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

[alsa-devel] [PATCH 1/2] swarm_cs4297: Provide definitions for AC97 registers
by Ezequiel Garcia 20 Jun '12

20 Jun '12

This patch removes the last usage of linux/ac97_codec.h by re-defining used AC97 registers. We can't use sound/ac97_codec.h here, since it is an OSS driver. Not even compilation tested. Cc: Ralf Baechle <ralf(a)linux-mips.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Ezequiel Garcia <elezegarcia(a)gmail.com> --- sound/oss/swarm_cs4297a.c | 17 ++++++++++++++++- 1 files changed, 16 insertions(+), 1 deletions(-) diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 09d4648..bd75f54 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -69,7 +69,6 @@ #include <linux/sound.h> #include <linux/slab.h> #include <linux/soundcard.h> -#include <linux/ac97_codec.h> #include <linux/pci.h> #include <linux/bitops.h> #include <linux/interrupt.h> @@ -199,6 +198,22 @@ static const char invalid_magic[] = } \ }) +/* AC97 registers */ +#define AC97_MASTER_VOL_STEREO 0x0002 /* Line Out */ +#define AC97_PCBEEP_VOL 0x000a /* none */ +#define AC97_PHONE_VOL 0x000c /* TAD Input (mono) */ +#define AC97_MIC_VOL 0x000e /* MIC Input (mono) */ +#define AC97_LINEIN_VOL 0x0010 /* Line Input (stereo) */ +#define AC97_CD_VOL 0x0012 /* CD Input (stereo) */ +#define AC97_AUX_VOL 0x0016 /* Aux Input (stereo) */ +#define AC97_PCMOUT_VOL 0x0018 /* Wave Output (stereo) */ +#define AC97_RECORD_SELECT 0x001a /* */ +#define AC97_RECORD_GAIN 0x001c +#define AC97_GENERAL_PURPOSE 0x0020 +#define AC97_3D_CONTROL 0x0022 +#define AC97_POWER_CONTROL 0x0026 +#define AC97_VENDOR_ID1 0x007c + struct list_head cs4297a_devs = { &cs4297a_devs, &cs4297a_devs }; typedef struct serdma_descr_s { -- 1.7.4.4

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Re: [alsa-devel] [PATCH 1/1] ASoC: lm49453: Fix compilation warning
by Reddy, M.R. Swami 20 Jun '12

20 Jun '12

This compilation warning issue was fixed. Patch submitted by Mark. http://mailman.alsa-project.org/pipermail/alsa-devel/2012-June/052425.html Please use the latest source repo. Thanks Swami -----Original Message----- From: Sachin Kamat [mailto:sachin.kamat@linaro.org] Sent: Wednesday, June 20, 2012 11:38 AM To: alsa-devel(a)alsa-project.org Cc: broonie(a)opensource.wolfsonmicro.com; Reddy, M.R. Swami; sachin.kamat(a)linaro.org; patches(a)linaro.org Subject: [PATCH 1/1] ASoC: lm49453: Fix compilation warning Fixes the following build warning: sound/soc/codecs/lm49453.c: In function ‘lm49453_i2c_probe’: sound/soc/codecs/lm49453.c:1510:24: warning: passing argument 3 of ‘snd_soc_register_codec’ discards ‘const’ qualifier from pointer target type [enabled by default] include/sound/soc.h:333:5: note: expected ‘struct snd_soc_dai_driver *’ but argument is of type ‘const struct snd_soc_dai_driver *’ The struct snd_soc_dai_driver lm49453_dai[] has been made non-const, in line with other snd_soc_dai_driver structure variables in other codec files. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/lm49453.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index c1bc945..99b0a9d 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1357,7 +1357,7 @@ static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { }; /* LM49453 dai structure. */ -static const struct snd_soc_dai_driver lm49453_dai[] = { +static struct snd_soc_dai_driver lm49453_dai[] = { { .name = "LM49453 Headset", .playback = { -- 1.7.4.1

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Re: [alsa-devel] [PATCH 0/2] snd_usb_audio: ignore ctrl errors on *
by Takashi Iwai 20 Jun '12

20 Jun '12

At Tue, 19 Jun 2012 22:21:17 +0200, Oleksij Rempel wrote: > > This two patches fix stability issues for two logitech cams. > I assume there is more of them, there are some bug reports on > different trackers which describe same symptoms. > > In my case i was able to reproduce this bug mostly with skype > capturing video and audio over usb on webcam. > E3500 can't work at all with pulseaudio without this patch. > The Notebook Pro cam can work but some times it get unstable. > > Ignoring ctrl errors seems to fix both cases. Thanks. Maybe digging deeper to look for the real culprit would be better, but your bandaid looks simple enough, so I applied the patches now. Takashi

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[alsa-devel] alsamixer - Mute AC97 "PCM Playback Switch" also mute "Master Playback Switch" and "Surround Playback Switch"
by Raymond Yau 20 Jun '12

20 Jun '12

Tested on Ubuntu 12.04 and Fedora 17 Live CD When PA server is running , using alsamixer -c0 to mute AC97 's "PCM Playback Switch" also mute "Master Playback Switch" and "Surround Playback Switch" but unmute "PCM Playback Switch" does not unmute "Master Playback Switch" and "Surround Playback Switch" of stac9708 alsamixer behave normally when shutdown PA server by echo "autospwan=off" > .pulse/client.conf and pulseaudio -k Why pulseaudio server behave in this way ?

