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

[alsa-devel] [PATCH 1/1] ALSA: CA0132 - Ensure SCP packet is little-endian format
by Ian Minett 08 Oct '12

08 Oct '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi Takashi, Great - good to hear the previous DSP loader patches are now living in the hda-ca0132-dsp branch. This patch is based on that branch, and addresses the SCP packet endianness. >> The firmware does need to be reloaded when the chip loses power, so it >> doesn't survive S3. >OK, we need suspend/resume hooks, too. Do we need to do anything specific to handle this? The firmware is already restored upon S3 by the current caching code. Thanks, - Ian --- The DSP firmware expects the SCP data to be in little-endian format regardless of the host architecture. Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2fd3121..4d7cda4 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -28,6 +28,7 @@ #include <linux/mutex.h> #include <linux/module.h> #include <linux/firmware.h> +#include <asm/unaligned.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -701,9 +702,13 @@ static int dspio_send(struct hda_codec *codec, unsigned int reg, unsigned int res; int retry = 50; + /* Handling for Big Endian machine */ + unsigned int val; + put_unaligned_le32(data, &val); + /* send bits of data specified by reg to dsp */ do { - res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data); + res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, val); if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY)) return res; } while (--retry); -- 1.7.4.1

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[alsa-devel] [PATCH 0/10] ALSA: Add DSP firmware loader (v3)
by Ian Minett 08 Oct '12

08 Oct '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi, thanks for the response. I've resubmitted the patch series with updates to the ones that needed altering based on the feedback. The MODULE_FIRMWARE line has been shifted to the firmware loading patch, DSP download is now guarded by the config switch and code changes that were bundled in with the previous comments patch have moved to a new patch. Also, hda_stream_format has been removed. Regarding the SCP packet endianness, the DSP image requires data to be sent as little-endian, so we may need to swizzle data being sent to the DSP before the transfer. Is there a recommended or standard way of detecting and handling the case of running on big-endian architectures? >Does the firmware on chip survive after S3? The firmware does need to be reloaded when the chip loses power, so it doesn't survive S3. Thanks very much, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- 1: - memalloc.h - pcm.h - pcm_memory.c - sgbuf.c Include Takashi's patch: Make snd_sgbuf_get_{ptr|addr}() available for non-SG cases. Passing struct snd_dma_buffer pointer instead, so that they work no matter whether real SG buffer is used or not. 2: - hda_intel.c - hda_codec.h Include Takashi's code: Pass DMA buffer pointers in calls to setup_bdle(). Add DSP loader callback routines to controller. Add new DSP loader switch to Kconfig to enable DSP firmware loading. 3: - patch_ca0132.c - ca0132_regs.h Add DSP register definitions header file 4 (updated): - patch_ca0132.c Add DSP firmware enums and defs to CA0132 codec MODULE_FIRMWARE not declared in this patch, moved into patch #5 5 (updated): - patch_ca0132.c Add calls to new DSP loader system to transfer firmware binary to the hardware. Add chip read/write routines, DSP I/O, SCP packet format helper functions and transfer DMA management. Add MODULE_FIRMWARE line for DSP firmware binary Protect DSP download using CONFIG_SND_HDA_DSP_LOADER switch 6: - patch_ca0132.c Add DSP firmware caching to CA0132 codec 7 (updated): - patch_ca0132.c Add comments and descriptions to functions. Previous code changes moved to patch #9 8: - hda_codec.h Change return value for load_dsp_prepare to -ENOSYS in case where DSP loader routines are not available. 9 (new): - patch_ca0132.c Merge chipio write address functions and update timeout mechanism in dsp_write_wait(). 10 (new): - patch_ca0132.c Remove unnecessary struct hda_stream_format and references to it.

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[alsa-devel] [PATCH] ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
by Felix Kaechele 08 Oct '12

08 Oct '12

The Lenovo IdeaPad U310 has an internal mic where the right channel is phase inverted. Signed-off-by: Felix Kaechele <felix(a)fetzig.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e22a8f..784017e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4462,6 +4462,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), 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), {} }; -- 1.7.11.4

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[alsa-devel] [PATCH 1/2] ALSA: hda - make a generic unsol event handler
by David Henningsson 08 Oct '12

