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

[alsa-devel] [PATCH] ASoC: dwc: Add missing __iomem annotations
by Mark Brown 28 Jun '12

28 Jun '12

Otherwise sparse gets very upset with us. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index e667e2b..1bd042b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -71,12 +71,12 @@ struct dw_i2s_dev { int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); }; -static inline void i2s_write_reg(void *io_base, int reg, u32 val) +static inline void i2s_write_reg(void __iomem *io_base, int reg, u32 val) { writel(val, io_base + reg); } -static inline u32 i2s_read_reg(void *io_base, int reg) +static inline u32 i2s_read_reg(void __iomem *io_base, int reg) { return readl(io_base + reg); } -- 1.7.10

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Mark Brown 28 Jun '12

28 Jun '12

On Thu, Jun 28, 2012 at 05:26:58PM +0530, Laxman Dewangan wrote: > Other driver which can use this API are davinci, OMAP, samsung but > because they are not using the snd_dmaengine_pcm driver's api as of > now so not be very much useful for them. There's no reason why we can't factor out more common code - this is essentially what Lars-Peter was saying. > Should I send one patch per driver change or squash all 4 file > changes in one patch? Whatever will be easy or recommended? It doesn't really matter that much, but please take on board Lars-Peter's comments.

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Mark Brown 28 Jun '12

28 Jun '12

On Thu, Jun 28, 2012 at 12:31:26PM +0530, Laxman Dewangan wrote: > The pcm mmap for the substream dma buffer is done using the > dma_mmap_writecombine(). > Some of the chip sound driver like Spear, Tegra, Davinci, > ep93xx,snd_imx, snd_mxs, NUC900, OMAP, Samsung are doing this > mapping locally. > Add api in the snd dmaengine driver so that different client of > snd dmaengine can use this rather than implementing locally. OK, but I'd expect this to come along with one or more patches converting at least some of the users to the new API...

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[alsa-devel] kcontrol get method
by raj raja 28 Jun '12

28 Jun '12

Hi, Generally get and set methods of Kcontrol called via mixer controls. I want to update variable value based on my kcontrol set method. I could not call get method, like as normal functions. So any way to call particular kcontrol get method in driver itself Thanks Raj

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[alsa-devel] ALS100: no interrupts in Linux but works fine in DOS
by Ondrej Zary 27 Jun '12