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[alsa-devel] [PATCH 1/3] maestro3: Remove unused AC97 register definitions
by Ezequiel Garcia 19 Jun '12

19 Jun '12

Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Ezequiel Garcia <elezegarcia(a)gmail.com> --- sound/pci/maestro3.c | 68 -------------------------------------------------- 1 files changed, 0 insertions(+), 68 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index deef213..adb3b4c 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -361,74 +361,6 @@ MODULE_PARM_DESC(amp_gpio, "GPIO pin number for external amp. (default = -1)"); #define DSP2HOST_REQ_I2SRATE 0x02 #define DSP2HOST_REQ_TIMER 0x04 -/* AC97 registers */ -/* XXX fix this crap up */ -/*#define AC97_RESET 0x00*/ - -#define AC97_VOL_MUTE_B 0x8000 -#define AC97_VOL_M 0x1F -#define AC97_LEFT_VOL_S 8 - -#define AC97_MASTER_VOL 0x02 -#define AC97_LINE_LEVEL_VOL 0x04 -#define AC97_MASTER_MONO_VOL 0x06 -#define AC97_PC_BEEP_VOL 0x0A -#define AC97_PC_BEEP_VOL_M 0x0F -#define AC97_SROUND_MASTER_VOL 0x38 -#define AC97_PC_BEEP_VOL_S 1 - -/*#define AC97_PHONE_VOL 0x0C -#define AC97_MIC_VOL 0x0E*/ -#define AC97_MIC_20DB_ENABLE 0x40 - -/*#define AC97_LINEIN_VOL 0x10 -#define AC97_CD_VOL 0x12 -#define AC97_VIDEO_VOL 0x14 -#define AC97_AUX_VOL 0x16*/ -#define AC97_PCM_OUT_VOL 0x18 -/*#define AC97_RECORD_SELECT 0x1A*/ -#define AC97_RECORD_MIC 0x00 -#define AC97_RECORD_CD 0x01 -#define AC97_RECORD_VIDEO 0x02 -#define AC97_RECORD_AUX 0x03 -#define AC97_RECORD_MONO_MUX 0x02 -#define AC97_RECORD_DIGITAL 0x03 -#define AC97_RECORD_LINE 0x04 -#define AC97_RECORD_STEREO 0x05 -#define AC97_RECORD_MONO 0x06 -#define AC97_RECORD_PHONE 0x07 - -/*#define AC97_RECORD_GAIN 0x1C*/ -#define AC97_RECORD_VOL_M 0x0F - -/*#define AC97_GENERAL_PURPOSE 0x20*/ -#define AC97_POWER_DOWN_CTRL 0x26 -#define AC97_ADC_READY 0x0001 -#define AC97_DAC_READY 0x0002 -#define AC97_ANALOG_READY 0x0004 -#define AC97_VREF_ON 0x0008 -#define AC97_PR0 0x0100 -#define AC97_PR1 0x0200 -#define AC97_PR2 0x0400 -#define AC97_PR3 0x0800 -#define AC97_PR4 0x1000 - -#define AC97_RESERVED1 0x28 - -#define AC97_VENDOR_TEST 0x5A - -#define AC97_CLOCK_DELAY 0x5C -#define AC97_LINEOUT_MUX_SEL 0x0001 -#define AC97_MONO_MUX_SEL 0x0002 -#define AC97_CLOCK_DELAY_SEL 0x1F -#define AC97_DAC_CDS_SHIFT 6 -#define AC97_ADC_CDS_SHIFT 11 - -#define AC97_MULTI_CHANNEL_SEL 0x74 - -/*#define AC97_VENDOR_ID1 0x7C -#define AC97_VENDOR_ID2 0x7E*/ - /* * ASSP control regs */ -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: wm8904: Remove unused variable in probe()
by Mark Brown 19 Jun '12

19 Jun '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8904.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index cc5fb43..0013afe 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2054,7 +2054,6 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) static int wm8904_probe(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - struct wm8904_pdata *pdata = wm8904->pdata; int ret; codec->control_data = wm8904->regmap; -- 1.7.10

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Re: [alsa-devel] [PATCH v2 1/5] dmaengine: mmp_tdma: add mmp tdma support
by Vinod Koul 19 Jun '12

19 Jun '12

On Mon, 2012-06-11 at 18:04 +0800, Zhangfei Gao wrote: > +static int mmp_tdma_clear_chan_irq(struct mmp_tdma_chan *tdmac) > +{ > + u32 reg = readl(tdmac->reg_base + TDISR); > + > + if (reg & TDISR_COMP) { > + /* clear irq */ > + reg &= ~TDISR_COMP; > + writel(reg, tdmac->reg_base + TDISR); > + > + return 0; > + } > + return -1; > +} -1? Please return a proper linux error code. -- ~Vinod -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 2

[alsa-devel] [PATCH] ASoC: wm5100: Remove stubs of ASoC-level register map code
by Mark Brown 19 Jun '12

19 Jun '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index ba52534..33aa363 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2517,13 +2517,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) return 0; } -static int wm5100_soc_volatile(struct snd_soc_codec *codec, - unsigned int reg) -{ - return true; -} - - static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .probe = wm5100_probe, .remove = wm5100_remove, @@ -2531,8 +2524,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .set_sysclk = wm5100_set_sysclk, .set_pll = wm5100_set_fll, .idle_bias_off = 1, - .reg_cache_size = WM5100_MAX_REGISTER, - .volatile_register = wm5100_soc_volatile, .seq_notifier = wm5100_seq_notifier, .controls = wm5100_snd_controls, -- 1.7.10

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[alsa-devel] [PATCH V2 13/16] ALSA: HDA: Make hda sound card usable for Loongson.
by Huacai Chen 19 Jun '12