08 Oct '12

Moving towards less duplication of code between codecs - this patch takes some of the common code of unsol event handling and makes it generic. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 32 ++++++++++++++++++++++++++++++-- sound/pci/hda/hda_jack.h | 9 +++++++++ 2 files changed, 39 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index c9333c9..5c690cb 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -192,8 +192,9 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect); /** * snd_hda_jack_detect_enable - enable the jack-detection */ -int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, - unsigned char action) +int snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, + unsigned char action, + hda_jack_callback cb) { struct hda_jack_tbl *jack = snd_hda_jack_tbl_new(codec, nid); if (!jack) @@ -203,10 +204,19 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, jack->jack_detect = 1; if (action) jack->action = action; + if (cb) + jack->callback = cb; return snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | jack->tag); } +EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable_callback); + +int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, + unsigned char action) +{ + return snd_hda_jack_detect_enable_callback(codec, nid, action, NULL); +} EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); /** @@ -411,3 +421,21 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctls); + +void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct hda_jack_tbl *event; + int tag = (res >> AC_UNSOL_RES_TAG_SHIFT) & 0x7f; + + event = snd_hda_jack_tbl_get_from_tag(codec, tag); + if (!event) + return; + event->jack_dirty = 1; + + if (event->callback) + event->callback(codec, event); + + snd_hda_jack_report_sync(codec); +} +EXPORT_SYMBOL_HDA(snd_hda_jack_unsol_event); + diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index a9803da..af8dd47 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -13,12 +13,16 @@ #define __SOUND_HDA_JACK_H struct auto_pin_cfg; +struct hda_jack_tbl; + +typedef void (*hda_jack_callback) (struct hda_codec *, struct hda_jack_tbl *); struct hda_jack_tbl { hda_nid_t nid; unsigned char action; /* event action (0 = none) */ unsigned char tag; /* unsol event tag */ unsigned int private_data; /* arbitrary data */ + hda_jack_callback callback; /* jack-detection stuff */ unsigned int pin_sense; /* cached pin-sense value */ unsigned int jack_detect:1; /* capable of jack-detection? */ @@ -61,6 +65,10 @@ void snd_hda_jack_set_dirty_all(struct hda_codec *codec); int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, unsigned char action); +int snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, + unsigned char action, + hda_jack_callback cb); + u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); @@ -74,5 +82,6 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, void snd_hda_jack_report_sync(struct hda_codec *codec); +void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res); #endif /* __SOUND_HDA_JACK_H */ -- 1.7.9.5

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Re: [alsa-devel] Internal Microphone in Lenovo P580 not working with alsa 1.0.25
by Raymond Yau 08 Oct '12