27 Jun '12

Hello, I have a Freetek PRO32PNP card - an ISA card with ALS100 chip and onboard Crystal wavetable MIDI. The card works fine as SB16 in DOS - without any initialization utility, PnP BIOS configures the card to 0x220, irq5, dma1, dma5. However it does not work in Linux. Driver loads fine (snd-als100, even tried old OSS sb) but when I try to play anything, the player hangs without producing any sound. /proc/interrupts show that no interrupts are generated. Mixer works fine. The only suspicious thing is "unknown tag 0xff" and "invalid checksum" in dmesg (this is from patched kernel that shows size of the unknown tag): [ 0.159650] isapnp: Scanning for PnP cards... [ 0.246683] pnp 01:01.00: dependent set 0 (preferred) io min 0x220 max 0x220 align 0 size 16 flags 0x2 [ 0.246889] pnp 01:01.00: dependent set 0 (preferred) irq 5 flags 0x1 [ 0.247072] pnp 01:01.00: dependent set 0 (preferred) dma 5 (bitmask 0x20) flags 0x72 [ 0.247253] pnp 01:01.00: dependent set 0 (preferred) dma 1 (bitmask 0x2) flags 0x68 [ 0.247872] pnp 01:01.00: dependent set 1 (acceptable) io min 0x220 max 0x280 align 32 size 16 flags 0x0 [ 0.248078] pnp 01:01.00: dependent set 1 (acceptable) irq 5 7 9 10 flags 0x1 [ 0.248265] pnp 01:01.00: dependent set 1 (acceptable) dma 5 6 7 (bitmask 0xe0) flags 0x72 [ 0.248450] pnp 01:01.00: dependent set 1 (acceptable) dma 0 1 3 (bitmask 0xb) flags 0x68 [ 0.248963] pnp 01:01.00: dependent set 2 (functional) io min 0x100 max 0x3f0 align 16 size 16 flags 0x0 [ 0.249213] pnp 01:01.00: dependent set 2 (functional) irq 5 7 9 10 11 12 15 flags 0xf [ 0.249404] pnp 01:01.00: dependent set 2 (functional) dma 0 1 3 5 6 7 (bitmask 0xeb) flags 0x79 [ 0.249996] pnp 01:01.01: independent io min 0x388 max 0x388 align 0 size 8 flags 0x2 [ 0.250535] pnp 01:01.02: independent io min 0x200 max 0x200 align 0 size 2 flags 0x2 [ 0.251171] pnp 01:01.03: dependent set 0 (preferred) io min 0x330 max 0x330 align 0 size 2 flags 0x2 [ 0.251365] pnp 01:01.03: dependent set 0 (preferred) irq 9 flags 0x1 [ 0.251984] pnp 01:01.03: dependent set 1 (acceptable) io min 0x100 max 0x3f0 align 16 size 2 flags 0x0 [ 0.252244] pnp 01:01.03: dependent set 1 (acceptable) irq 5 7 9 10 11 12 15 flags 0xf [ 0.256256] pnp 01:01.03: unknown tag 0xff (card 1) size=18041, ignored [ 4.527093] 01:01: invalid checksum 0xad [ 4.527257] Switching to clocksource tsc [ 4.528594] 01:01: card 'ALS100 Media Audio Controller' [ 4.528699] isapnp: 1 Plug & Play card detected total I've even tried to disable Linux pnp completely (noisapnp pnpbios=off) and load snd-sb16 module manually (modprobe snd-sb16 isapnp=0) with exactly the same result (everything seems to be OK but there's no sound and interrupts). Any ideas? -- Ondrej Zary

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[alsa-devel] periods_max initialization in snd_usb_hardware
by Amit Nagal 27 Jun '12

27 Jun '12

Hi , snd_usb_hardware is defined in sound/usb/pcm.c as : static struct snd_pcm_hardware snd_usb_hardware = { ..... .buffer_bytes_max = 1024 * 1024, .period_bytes_min = 64, .period_bytes_max = 512 * 1024, .periods_min = 2, .periods_max = 1024, }; here periods_max is intialized with value = 1024 . but going by buffer_bytes_max and period_bytes_min value defined above , periods_max = ( buffer_bytes_max / period_bytes_min ) = 16384 . why this upper cap of 1024 is maintained for periods_max even though from calculation it comes as 16384 ? Thanx & Regards Amit Nagal

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[alsa-devel] how to update hw_ptr and appl_ptr
by Zhen Fu 27 Jun '12

27 Jun '12

Hi developer, I recently look the function snd_pcm_update_hw_ptr0 want to find how to update status->hw_ptr and control->appl_ptr. If you have to know this, can you tell me? Thank you very much. -- Best Regards, Zhen Fu

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[alsa-devel] [PATCH 1/2] ALSA: HDA: Support single 3-pin jack without VREF on the actual pin
by David Henningsson 27 Jun '12