19 Jun '12

Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec, this patch make it usable: 1, Use 32-bit DMA as a workaround (Loongson has a hardware bug that it doesn't support DMA address above 4GB). 2, Modify patch_conexant.c to add Lemote specific code. Signed-off-by: Huacai Chen <chenhc(a)lemote.com> Signed-off-by: Hongliang Tao <taohl(a)lemote.com> Signed-off-by: Hua Yan <yanh(a)lemote.com> Cc: alsa-devel(a)alsa-project.org --- include/linux/pci_ids.h | 2 + sound/pci/hda/hda_intel.c | 5 ++++ sound/pci/hda/patch_conexant.c | 52 ++++++++++++++++++++++++++++++++++++++- 3 files changed, 57 insertions(+), 2 deletions(-) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index ab741b0..d8b0a52 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -2906,3 +2906,5 @@ #define PCI_DEVICE_ID_XEN_PLATFORM 0x0001 #define PCI_VENDOR_ID_OCZ 0x1b85 + +#define PCI_VENDOR_ID_LEMOTE 0x1c06 diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2b6392b..2b73ed4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3013,6 +3013,11 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) gcap &= ~ICH6_GCAP_64OK; } +#ifdef CONFIG_CPU_LOONGSON3 + /* Workaround: Loongson 3 doesn't support 64-bit DMA */ + gcap &= ~ICH6_GCAP_64OK; +#endif + /* disable buffer size rounding to 128-byte multiples if supported */ if (align_buffer_size >= 0) chip->align_buffer_size = !!align_buffer_size; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3acb582..1c8dfb9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int thinkpad:1; unsigned int hp_laptop:1; unsigned int asus:1; + unsigned int lemote:1; unsigned int pin_eapd_ctrls:1; unsigned int fixup_stereo_dmic:1; @@ -2278,7 +2279,7 @@ static void cxt5066_automic(struct hda_codec *codec) cxt5066_thinkpad_automic(codec); else if (spec->hp_laptop) cxt5066_hp_laptop_automic(codec); - else if (spec->asus) + else if (spec->asus || spec->lemote) cxt5066_asus_automic(codec); } @@ -2911,6 +2912,32 @@ static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = { { } /* end */ }; +static struct hda_verb cxt5066_init_verbs_lemote[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1: Connection index: 0 */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + /* ADC2, ADC3: unused */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Disable digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -2948,6 +2975,8 @@ enum { CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ + CXT5066_LEMOTE_A1004, /* Lemote Laptop A1004 */ + CXT5066_LEMOTE_A1205, /* Lemote All-In-One A1205 */ CXT5066_AUTO, /* BIOS auto-parser */ CXT5066_MODELS }; @@ -2961,6 +2990,8 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_THINKPAD] = "thinkpad", [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", + [CXT5066_LEMOTE_A1004] = "lemote-laptop-a1004", + [CXT5066_LEMOTE_A1205] = "lemote-aio-a1205", [CXT5066_AUTO] = "auto", }; @@ -2993,6 +3024,8 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), + SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT5066_LEMOTE_A1004), + SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT5066_LEMOTE_A1205), SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; @@ -3073,7 +3106,22 @@ static int patch_cxt5066(struct hda_codec *codec) spec->port_d_mode = 0; spec->mic_boost = 3; /* default 30dB gain */ break; - + case CXT5066_LEMOTE_A1004: + case CXT5066_LEMOTE_A1205: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_unsol_event; + spec->init_verbs[spec->num_init_verbs] = + cxt5066_init_verbs_lemote; + spec->num_init_verbs++; + spec->lemote = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; -- 1.7.7.3

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Re: [alsa-devel] ASoC: codecs: Add AB8500 codec-driver
by Dan Carpenter 19 Jun '12

19 Jun '12

Hello Ola Lilja, The patch 679d7abdc754: "ASoC: codecs: Add AB8500 codec-driver" from Jun 7, 2012, leads to the following Sparse warnings: sound/soc/codecs/ab8500-codec.c:1960:53: warning: cast truncates bits from constant value (1013 becomes 13) sound/soc/codecs/ab8500-codec.c:1968:53: warning: cast truncates bits from constant value (1013 becomes 13) sound/soc/codecs/ab8500-codec.c 1958 /* Set DMic-clocks to outputs */ 1959 status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC, 1960 (u8)AB8500_GPIO_DIR4_REG, ^^^^^^^^^^^^^^^^^^^^ 1961 &value8); 1962 if (status < 0) 1963 return status; 1964 value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT | 1965 GPIO31_DIR_OUTPUT; 1966 status = abx500_set_register_interruptible(codec->dev, 1967 (u8)AB8500_MISC, 1968 (u8)AB8500_GPIO_DIR4_REG, ^^^^^^^^^^^^^^^^^^^^^^^^ 1969 value); In my quick check we only seem to use the truncated values of these defines so I'm not sure this was intended. regards, dan carpenter

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Re: [alsa-devel] [PATCH 3/3] ASoC: follow the updated samsung DMA common operations
by Kukjin Kim 19 Jun '12

19 Jun '12

Boojin Kim wrote: > > Mark Brown wrote > > > This patch uses config() function to configure DMA transmit options. > > > > Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > Thanks for your ack and let me pick this up in Samsung tree. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd. > > It'd be good if we could get all the older Samsung platforms moved over > to dmaengine... > Thanks for your ack :) > And, Our engineer's preparing to move the older Samsung dma driver into > dmaengine.

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Re: [alsa-devel] [PATCH 1/3] ARM: SAMSUNG: Add config() function in DMA common operations
by Kukjin Kim 19 Jun '12