08 Oct '12

2012-10-5 下午3:39 於 "mefis" <mefistof(a)gmx.com> 寫道： > > On 10/05/2012 02:09 AM, Raymond Yau wrote: >> >> 2012-10-5 下午2:20 於 "mefis" <mefistof(a)gmx.com> 寫道： >>> >>> Hi, >>> >>> Mic not working in Lenovo P580 with alsa 1.0.25 >>> >>> Alsa Information script output: >>> >>> http://www.alsa-project.org/db/?f=001523ed61bc76fb5b2ad0d3c354c043f89e5a33 >>> >>> Hope it helps. >>> >>> >>> >>> !!################################ >>> !!ALSA Information Script v 0.4.61 >>> !!################################ >>> >>> !!Script ran on: Fri Oct 5 05:56:38 UTC 2012 >>> >>> >>> !!Linux Distribution >>> !!------------------ >>> >>> Ubuntu 12.04.1 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu >> >> 12.04.1 LTS" >>> >>> >>> !!DMI Information >>> !!--------------- >>> >>> Manufacturer: LENOVO >>> Product Name: Lenovo IdeaPad P580 >>> Product Version: Lenovo IdeaPad P580 >>> Firmware Version: 5ECN33WW(V2.03) >>> >>> >>> !!Kernel Information >>> !!------------------ >>> >>> Kernel release: 3.2.0-31-generic >>> Operating System: GNU/Linux >>> Architecture: x86_64 >>> Processor: x86_64 >>> SMP Enabled: Yes >>> >>> >>> !!ALSA Version >>> !!------------ >>> >>> Driver version: 1.0.25 >>> Library version: 1.0.16 >>> Utilities version: 1.0.25 >> >> is this an upgrade since alsa-lib is quite old ? > > sorry, this was an old log. The newest is here: http://www.alsa-project.org/db/?f=08ae62e61ccc89d92a6154d4e5907b4f676c164a > >> >> it is strange that unsolicied event enabled at fixed mic >> instead of mic jack ? >> > > !!################################ > !!ALSA Information Script v 0.4.61 > !!################################ > > !!Script ran on: Fri Oct 5 07:34:06 UTC 2012 > > > > !!Linux Distribution > !!------------------ > > Ubuntu 12.04.1 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 12.04.1 LTS" > > > !!DMI Information > !!--------------- > > Manufacturer: LENOVO > Product Name: Lenovo IdeaPad P580 > Product Version: Lenovo IdeaPad P580 > Firmware Version: 5ECN33WW(V2.03) > > > !!Kernel Information > !!------------------ > > Kernel release: 3.2.0-31-generic > Operating System: GNU/Linux > Architecture: x86_64 > Processor: x86_64 > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.25 > Library version: 1.0.25 > > Utilities version: 1.0.25 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > > > !!Sound Servers on this system > !!---------------------------- > > Pulseaudio: > Installed - Yes (/usr/bin/pulseaudio) > Running - Yes > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [PCH ]: HDA-Intel - HDA Intel PCH > HDA Intel PCH at 0xe0610000 irq 46 > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation Panther Point High Definition Audio Controller (rev 04) > > > !!Advanced information - PCI Vendor/Device/Subsystem ID's > !!------------------------------------------------------- > > 00:1b.0 0403: 8086:1e20 (rev 04) > Subsystem: 17aa:3977 > > > !!Modprobe options (Sound related) > !!-------------------------------- > > snd-atiixp-modem: index=-2 > snd-intel8x0m: index=-2 > snd-via82xx-modem: index=-2 > snd-usb-audio: index=-2 > snd-usb-caiaq: index=-2 > snd-usb-ua101: index=-2 > snd-usb-us122l: index=-2 > snd-usb-usx2y: index=-2 > snd-cmipci: mpu_port=0x330 fm_port=0x388 > snd-pcsp: index=-2 > snd-usb-audio: index=-2 > snd-hda-intel: model=basic try model=auto http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=blob_plain;f=Do… Conexant 5066 ============= laptopBasic Laptop config (default) hp-laptopHP laptops, e g G60 asusAsus K52JU, Lenovo G560 dell-laptopDell laptops dell-vostroDell Vostro olpc-xo-1_5OLPC XO 1.5 ideapad Lenovo IdeaPad U150 thinkpadLenovo Thinkpad the model "auto" is not listed in the documentation http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=blob_plain;f=so… static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_LAPTOP]= "laptop", [CXT5066_DELL_LAPTOP]= "dell-laptop", [CXT5066_OLPC_XO_1_5]= "olpc-xo-1_5", [CXT5066_DELL_VOSTRO]= "dell-vostro", [CXT5066_IDEAPAD]= "ideapad", [CXT5066_THINKPAD]= "thinkpad", [CXT5066_ASUS]= "asus", [CXT5066_HP_LAPTOP]= "hp-laptop", [CXT5066_AUTO]= "auto", }; > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: Conexant CX20590 > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x14f1506e > Subsystem Id: 0x17aac02d > Revision Id: 0x100003 > Node 0x1a [Pin Complex] wcaps 0x400481: Stereo > Pincap 0x00001324: IN Detect > Vref caps: HIZ 50 80 > Pin Default 0x04a11030: [Jack] Mic at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > > Power states: D0 D1 D2 D3 D3cold EPSS > Power: setting=D0, actual=D0 > Node 0x1b [Pin Complex] wcaps 0x400581: Stereo > Pincap 0x00011334: IN OUT EAPD Detect > Vref caps: HIZ 50 80 > EAPD 0x2: EAPD > Pin Default 0x95a70120: [Fixed] Mic at Int Top > Conn = Analog, Color = Unknown > DefAssociation = 0x2, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: VREF_HIZ > Unsolicited: tag=38, enabled=1 > Power states: D0 D1 D2 D3 D3cold EPSS > Power: setting=D0, actual=D0 > Connection: 2 > 0x10* 0x11 >

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Re: [alsa-devel] sound/soc/codecs/wm2200.c:2082:2: error: unknown field 'num_ranges' specified in initializer
by Mark Brown 06 Oct '12

06 Oct '12

On Sat, Oct 06, 2012 at 09:23:04AM +0800, Fengguang Wu wrote: > All error/warnings: > sound/soc/codecs/wm2200.c:2082:2: error: unknown field 'num_ranges' specified in initializer Not a real error (nor is anything else like this in my trees right now), all the branches will be rebased after the merge window which will pull in the dependencies.