27 Jun '12

Some ASUS device has a single 3-pin jack that can either be a mic or a headphone, but the pin does not have VREF capabilities. We've been told by Realtek to instead enable VREF on pin 0x18 in that case. BugLink: https://bugs.launchpad.net/bugs/1018262 Tested-by: Chih-Hsyuan Ho <chih.ho(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 49 ++++++++++++++++++++++++++++------------- 1 file changed, 34 insertions(+), 15 deletions(-) Alsa info: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1018262/+attachment/32… diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5c81ee9..6174d4a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -303,6 +303,38 @@ static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx) static void call_update_outputs(struct hda_codec *codec); static void alc_inv_dmic_sync(struct hda_codec *codec, bool force); +/* for shared I/O, change the pin-control accordingly */ +static void update_shared_mic_hp(struct hda_codec *codec, bool set_as_mic) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + hda_nid_t pin = spec->autocfg.inputs[1].pin; + /* NOTE: this assumes that there are only two inputs, the + * first is the real internal mic and the second is HP/mic jack. + */ + + val = snd_hda_get_default_vref(codec, pin); + + /* This pin does not have vref caps - let's instead enable vref on pin 0x18 + instead, as suggested by Realtek */ + if (val == AC_PINCTL_VREF_HIZ) { + const hda_nid_t vref_pin = 0x18; + /* Sanity check pin 0x18 */ + if (get_wcaps_type(get_wcaps(codec, vref_pin)) == AC_WID_PIN && + get_defcfg_connect(snd_hda_codec_get_pincfg(codec, vref_pin)) == AC_JACK_PORT_NONE) { + unsigned int vref_val = snd_hda_get_default_vref(codec, vref_pin); + if (vref_val != AC_PINCTL_VREF_HIZ) + snd_hda_set_pin_ctl(codec, vref_pin, PIN_IN | (set_as_mic ? vref_val : 0)); + } + } + + val = set_as_mic ? val | PIN_IN : PIN_HP; + snd_hda_set_pin_ctl(codec, pin, val); + + spec->automute_speaker = !set_as_mic; + call_update_outputs(codec); +} + /* select the given imux item; either unmute exclusively or select the route */ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, unsigned int idx, bool force) @@ -329,21 +361,8 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, return 0; spec->cur_mux[adc_idx] = idx; - /* for shared I/O, change the pin-control accordingly */ - if (spec->shared_mic_hp) { - unsigned int val; - hda_nid_t pin = spec->autocfg.inputs[1].pin; - /* NOTE: this assumes that there are only two inputs, the - * first is the real internal mic and the second is HP jack. - */ - if (spec->cur_mux[adc_idx]) - val = snd_hda_get_default_vref(codec, pin) | PIN_IN; - else - val = PIN_HP; - snd_hda_set_pin_ctl(codec, pin, val); - spec->automute_speaker = !spec->cur_mux[adc_idx]; - call_update_outputs(codec); - } + if (spec->shared_mic_hp) + update_shared_mic_hp(codec, spec->cur_mux[adc_idx]); if (spec->dyn_adc_switch) { alc_dyn_adc_pcm_resetup(codec, idx); -- 1.7.9.5

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Re: [alsa-devel] emu10k1 driver can't allocate memory
by Takashi Iwai 27 Jun '12

27 Jun '12

At Sun, 20 May 2012 03:04:14 +0200, Mikael Magnusson wrote: > > Hi, (please CC me on replies, not subscribed) > > I recently upgraded to 3.3.6 from 2.6.33.15 (yeah, I know). Since then > I noticed sometimes sounds were sometimes missing, then I noticed > thousands of these in dmesg, > [37280.283484] max memory size is 0x7fffffff (addr = 0x1d5d70071038000)!! I guess you are enabling IOMMU? I guess it translates the address somehow in a way emu10k1 doesn't like. There are really little changes in emu10k1 driver itself, so if the kernel upgrade changes the behavior, it must be in the core part such like IOMMU handling. Try to boot without iommu and see whether it changes the behavior again. Takashi > [37280.283486] emu: failure page = 32 > I also checked asfxload -M output, right after boot it shows all the > synth memory free as you would expect, but after a while of usage, it > is close to 0 (around 50KiB or so). Trying to run a simple program > that writes to /dev/audio fails with > write(3</dev/audio>, > "\32\21\t\3\0\0\4\n\23\34&/6:;94,#\31\17\10\2\0\0\4\v\24\36(07"..., > 4096) = -1 ENOMEM (Cannot allocate memory) > You can also note the output from asfxload -M dropping rapidly while > the program is making sound before it fails. Native alsa apps seem to > fare slightly better as the memory decrease caused by them is actually > returned when they close the device again. > > Loading and unloading soundfonts seem to work as normal, the memory is > returned when the soundfont is unloaded afaict. (I have the patch > applied that makes it possible to load soundfonts (changes the DMA32 > zone to 2GB). I don't think this is the cause of my problem since the > given addresses are way way outside the 4GB range as well.) > > Note that even well before the reported memory is at 0, apps randomly > fail to open the sound output and are silent. Curiously espeak when > successful doesn't cause a decrease in available memory, but when it > fails, it causes about 100KiB drop. > > I can't (don't dare) bisect because this is my production machine, but > I can test patches, or check the major releases between 2.6.33 and > 3.3.6 if you have no idea what causes this. > > I'm not the only one with the problem either, > http://www.google.com/search?nfpr=1&tbs=li:1&q=emu%3A+failure+page+%3D+32+m… > > -- > Mikael Magnusson >

