dummy platform will be used. + */ + if (link->platform_name && link->platform_of_node) { + dev_err(card->dev, + "ASoC: Both platform name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + + /* + * CPU device may be specified by either name or OF node, but + * can be left unspecified, and will be matched based on DAI + * name alone.. + */ + if (link->cpu_name && link->cpu_of_node) { + dev_err(card->dev, + "ASoC: Neither/both cpu name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* + * At least one of CPU DAI name or CPU device name/node must be + * specified + */ + if (!link->cpu_dai_name && + !(link->cpu_name || link->cpu_of_node)) { + dev_err(card->dev, + "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + + return 0; +} + static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_component *component) { @@ -2363,33 +2457,6 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); -static int snd_soc_init_multicodec(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link) -{ - /* Legacy codec/codec_dai link is a single entry in multicodec */ - if (dai_link->codec_name || dai_link->codec_of_node || - dai_link->codec_dai_name) { - dai_link->num_codecs = 1; - - dai_link->codecs = devm_kzalloc(card->dev, - sizeof(struct snd_soc_dai_link_component), - GFP_KERNEL); - if (!dai_link->codecs) - return -ENOMEM; - - dai_link->codecs[0].name = dai_link->codec_name; - dai_link->codecs[0].of_node = dai_link->codec_of_node; - dai_link->codecs[0].dai_name = dai_link->codec_dai_name; - } - - if (!dai_link->codecs) { - dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); - return -EINVAL; - } - - return 0; -} - /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2398,7 +2465,7 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card, */ int snd_soc_register_card(struct snd_soc_card *card) { - int i, j, ret; + int i, ret; struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) @@ -2407,63 +2474,11 @@ int snd_soc_register_card(struct snd_soc_card *card) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai_link *link = &card->dai_link[i]; - ret = snd_soc_init_multicodec(card, link); + ret = soc_init_dai_link(card, link); if (ret) { - dev_err(card->dev, "ASoC: failed to init multicodec\n"); - return ret; - } - - for (j = 0; j < link->num_codecs; j++) { - /* - * Codec must be specified by 1 of name or OF node, - * not both or neither. - */ - if (!!link->codecs[j].name == - !!link->codecs[j].of_node) { - dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", - link->name); - return -EINVAL; - } - /* Codec DAI name must be specified */ - if (!link->codecs[j].dai_name) { - dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", - link->name); - return -EINVAL; - } - } - - /* - * Platform may be specified by either name or OF node, but - * can be left unspecified, and a dummy platform will be used. - */ - if (link->platform_name && link->platform_of_node) { - dev_err(card->dev, - "ASoC: Both platform name/of_node are set for %s\n", + dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); - return -EINVAL; - } - - /* - * CPU device may be specified by either name or OF node, but - * can be left unspecified, and will be matched based on DAI - * name alone.. - */ - if (link->cpu_name && link->cpu_of_node) { - dev_err(card->dev, - "ASoC: Neither/both cpu name/of_node are set for %s\n", - link->name); - return -EINVAL; - } - /* - * At least one of CPU DAI name or CPU device name/node must be - * specified - */ - if (!link->cpu_dai_name && - !(link->cpu_name || link->cpu_of_node)) { - dev_err(card->dev, - "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", - link->name); - return -EINVAL; + return ret; } } -- 1.9.1

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[alsa-devel] [PATCH v1 1/1] sst: replace custom implementation of readq / writeq
by Andy Shevchenko 24 Nov '15

24 Nov '15

The readq() and writeq() helpers are available in the asm-generic/io-64-nonatomic-hi-lo.h and asm-generic/io-64-nonatomic-lo-hi.h headers. Replace custom implementation by the generic helpers. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/atom/sst/sst_pvt.c | 16 ++++++---------- sound/soc/intel/common/sst-dsp.c | 9 ++++----- 2 files changed, 10 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index adb32fe..6cc222d 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -31,7 +31,10 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/compress_driver.h> + +#include <asm-generic/io-64-nonatomic-lo-hi.h> #include <asm/platform_sst_audio.h> + #include "../sst-mfld-platform.h" #include "sst.h" #include "../../common/sst-dsp.h" @@ -49,25 +52,18 @@ u32 sst_shim_read(void __iomem *addr, int offset) u64 sst_reg_read64(void __iomem *addr, int offset) { - u64 val = 0; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - - return val; + return readq(addr + offset); } int sst_shim_write64(void __iomem *addr, int offset, u64 value) { - memcpy_toio(addr + offset, &value, sizeof(value)); + writeq(value, addr + offset); return 0; } u64 sst_shim_read64(void __iomem *addr, int offset) { - u64 val = 0; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - return val; + return readq(addr + offset); } void sst_set_fw_state_locked( diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index a627236..5274482 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -22,6 +22,8 @@ #include <linux/io.h> #include <linux/delay.h> +#include <asm-generic/io-64-nonatomic-lo-hi.h> + #include "sst-dsp.h" #include "sst-dsp-priv.h" @@ -43,16 +45,13 @@ EXPORT_SYMBOL_GPL(sst_shim32_read); void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value) { - memcpy_toio(addr + offset, &value, sizeof(value)); + writeq(value, addr + offset); } EXPORT_SYMBOL_GPL(sst_shim32_write64); u64 sst_shim32_read64(void __iomem *addr, u32 offset) { - u64 val; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - return val; + return readq(addr + offset); } EXPORT_SYMBOL_GPL(sst_shim32_read64); -- 2.5.0

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