19 Jun '12

Boojin Kim wrote: > > This patch adds config() that configures DMA transmit option. > This function was originally included in request(). > But, Some DMA client driver requires to change the configuration after > request(). > So, This patch picks up it from request(). > > Signed-off-by: Boojin Kim <boojin.kim(a)samsung.com> > --- > arch/arm/plat-samsung/dma-ops.c | 77 ++++++++++++++----------- > arch/arm/plat-samsung/include/plat/dma-ops.h | 20 ++++--- > arch/arm/plat-samsung/s3c-dma-ops.c | 39 +++++++------ > 3 files changed, 77 insertions(+), 59 deletions(-) > > diff --git a/arch/arm/plat-samsung/dma-ops.c b/arch/arm/plat-samsung/dma- > ops.c > index eb9f4f5..51d8590 100644 > --- a/arch/arm/plat-samsung/dma-ops.c > +++ b/arch/arm/plat-samsung/dma-ops.c > @@ -19,72 +19,80 @@ > #include <mach/dma.h> > > static unsigned samsung_dmadev_request(enum dma_ch dma_ch, > - struct samsung_dma_info *info) > + struct samsung_dma_req *param) > { > - struct dma_chan *chan; > dma_cap_mask_t mask; > - struct dma_slave_config slave_config; > void *filter_param; > > dma_cap_zero(mask); > - dma_cap_set(info->cap, mask); > + dma_cap_set(param->cap, mask); > > /* > * If a dma channel property of a device node from device tree is > * specified, use that as the fliter parameter. > */ > - filter_param = (dma_ch == DMACH_DT_PROP) ? (void *)info- > >dt_dmach_prop : > - (void *)dma_ch; > - chan = dma_request_channel(mask, pl330_filter, filter_param); > + filter_param = (dma_ch == DMACH_DT_PROP) ? > + (void *)param->dt_dmach_prop : (void *)dma_ch; > + return (unsigned)dma_request_channel(mask, pl330_filter, > filter_param); > +} > > - if (info->direction == DMA_DEV_TO_MEM) { > +static int samsung_dmadev_release(unsigned ch, > + struct s3c2410_dma_client *client) > +{ > + dma_release_channel((struct dma_chan *)ch); > + > + return 0; > +} > + > +static int samsung_dmadev_config(unsigned ch, > + struct samsung_dma_config *param) > +{ > + struct dma_chan *chan = (struct dma_chan *)ch; > + struct dma_slave_config slave_config; > + > + if (param->direction == DMA_DEV_TO_MEM) { > memset(&slave_config, 0, sizeof(struct dma_slave_config)); > - slave_config.direction = info->direction; > - slave_config.src_addr = info->fifo; > - slave_config.src_addr_width = info->width; > + slave_config.direction = param->direction; > + slave_config.src_addr = param->fifo; > + slave_config.src_addr_width = param->width; > slave_config.src_maxburst = 1; > dmaengine_slave_config(chan, &slave_config); > - } else if (info->direction == DMA_MEM_TO_DEV) { > + } else if (param->direction == DMA_MEM_TO_DEV) { > memset(&slave_config, 0, sizeof(struct dma_slave_config)); > - slave_config.direction = info->direction; > - slave_config.dst_addr = info->fifo; > - slave_config.dst_addr_width = info->width; > + slave_config.direction = param->direction; > + slave_config.dst_addr = param->fifo; > + slave_config.dst_addr_width = param->width; > slave_config.dst_maxburst = 1; > dmaengine_slave_config(chan, &slave_config); > + } else { > + printk(KERN_WARNING "unsupported direction\n"); > + return -EINVAL; > } > > - return (unsigned)chan; > -} > - > -static int samsung_dmadev_release(unsigned ch, > - struct s3c2410_dma_client *client) > -{ > - dma_release_channel((struct dma_chan *)ch); > - > return 0; > } > > static int samsung_dmadev_prepare(unsigned ch, > - struct samsung_dma_prep_info *info) > + struct samsung_dma_prep *param) > { > struct scatterlist sg; > struct dma_chan *chan = (struct dma_chan *)ch; > struct dma_async_tx_descriptor *desc; > > - switch (info->cap) { > + switch (param->cap) { > case DMA_SLAVE: > sg_init_table(&sg, 1); > - sg_dma_len(&sg) = info->len; > - sg_set_page(&sg, pfn_to_page(PFN_DOWN(info->buf)), > - info->len, offset_in_page(info->buf)); > - sg_dma_address(&sg) = info->buf; > + sg_dma_len(&sg) = param->len; > + sg_set_page(&sg, pfn_to_page(PFN_DOWN(param->buf)), > + param->len, offset_in_page(param->buf)); > + sg_dma_address(&sg) = param->buf; > > desc = dmaengine_prep_slave_sg(chan, > - &sg, 1, info->direction, DMA_PREP_INTERRUPT); > + &sg, 1, param->direction, DMA_PREP_INTERRUPT); > break; > case DMA_CYCLIC: > - desc = dmaengine_prep_dma_cyclic(chan, > - info->buf, info->len, info->period, info->direction); > + desc = dmaengine_prep_dma_cyclic(chan, param->buf, > + param->len, param->period, param->direction); > break; > default: > dev_err(&chan->dev->device, "unsupported format\n"); > @@ -96,8 +104,8 @@ static int samsung_dmadev_prepare(unsigned ch, > return -EFAULT; > } > > - desc->callback = info->fp; > - desc->callback_param = info->fp_param; > + desc->callback = param->fp; > + desc->callback_param = param->fp_param; > > dmaengine_submit((struct dma_async_tx_descriptor *)desc); > > @@ -119,6 +127,7 @@ static inline int samsung_dmadev_flush(unsigned ch) > static struct samsung_dma_ops