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Re: [alsa-devel] [PATCH 0/2] Convert davinci ASoC to genalloc SRAM
by Matt Porter 05 Oct '12

05 Oct '12

On Thu, Oct 04, 2012 at 06:15:43AM -0400, Matt Porter wrote: > This series converts davinci ASoC to use genalloc and enables > that support on DA850. It applies on top of the uio_pruss/genalloc > series [1] which allows DaVinci to provide a gen_pool via pdata > for driver use. > > I've tested this on the AM180x EVM. Note that prior to this, > the SRAM paths in the driver were completely unused. I've only > enabled ping-pong buffering on the platform I can test as it's > best to allow those with DM644x and similar platforms to set the > playback/capture sram size to something that's known to work > for them. > > [1] http://www.spinics.net/lists/arm-kernel/msg198854.html Tested again on top of v4 of the uio_pruss/genalloc series http://www.spinics.net/lists/arm-kernel/msg199255.html on next-20121005 since there's some conflicts falling out of of arm-soc and asoc/sound header move patches on mainline atm. -Matt

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[alsa-devel] [PATCH 1/7] ASoC: wm2200: Convert to devm APIs
by Mark Brown 05 Oct '12

05 Oct '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 39 ++++++++++++++------------------------- 1 file changed, 14 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index eab64a1..6ca9bcc 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2011,7 +2011,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, wm2200->dev = &i2c->dev; init_completion(&wm2200->fll_lock); - wm2200->regmap = regmap_init_i2c(i2c, &wm2200_regmap); + wm2200->regmap = devm_regmap_init_i2c(i2c, &wm2200_regmap); if (IS_ERR(wm2200->regmap)) { ret = PTR_ERR(wm2200->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2027,8 +2027,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, for (i = 0; i < ARRAY_SIZE(wm2200->core_supplies); i++) wm2200->core_supplies[i].supply = wm2200_core_supply_names[i]; - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); @@ -2044,8 +2045,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, } if (wm2200->pdata.ldo_ena) { - ret = gpio_request_one(wm2200->pdata.ldo_ena, - GPIOF_OUT_INIT_HIGH, "WM2200 LDOENA"); + ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.ldo_ena, + GPIOF_OUT_INIT_HIGH, + "WM2200 LDOENA"); if (ret < 0) { dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", wm2200->pdata.ldo_ena, ret); @@ -2055,8 +2057,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, } if (wm2200->pdata.reset) { - ret = gpio_request_one(wm2200->pdata.reset, - GPIOF_OUT_INIT_HIGH, "WM2200 /RESET"); + ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.reset, + GPIOF_OUT_INIT_HIGH, + "WM2200 /RESET"); if (ret < 0) { dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", wm2200->pdata.reset, ret); @@ -2166,23 +2169,16 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, err_pm_runtime: pm_runtime_disable(&i2c->dev); err_reset: - if (wm2200->pdata.reset) { + if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); - gpio_free(wm2200->pdata.reset); - } err_ldo: - if (wm2200->pdata.ldo_ena) { + if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); - gpio_free(wm2200->pdata.ldo_ena); - } err_enable: regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), wm2200->core_supplies); err_core: - regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); err_regmap: - regmap_exit(wm2200->regmap); err: return ret; } @@ -2194,17 +2190,10 @@ static __devexit int wm2200_i2c_remove(struct i2c_client *i2c) snd_soc_unregister_codec(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm2200); - if (wm2200->pdata.reset) { + if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); - gpio_free(wm2200->pdata.reset); - } - if (wm2200->pdata.ldo_ena) { + if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); - gpio_free(wm2200->pdata.ldo_ena); - } - regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); - regmap_exit(wm2200->regmap); return 0; } -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: have unique stream_names in codecs
by Peter Meerwald 05 Oct '12