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[alsa-devel] Configuring an audio path within ALSA
by rashmi narayan 27 Jun '12

27 Jun '12

Dear ALSA Folks, I am a newbie to ALSA .I am trying to configure a path within the driver to open up /power up the path within the codec from MIC-IN to LineOUT. How can I do this within ALSA. Please send me some pointers. Thank you, Rashmi

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[alsa-devel] [PATCH V3 1/1] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 27 Jun '12

27 Jun '12

In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as differently(like its stretched). Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- chnages in V2: modify subject to indicate ASOC and codec as tlv320aic3x changes in V3: use snd_soc_update_bits() instead of read-mask-write sound/soc/codecs/tlv320aic3x.c | 4 +--- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64d2a4f..e9b62b5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -935,9 +935,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - data | (pll_p << PLLP_SHIFT)); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6f097fb..08c7f66 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 -- 1.7.1

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[alsa-devel] [PATCH] ALSA: ASoC: Remove const attribute from lm49453's snd_soc_dai_driver
by Daniel Mack 27 Jun '12

27 Jun '12

Fixes the following build warning. CC sound/soc/codecs/lm49453.o sound/soc/codecs/lm49453.c: In function ‘lm49453_i2c_probe’: sound/soc/codecs/lm49453.c:1511:24: warning: passing argument 3 of ‘snd_soc_register_codec’ discards qualifiers from pointer target type include/sound/soc.h:333:5: note: expected ‘struct snd_soc_dai_driver *’ but argument is of type ‘const struct snd_soc_dai_driver *’ Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/lm49453.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 802b9f1..5bd1294 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1358,7 +1358,7 @@ 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 = { -- 1.7.10.2

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add AGC, MIC_BIAS
by Prchal Jiří 27 Jun '12