dmadev_ops = { > .request = samsung_dmadev_request, > .release = samsung_dmadev_release, > + .config = samsung_dmadev_config, > .prepare = samsung_dmadev_prepare, > .trigger = samsung_dmadev_trigger, > .started = NULL, > diff --git a/arch/arm/plat-samsung/include/plat/dma-ops.h b/arch/arm/plat- > samsung/include/plat/dma-ops.h > index 71a6827..f5144cd 100644 > --- a/arch/arm/plat-samsung/include/plat/dma-ops.h > +++ b/arch/arm/plat-samsung/include/plat/dma-ops.h > @@ -16,7 +16,13 @@ > #include <linux/dmaengine.h> > #include <mach/dma.h> > > -struct samsung_dma_prep_info { > +struct samsung_dma_req { > + enum dma_transaction_type cap; > + struct property *dt_dmach_prop; > + struct s3c2410_dma_client *client; > +}; > + > +struct samsung_dma_prep { > enum dma_transaction_type cap; > enum dma_transfer_direction direction; > dma_addr_t buf; > @@ -26,19 +32,17 @@ struct samsung_dma_prep_info { > void *fp_param; > }; > > -struct samsung_dma_info { > - enum dma_transaction_type cap; > +struct samsung_dma_config { > enum dma_transfer_direction direction; > enum dma_slave_buswidth width; > dma_addr_t fifo; > - struct s3c2410_dma_client *client; > - struct property *dt_dmach_prop; > }; > > struct samsung_dma_ops { > - unsigned (*request)(enum dma_ch ch, struct samsung_dma_info *info); > - int (*release)(unsigned ch, struct s3c2410_dma_client *client); > - int (*prepare)(unsigned ch, struct samsung_dma_prep_info *info); > + unsigned (*request)(enum dma_ch ch, struct samsung_dma_req *param); > + int (*release)(unsigned ch, void *param); > + int (*config)(unsigned ch, struct samsung_dma_config *param); > + int (*prepare)(unsigned ch, struct samsung_dma_prep *param); > int (*trigger)(unsigned ch); > int (*started)(unsigned ch); > int (*flush)(unsigned ch); > diff --git a/arch/arm/plat-samsung/s3c-dma-ops.c b/arch/arm/plat- > samsung/s3c-dma-ops.c > index 7814949..f99448c 100644 > --- a/arch/arm/plat-samsung/s3c-dma-ops.c > +++ b/arch/arm/plat-samsung/s3c-dma-ops.c > @@ -36,30 +36,26 @@ static void s3c_dma_cb(struct s3c2410_dma_chan > *channel, void *param, > } > > static unsigned s3c_dma_request(enum dma_ch dma_ch, > - struct samsung_dma_info *info) > + struct samsung_dma_req *param) > { > struct cb_data *data; > > - if (s3c2410_dma_request(dma_ch, info->client, NULL) < 0) { > - s3c2410_dma_free(dma_ch, info->client); > + if (s3c2410_dma_request(dma_ch, param->client, NULL) < 0) { > + s3c2410_dma_free(dma_ch, param->client); > return 0; > } > > + if (param->cap == DMA_CYCLIC) > + s3c2410_dma_setflags(dma_ch, S3C2410_DMAF_CIRCULAR); > + > data = kzalloc(sizeof(struct cb_data), GFP_KERNEL); > data->ch = dma_ch; > list_add_tail(&data->node, &dma_list); > > - s3c2410_dma_devconfig(dma_ch, info->direction, info->fifo); > - > - if (info->cap == DMA_CYCLIC) > - s3c2410_dma_setflags(dma_ch, S3C2410_DMAF_CIRCULAR); > - > - s3c2410_dma_config(dma_ch, info->width); > - > return (unsigned)dma_ch; > } > > -static int s3c_dma_release(unsigned ch, struct s3c2410_dma_client *client) > +static int s3c_dma_release(unsigned ch, void *param) > { > struct cb_data *data; > > @@ -68,16 +64,24 @@ static int s3c_dma_release(unsigned ch, struct > s3c2410_dma_client *client) > break; > list_del(&data->node); > > - s3c2410_dma_free(ch, client); > + s3c2410_dma_free(ch, param); > kfree(data); > > return 0; > } > > -static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep_info > *info) > +static int s3c_dma_config(unsigned ch, struct samsung_dma_config *param) > +{ > + s3c2410_dma_devconfig(ch, param->direction, param->fifo); > + s3c2410_dma_config(ch, param->width); > + > + return 0; > +} > + > +static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param) > { > struct cb_data *data; > - int len = (info->cap == DMA_CYCLIC) ? info->period : info->len; > + int len = (param->cap == DMA_CYCLIC) ? param->period : param->len; > > list_for_each_entry(data, &dma_list, node) > if (data->ch == ch) > @@ -85,11 +89,11 @@ static int s3c_dma_prepare(unsigned ch, struct > samsung_dma_prep_info *info) > > if (!data->fp) { > s3c2410_dma_set_buffdone_fn(ch, s3c_dma_cb); > - data->fp = info->fp; > - data->fp_param = info->fp_param; > + data->fp = param->fp; > + data->fp_param = param->fp_param; > } > > - s3c2410_dma_enqueue(ch, (void *)data, info->buf, len); > + s3c2410_dma_enqueue(ch, (void *)data, param->buf, len); > > return 0; > } > @@ -117,6 +121,7 @@ static inline int s3c_dma_stop(unsigned ch) > static struct samsung_dma_ops s3c_dma_ops = { > .request = s3c_dma_request, > .release = s3c_dma_release, > + .config = s3c_dma_config, > .prepare = s3c_dma_prepare, > .trigger = s3c_dma_trigger, > .started = s3c_dma_started, > -- > 1.7.1 Looks good to me but need to ack on spi patch in this series. I think, this series should be picked up together. Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd.