05 Oct '12

From: Peter Meerwald <p.meerwald(a)bct-electronic.com> for example, there are two 'Capture' stream_names in twl4030, resulting in twl4030-codec twl4030-codec: ASoC: Failed to create Capture debugfs file with debug enabled (fails to create two sysfs entries named 'Capture' under /sys/kernel/debug/asoc/omap3beagle/twl4030-codec/dapm this patch tries to avoid duplicate stream_names Signed-off-by: Peter Meerwald <p.meerwald(a)bct-electronic.com> --- sound/soc/codecs/stac9766.c | 4 ++-- sound/soc/codecs/twl4030.c | 4 ++-- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/uda1380.c | 4 ++-- sound/soc/codecs/wm8753.c | 4 ++-- 5 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 33c0f3d..3379119 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -300,14 +300,14 @@ static struct snd_soc_dai_driver stac9766_dai[] = { /* stream cababilities */ .playback = { - .stream_name = "stac9766 analog", + .stream_name = "stac9766 playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SND_SOC_STD_AC97_FMTS, }, .capture = { - .stream_name = "stac9766 analog", + .stream_name = "stac9766 capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 391fcfc..55582db 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2172,7 +2172,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .formats = TWL4030_FORMATS, .sig_bits = 24,}, .capture = { - .stream_name = "Capture", + .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 4, .rates = TWL4030_RATES, @@ -2189,7 +2189,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .capture = { - .stream_name = "Capture", + .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index c084c54..6f3a9cf 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1046,7 +1046,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { { .name = "twl6040-ul", .capture = { - .stream_name = "Capture", + .stream_name = "Capture UL", .channels_min = 1, .channels_max = 2, .rates = TWL6040_RATES, diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 2502214..d780fe7 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -671,7 +671,7 @@ static struct snd_soc_dai_driver uda1380_dai[] = { { /* playback only - dual interface */ .name = "uda1380-hifi-playback", .playback = { - .stream_name = "Playback", + .stream_name = "Playback Only", .channels_min = 1, .channels_max = 2, .rates = UDA1380_RATES, @@ -682,7 +682,7 @@ static struct snd_soc_dai_driver uda1380_dai[] = { { /* capture only - dual interface*/ .name = "uda1380-hifi-capture", .capture = { - .stream_name = "Capture", + .stream_name = "Capture Only", .channels_min = 1, .channels_max = 2, .rates = UDA1380_RATES, diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 13bff87..783ac20 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1400,7 +1400,7 @@ static struct snd_soc_dai_driver wm8753_dai[] = { .formats = WM8753_FORMATS }, .capture = { /* dummy for fast DAI switching */ - .stream_name = "Capture", + .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, @@ -1418,7 +1418,7 @@ static struct snd_soc_dai_driver wm8753_dai[] = { .formats = WM8753_FORMATS, }, .capture = { - .stream_name = "Capture", + .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, -- 1.7.9.5

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Re: [alsa-devel] issue of shdma-base library
by Guennadi Liakhovetski 05 Oct '12