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Re: [alsa-devel] [PATCH 2/3] spi: Add the use of DMA config operation
by Kukjin Kim 19 Jun '12

19 Jun '12

Boojin Kim wrote: > > Config operation is separated from request operation in DMA common > operation. > Because spi driver can change the DMA config for every transfer. > So this patch is using the separated DMA config operation. > > Signed-off-by: Boojin Kim <boojin.kim(a)samsung.com> > Signed-off-by: Kyoungil Kim <ki0351.kim(a)samsung.com> > --- > drivers/spi/spi-s3c64xx.c | 33 +++++++++++++++++++-------------- > 1 files changed, 19 insertions(+), 14 deletions(-) > > diff --git a/drivers/spi/spi-s3c64xx.c b/drivers/spi/spi-s3c64xx.c > index 15f864d..f4e2341 100644 > --- a/drivers/spi/spi-s3c64xx.c > +++ b/drivers/spi/spi-s3c64xx.c > @@ -262,14 +262,24 @@ static void prepare_dma(struct s3c64xx_spi_dma_data > *dma, > unsigned len, dma_addr_t buf) > { > struct s3c64xx_spi_driver_data *sdd; > - struct samsung_dma_prep_info info; > + struct samsung_dma_prep info; > + struct samsung_dma_config config; > > - if (dma->direction == DMA_DEV_TO_MEM) > + if (dma->direction == DMA_DEV_TO_MEM) { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, rx_dma); > - else > + config.direction = sdd->rx_dma.direction; > + config.fifo = sdd->sfr_start + S3C64XX_SPI_RX_DATA; > + config.width = sdd->cur_bpw / 8; > + sdd->ops->config(sdd->rx_dma.ch, &config); > + } else { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, tx_dma); > + config.direction = sdd->tx_dma.direction; > + config.fifo = sdd->sfr_start + S3C64XX_SPI_TX_DATA; > + config.width = sdd->cur_bpw / 8; > + sdd->ops->config(sdd->tx_dma.ch, &config); > + } > > info.cap = DMA_SLAVE; > info.len = len; > @@ -284,20 +294,15 @@ static void prepare_dma(struct s3c64xx_spi_dma_data > *dma, > > static int acquire_dma(struct s3c64xx_spi_driver_data *sdd) > { > - struct samsung_dma_info info; > + struct samsung_dma_req req; > > sdd->ops = samsung_dma_get_ops(); > > - info.cap = DMA_SLAVE; > - info.client = &s3c64xx_spi_dma_client; > - info.width = sdd->cur_bpw / 8; > - > - info.direction = sdd->rx_dma.direction; > - info.fifo = sdd->sfr_start + S3C64XX_SPI_RX_DATA; > - sdd->rx_dma.ch = sdd->ops->request(sdd->rx_dma.dmach, &info); > - info.direction = sdd->tx_dma.direction; > - info.fifo = sdd->sfr_start + S3C64XX_SPI_TX_DATA; > - sdd->tx_dma.ch = sdd->ops->request(sdd->tx_dma.dmach, &info); > + req.cap = DMA_SLAVE; > + req.client = &s3c64xx_spi_dma_client; > + > + sdd->rx_dma.ch = sdd->ops->request(sdd->rx_dma.dmach, &req); > + sdd->tx_dma.ch = sdd->ops->request(sdd->tx_dma.dmach, &req); > > return 1; > } > -- > 1.7.1 Hi Grant, I think, this is needed to support Samsung SPI with DMA and this has dependency with other patches in this series. If you're ok on this, please reply with your ack :) so that I can take this series together. Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd.

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[alsa-devel] [PATCH] ALSA-LIB - remove Logitech Web Camera from blacklist
by Raymond Yau 19 Jun '12

19 Jun '12

- the commit which add Logitech Web Camera to the blacklist is redundant since Web Camera does not has any playback device and Pulseaudio does not support digital capture

2 2

Re: [alsa-devel] [PATCH 2/2] sound: tegra_alc5632: Adjust to of_get_named_gpio() change
by Leon Romanovsky 19 Jun '12

19 Jun '12

On Mon, Jun 18, 2012 at 8:28 PM, Alexandre Pereira da Silva <aletes.xgr(a)gmail.com> wrote: > On Mon, Jun 18, 2012 at 1:42 PM, Roland Stigge <stigge(a)antcom.de> wrote: >> of_get_named_gpio() was changed to return -EPROBE_DEFER in case of >> gpios not probed yet. This patch adjusts tegra_alc5632 to this. >> >> Signed-off-by: Roland Stigge <stigge(a)antcom.de> > Acked-by: Alexandre Pereira da Silva <aletes.xgr(a)gmail.com> Hi Alexandre, Sorry to be rude, but can you explain me why did you "Acked-by" on this patch ? I don't see your name in git log for tegra_alc5632.c file: http://git.kernel.org/?p=linux/kernel/git/broonie/sound.git;a=history;f=sou… Thanks. -- Leon Romanovsky | Independent Linux Consultant www.leon.nu | leon(a)leon.nu

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Re: [alsa-devel] [PATCH] sound/gpio: of_get_named_gpio() return -EPROBE_DEFER if GPIO deferred
by Stephen Warren 18 Jun '12

18 Jun '12

On 06/18/2012 11:22 AM, Roland Stigge wrote: > of_get_named_gpio_flags() and of_get_named_gpio() return -EPROBE_DEFER if the > respective GPIO is not (yet) available. This is useful if driver's probe() > functions try to get a GPIO whose controller isn't probed yet. Thus, the driver > can be probed again later on. > > The function still returns -EINVAL on other errors (parse error or node doesn't > exist). This way, the case of an optional/intentionally missing GPIO is handled > appropriately. > > For this change, two sound drivers (tegra_{alc5632,wm8903}) need to be > adjusted. Acked-by: Stephen Warren <swarren(a)wwwdotorg.org> (That applies to the two separate sound patches too, if needed!)

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Re: [alsa-devel] [PATCH 1/2] sound: tegra_wm8903: Adjust to of_get_named_gpio() change
by Mark Brown 18 Jun '12

18 Jun '12

On Mon, Jun 18, 2012 at 07:00:14PM +0200, Roland Stigge wrote: > Basically, yes. But this would mean that doing via gpio, we would need > to pull other stuff from the sound tree over to gpio to build upon > because "my" tegra_* sound patches won't apply to the gpio branch currently. Oh, yes. Stephen's gone and reorganised all that code which means it's not a trivial merge. Ho hum. I'm happy to apply them if folks are comfortable with it, I guess ideally I'd be able to pull a branch with the gpio changes in and then apply on top of that.

1 0

Re: [alsa-devel] [PATCH 1/2] sound: tegra_wm8903: Adjust to of_get_named_gpio() change
by Mark Brown 18 Jun '12

18 Jun '12

On Mon, Jun 18, 2012 at 06:42:21PM +0200, Roland Stigge wrote: > of_get_named_gpio() was changed to return -EPROBE_DEFER in case of > gpios not probed yet. This patch adjusts tegra_wm8903 to this. So, I saw from the other thread that this didn't actually happen yet - I guess the best thing is to merge this along with the changes for introducing -EPROBE_DEFER. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] How to redefine "default" to be equivalent to "sysdefault"?
by Tanu Kaskinen 18 Jun '12

18 Jun '12

Hi, I have a file with the following contents: # This file an alsa configuration file, used by pasuspender to make the # "default" Alsa device work while PulseAudio is in suspended state. # Include the system configuration first. Is there any easier way to do this? # 14 lines for including one file is a bit much... @hooks [ { func load files [ { @func concat strings [ { @func datadir } "/alsa.conf" ] } ] } ] # Redefine "default" to be the same as "sysdefault". On most systems # "sysdefault" is configured to use dmix. pcm.!default sysdefault That doesn't do what I want it to do. When I run ALSA_CONFIG_PATH=/path/to/file.conf aplay -Ddefault ~/test.wav aplay still uses pulseaudio. Using "-Dsysdefault" instead of "-Ddefault" doesn't use pulseaudio, so "sysdefault" does what it's supposed to do. The problem is then: why doesn't "default" behave the same way as "sysdefault"? What error did I make? Below is "aplay -L" output with and without setting the environment variable. When the environment variable is set, "default" is listed twice, and "null" and "pulse" are included in the list while they are not included in the list when running aplay without the environment variable. What causes those differences? > ALSA_CONFIG_PATH=/path/to/file.conf aplay -L default Playback/recording through the PulseAudio sound server null Discard all samples (playback) or generate zero samples (capture) pulse Playback/recording through the PulseAudio sound server default Playback/recording through the PulseAudio sound server sysdefault:CARD=Intel HDA Intel, AD198x Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, AD198x Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Intel,DEV=0 HDA Intel, AD198x Digital IEC958 (S/PDIF) Digital Audio Output > aplay -L default Playback/recording through the PulseAudio sound server sysdefault:CARD=Intel HDA Intel, AD198x Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, AD198x Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, AD198x Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Intel,DEV=0 HDA Intel, AD198x Digital IEC958 (S/PDIF) Digital Audio Output -- Tanu

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[alsa-devel] [PATCH] ALSA: Add snd_pcm_rate_bit_to_rate()
by Dimitris Papastamos 18 Jun '12

18 Jun '12

This is essentially the reverse of snd_pcm_rate_to_rate_bit(). This is generally useful as the Compress API uses the rate bit directly and it helps to be able to map back to the actual sample rate. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/pcm.h | 1 + sound/core/pcm_misc.c | 18 ++++++++++++++++++ 2 files changed, 19 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1d58d79..18ad1f1 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -892,6 +892,7 @@ extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates; int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime); unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate); +unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit); static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream, struct snd_dma_buffer *bufp) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 9c9eff9..d4fc1bf 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -488,3 +488,21 @@ unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate) return SNDRV_PCM_RATE_KNOT; } EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit); + +/** + * snd_pcm_rate_bit_to_rate - converts SNDRV_PCM_RATE_xxx bit to sample rate + * @rate_bit: the rate bit to convert + * + * Returns the sample rate that corresponds to the given SNDRV_PCM_RATE_xxx flag + * or 0 for an unknown rate bit + */ +unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit) +{ + unsigned int i; + + for (i = 0; i < snd_pcm_known_rates.count; i++) + if ((1u << i) == rate_bit) + return snd_pcm_known_rates.list[i]; + return 0; +} +EXPORT_SYMBOL(snd_pcm_rate_bit_to_rate); -- 1.7.10.4

2 2

[alsa-devel] [PATCH] snd-opti9xx: fixes for MED3931 card (opti931)
by Ondrej Zary 18 Jun '12

18 Jun '12

MED3931 card did not work (failed with "OPTI chip not found") because snd-opti9xx gets mc_indir_index from pnp by adding 2 to the pnp-reported port. It probably works for some cards but not for this one. Datasheet says that the port is always at 0xe?e so just force the lowest nibble to be 0xe. Also this card powers up with (ugly) 3D sound enabled. As there's no mixer control for this, just disable it. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -135,7 +135,6 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; - unsigned long mc_indir_size; struct resource *res_mc_indir; struct snd_wss *codec; #endif /* OPTi93X */ @@ -245,10 +244,8 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d; - if (!chip->mc_indir_index) { + if (!chip->mc_indir_index) chip->mc_indir_index = 0xe0e; - chip->mc_indir_size = 2; - } chip->password = 0xe4; chip->pwd_reg = 0; break; @@ -403,7 +400,9 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, #else /* OPTi93X */ case OPTi9XX_HW_82C931: - case OPTi9XX_HW_82C933: + /* disable 3D sound (set GPIO1 as output, low) */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c); + case OPTi9XX_HW_82C933: /* FALL THROUGH */ /* * The BTC 1817DW has QS1000 wavetable which is connected * to the serial digital input of the OPTI931. @@ -696,8 +695,7 @@ static int __devinit snd_opti9xx_read_check(struct snd_opti9xx *chip) if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1))) return 0; #else /* OPTi93X */ - chip->res_mc_indir = request_region(chip->mc_indir_index, - chip->mc_indir_size, + chip->res_mc_indir = request_region(chip->mc_indir_index, 2, "OPTi93x MC"); if (chip->res_mc_indir == NULL) return -EBUSY; @@ -770,8 +768,9 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, #ifdef OPTi93X port = pnp_port_start(pdev, 0) - 4; fm_port = pnp_port_start(pdev, 1) + 8; - chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; - chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; + /* adjust mc_indir_index - some cards report it at 0xe?d, + other at 0xe?c but it really is always at 0xe?e */ + chip->mc_indir_index = (pnp_port_start(pdev, 3) & ~0xf) | 0xe; #else devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); if (devmc == NULL) -- Ondrej Zary