05 Oct '12

Mark, could you take this patch or would you like me to resend it with "subject" in the actual mail subject line or does it also work from the mail body? Thanks Guennadi On Thu, 4 Oct 2012, Do Q.Thang wrote: > Hi Guennadi-san > > Thank you for your hard work. > > On 10/03/2012 09:33 PM, Guennadi Liakhovetski wrote: > > Subject: [PATCH] ASoC: fsi: don't reschedule DMA from an atomic context > > > > shdma doesn't support transfer re-scheduling or triggering from callbacks > > or from atomic context. The fsi driver issues DMA transfers from a tasklet > > context, which is a bug. To fix it convert tasklet to a work. > > > > Reported-by: Do Q.Thang <dq-thang(a)jinso.co.jp> > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > > --- > > > > Thang-san, Please, check, whether the attached patch fixes your issues. It > > seems to fix them for me. > > This patch also fix them for me. > > Thanks > --- > Thang > > > Mark, please, wait with committing this patch until the guys confirm, it > > really helps. It then will also have to go to 3.6-stable. > > > > Thanks > > Guennadi > > > > On Tue, 2 Oct 2012, Do Q.Thang wrote: > > > > > Hi Guennadi-san, > > > > > > I'm testing uptream kernel (v3.6-rc3 ~) on the kzm-a9-gt & > > > armadillo-800eva > > > board. > > > I found 2 issues with the sh-dma driver while testing audio playback. > > > > > > issue 1: this log appeared while playback with the aplay, then aplay was > > > stopped. > > > > > > ------- log ----------------------------- > > > # aplay audio_s16le_44100.wav > > > Playing WAVE 'audio_s16le_44100.wav' : Signed 16 bit Little Endian, Rate > > > 44100 Hz, Stereo > > > aplay: pcm_write:1710: write error: Input/output error > > > ----------------------------------------- > > > > > > issue 2: this log appeared while playback with the aplay, then aplay was > > > stopped. > > > ( this issue isn't appear on the kzm-a9-gt board only ) > > > > > > ------- log ----------------------------- > > > # aplay audio_s16le_44100.wav > > > Playing WAVE 'audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, > > > Stereo > > > BUG: scheduling while atomic: kworker/1:1/273/0x00000103 > > > aplay: pcm_write:1710: write error: Input/output error > > > ----------------------------------------- > > > > > > I think these issues are introduced by "dmaengine:add an shdma-base > > > library" > > > commit. > > > > > > ---------- commit ----------------------- > > > commit 9a7b8e002e331d0599127f16613c32f425a14f2c > > > Author: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > > > Date: Wed May 9 17:09:13 2012 +0200 > > > > > > dmaengine: add an shdma-base library > > > > > > This patch extracts code from shdma.c, that does not directly deal > > > with > > > hardware implementation details and can be re-used with diverse DMA > > > controller variants, found on SH-based SoCs. > > > > > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > > > Cc: Sascha Hauer <s.hauer(a)pengutronix.de> > > > Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> > > > ------------------------------------------ > > > > > > About issue 1: I found that > > > The fsi.c uses fsi_dma_do_tasklet() to setup dma-transfer.This is a > > > function > > > of a tasklet. > > > When one dma-transfer is finished shdma_chan_ld_cleanup() is called.In > > > this > > > function > > > __ld_cleanup() is called to cleanup the schan->ld_queue. In behavior of > > > __ld_cleanup() > > > if descriptor is the last chunk its callback is called, then its node is > > > cleanup in the > > > next call of __ld_cleanup().In callback of the last descriptor, > > > fsi_dma_do_tasklet() is > > > scheduled then is called after the shdma_chan_ld_cleanup() is finished. > > > But when this issue appear, shdma_chan_ld_cleanup() is called while > > > shdma_chan_ld_cleanup() > > > is not finished.At that time, schan->ld_queue is not empty, so in > > > shdma_tx_submit() > > > schan->pm_state is set to SHDMA_PM_PENDING, and then dma-transfer isn't > > > started in > > > shdma_issue_pending(). > > > > > > About issue 2: > > > I found that this issue is appear when pm_runtime_barrier() is running. > > > This issue is not appear when disable CONFIG_PM_RUNTIME config. > > > > > > What do you think about these issues? > > > > > > Best regards > > > --- > > > Thang > > diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c > > index 0540408..1bb0d58c 100644 > > --- a/sound/soc/sh/fsi.c > > +++ b/sound/soc/sh/fsi.c > > @@ -20,6 +20,7 @@ > > #include <linux/sh_dma.h> > > #include <linux/slab.h> > > #include <linux/module.h> > > +#include <linux/workqueue.h> > > #include <sound/soc.h> > > #include <sound/sh_fsi.h> > > @@ -223,7 +224,7 @@ struct fsi_stream { > > */ > > struct dma_chan *chan; > > struct sh_dmae_slave slave; /* see fsi_handler_init() */ > > - struct tasklet_struct tasklet; > > + struct work_struct work; > > dma_addr_t dma; > > }; > > @@ -1085,9 +1086,9 @@ static void fsi_dma_complete(void *data) > > snd_pcm_period_elapsed(io->substream); > > } > > -static void fsi_dma_do_tasklet(unsigned long data) > > +static void fsi_dma_do_work(struct work_struct *work) > > { > > - struct fsi_stream *io = (struct fsi_stream *)data; > > + struct fsi_stream *io = container_of(work, struct fsi_stream, work); > > struct fsi_priv *fsi = fsi_stream_to_priv(io); > > struct snd_soc_dai *dai; > > struct dma_async_tx_descriptor *desc; > > @@ -1129,7 +1130,7 @@ static void fsi_dma_do_tasklet(unsigned long data) > > * FIXME > > * > > * In DMAEngine case, codec and FSI cannot be started simultaneously > > - * since FSI is using tasklet. > > + * since FSI is using the scheduler work queue. > > * Therefore, in capture case, probably FSI FIFO will have got > > * overflow error in this point. > > * in that case, DMA cannot start transfer until error was cleared. > > @@ -1153,7 +1154,7 @@ static bool fsi_dma_filter(struct dma_chan *chan, void > > *param) > > static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) > > { > > - tasklet_schedule(&io->tasklet); > > + schedule_work(&io->work); > > return 0; > > } > > @@ -1195,14 +1196,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, > > struct fsi_stream *io, struct dev > > return fsi_stream_probe(fsi, dev); > > } > > - tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); > > + INIT_WORK(&io->work, fsi_dma_do_work); > > return 0; > > } > > static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) > > { > > - tasklet_kill(&io->tasklet); > > + cancel_work_sync(&io->work); > > fsi_stream_stop(fsi, io); > > > > > --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

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