2 1

[alsa-devel] [PATCH] ALSA: hda - Handle open while transitioning to D3.
by Dylan Reid 18 Jun '12

18 Jun '12

This addresses an issue encountered when a pcm is opened while transitioning to low power state (codec->power_on == 1 && codec->power_transition == -1). Add snd_pcm_power_up_d3wait to hda_codec. This function is used to power up from azx_open as opposed to snd_hda_power_up used from codec_exec_verb. When powering up from azx_open, wait for pending power downs to complete, avoiding the power up continuing in parallel with the power down on the work queue. The specific issue seen was with the CS4210 codec, it powers off the ADC and DAC nid in its suspend handler. If it is re-opened before the ~100ms power down process completes, the ADC and DAC nid are initialized while powered down and audio is lost until another suspend/resume cycle. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_codec.c | 46 ++++++++++++++++++++++++++++++++++++-------- sound/pci/hda/hda_codec.h | 2 + sound/pci/hda/hda_intel.c | 2 +- 3 files changed, 40 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 851e6ec..604699c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4423,20 +4423,19 @@ void snd_hda_update_power_acct(struct hda_codec *codec) codec->power_jiffies += delta; } -/** - * snd_hda_power_up - Power-up the codec - * @codec: HD-audio codec - * - * Increment the power-up counter and power up the hardware really when - * not turned on yet. - */ -void snd_hda_power_up(struct hda_codec *codec) +/* Transition to powered up, if wait_power_down then wait for a pending + * transition to D3 to complete. A pending D3 transition is indicated + * with power_transition == -1. */ +static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) { struct hda_bus *bus = codec->bus; spin_lock(&codec->power_lock); codec->power_count++; - if (codec->power_on || codec->power_transition > 0) { + /* Return if power_on or transitioning to power_on, unless currently + * powering down. */ + if ((codec->power_on || codec->power_transition > 0) && + !(wait_power_down && codec->power_transition < 0)) { spin_unlock(&codec->power_lock); return; } @@ -4460,8 +4459,37 @@ void snd_hda_power_up(struct hda_codec *codec) codec->power_transition = 0; spin_unlock(&codec->power_lock); } + +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the power-up counter and power up the hardware really when + * not turned on yet. + */ +void snd_hda_power_up(struct hda_codec *codec) +{ + __snd_hda_power_up(codec, false); +} EXPORT_SYMBOL_HDA(snd_hda_power_up); +/** + * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending + * D3 transition to complete. This differs from snd_hda_power_up() when + * power_transition == -1. snd_hda_power_up sees this case as a nop, + * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers + * back up. + * @codec: HD-audio codec + * + * Cancel any power down operation hapenning on the work queue, then power up. + */ +void snd_hda_power_up_d3wait(struct hda_codec *codec) +{ + /* This will cancel and wait for pending power_work to complete. */ + __snd_hda_power_up(codec, true); +} +EXPORT_SYMBOL_HDA(snd_hda_power_up_d3wait); + #define power_save(codec) \ ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 71864cdd..a4ac1de 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1059,10 +1059,12 @@ const char *snd_hda_get_jack_location(u32 cfg); */ #ifdef CONFIG_SND_HDA_POWER_SAVE void snd_hda_power_up(struct hda_codec *codec); +void snd_hda_power_up_d3wait(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); void snd_hda_update_power_acct(struct hda_codec *codec); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} +static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8f85d30..86758dd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1767,7 +1767,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) buff_step); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); - snd_hda_power_up(apcm->codec); + snd_hda_power_up_d3wait(apcm->codec); err = hinfo->ops.open(hinfo, apcm->codec, substream); if (err < 0) { azx_release_device(azx_dev); -- 1.7.9.rc0

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[alsa-devel] [PATCH] ALSA: snd-usb: fix sync pipe check
by Daniel Mack 18 Jun '12

18 Jun '12

Fix a bogus sanity check for sync pipe in pcm.c. This flaw was introduced during the streaming logic refactorization. While at it, improve the error messages that are generated in such cases. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: <ben(a)b1c1l1.com> --- sound/usb/pcm.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index cdf8b76..fc7ce7c 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -354,17 +354,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && get_endpoint(alts, 1)->bSynchAddress != 0 && !implicit_fb)) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + get_endpoint(alts, 1)->bmAttributes, + get_endpoint(alts, 1)->bLength, + get_endpoint(alts, 1)->bSynchAddress); return -EINVAL; } ep = get_endpoint(alts, 1)->bEndpointAddress; - if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + if (!implicit_fb && + get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || - (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) || - ( is_playback && !implicit_fb))) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + is_playback, ep, get_endpoint(alts, 0)->bSynchAddress); return -EINVAL; } -- 1.7.10.2

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[alsa-devel] [PATCH] ALSA: 6fire: use NULL instead of 0 for pointer assignment
by Daniel Mack 18 Jun '12

18 Jun '12

Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Torsten Schenk <torsten.schenk(a)zoho.com> --- sound/usb/6fire/firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 6f9715a..56ad923 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -209,7 +209,7 @@ static int usb6fire_fw_ezusb_upload( int ret; u8 data; struct usb_device *device = interface_to_usbdev(intf); - const struct firmware *fw = 0; + const struct firmware *fw = NULL; struct ihex_record *rec = kmalloc(sizeof(struct ihex_record), GFP_KERNEL); -- 1.7.10.2

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[alsa-devel] [PATCH] ALSA: snd-usb: make snd_usb_substream_capture_trigger static
by Daniel Mack 18 Jun '12

18 Jun '12

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index cdf8b76..ce22f5f 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1147,7 +1147,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea return -EINVAL; } -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) +static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, + int cmd) { int err; struct snd_usb_substream *subs = substream->runtime->private_data; -- 1.7.10.2

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