27 Jun '12

This patch adds settings for TLV320AIC3106 of AGC, MIC_BIAS, De-emphasis. Also added default settings of output pop-up reduction, short circuit protection and mix both channels to mono_out to get maximum level. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.new @@ -194,6 +194,11 @@ static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; static const char *aic3x_adc_hpf[] = { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; +static const char *aic3x_mic_bias[] = { "off", "2V", "2.5V", "AVDD" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -206,6 +211,13 @@ #define LINE2L_ENUM 8 #define LINE2R_ENUM 9 #define ADC_HPF_ENUM 10 +#define LAGC_LEV_ENUM 11 +#define RAGC_LEV_ENUM 12 +#define LAGC_ATT_ENUM 13 +#define RAGC_ATT_ENUM 14 +#define LAGC_DEC_ENUM 15 +#define RAGC_DEC_ENUM 16 +#define MIC_BIAS_ENUM 17 static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), @@ -219,6 +231,13 @@ SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(MICBIAS_CTRL, 6, 4, aic3x_mic_bias), }; /* @@ -345,6 +364,15 @@ * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_enum[LAGC_LEV_ENUM]), + SOC_ENUM("Right AGC Target level", aic3x_enum[RAGC_LEV_ENUM]), + SOC_ENUM("Left AGC Attack time", aic3x_enum[LAGC_ATT_ENUM]), + SOC_ENUM("Right AGC Attack time", aic3x_enum[RAGC_ATT_ENUM]), + SOC_ENUM("Left AGC Decay time", aic3x_enum[LAGC_DEC_ENUM]), + SOC_ENUM("Right AGC Decay time", aic3x_enum[RAGC_DEC_ENUM]), + + /* De-emphasis */ + SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, @@ -352,6 +380,8 @@ SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + + SOC_ENUM("Mic Bias", aic3x_enum[MIC_BIAS_ENUM]), }; /* @@ -1261,6 +1291,13 @@ snd_soc_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); + /* set to avoid artifacts on the audio output during power-on/off */ + snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_B, 0x80); /*ac-coupled*/ + snd_soc_write(codec, HPOUT_POP_REDUCTION, 0x4e); /* 10 + 4 ms, reference*/ + + /* short circuit protection */ + snd_soc_write(codec, HPRCOM_CFG, 0x04); + /* DAC default volume and mute */ snd_soc_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON); snd_soc_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON); @@ -1274,8 +1311,9 @@ snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* DAC to Mono Line Out default volume and route to Output mixer */ - snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + /* mix both channels with -6dB level */ + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, (DEFAULT_VOL + 12) | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, (DEFAULT_VOL + 12) | ROUTE_ON); /* unmute all outputs */ snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE); @@ -1292,6 +1330,12 @@ /* By default route Line1 to ADC PGA mixer */ snd_soc_write(codec, LINE1L_2_LADC_CTRL, 0x0); snd_soc_write(codec, LINE1R_2_RADC_CTRL, 0x0); + + /* AGC to -10dB, 20 / 500ms, no clip stepping, noise gate -90dB, hysteresis 3dB*/ + snd_soc_write(codec, LAGC_CTRL_A, 0x2f); + snd_soc_write(codec, RAGC_CTRL_A, 0x2f); + snd_soc_write(codec, LAGC_CTRL_C, 0xbe); + snd_soc_write(codec, RAGC_CTRL_C, 0xbe); /* PGA to HP Bypass default volume, disconnect from Output Mixer */ snd_soc_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL);

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[alsa-devel] [PATCH V2 0/9] Add ASoC drivers for SPEAr platform & I2S driver for Designware module
by Rajeev Kumar 27 Jun '12

27 Jun '12

Changes since V1: 1. Move Synopsys i2s designware controller from 'spear' folder to a new folder 'dwc' 2. Implemented regmap in sta529 codec. 3. In-corporated review comments received on V1. Rajeev Kumar (6): sound:asoc: Add support for STA529 Audio Codec sound:asoc: Add support for synopsys i2s controller as per ASoC framework. sound:asoc: Add support for SPEAr ASoC pcm layer. sound:asoc: Add support for SPEAr ASoC machine driver. sound:asoc: Add Kconfig and Makefile to support SPEAr audio driver sound:asoc: Update Kconfig and Makefile(sound/soc/) to support SPEAr audio Vipin Kumar (3): sound:asoc: Add support for spdif in Audio Codec sound:asoc:spdif_in: Add spdif IN support sound:asoc:spdif_out: Add spdif out support include/sound/designware_i2s.h | 69 ++++++ include/sound/spear_dma.h | 35 +++ include/sound/spear_spdif.h | 29 +++ sound/soc/Kconfig | 2 + sound/soc/Makefile | 2 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/spdif_receiver.c | 76 ++++++ sound/soc/codecs/sta529.c | 477 +++++++++++++++++++++++++++++++++++++ sound/soc/dwc/Kconfig | 8 + sound/soc/dwc/Makefile | 3 + sound/soc/dwc/designware_i2s.c | 454 +++++++++++++++++++++++++++++++++++ sound/soc/spear/Kconfig | 40 +++ sound/soc/spear/Makefile | 7 + sound/soc/spear/spdif_in.c | 297 +++++++++++++++++++++++ sound/soc/spear/spdif_in_regs.h | 60 +++++ sound/soc/spear/spdif_out.c | 397 ++++++++++++++++++++++++++++++ sound/soc/spear/spdif_out_regs.h | 79 ++++++ sound/soc/spear/spear_evb.c | 206 ++++++++++++++++ sound/soc/spear/spear_pcm.c | 214 +++++++++++++++++ 20 files changed, 2462 insertions(+), 1 deletions(-) create mode 100644 include/sound/designware_i2s.h create mode 100644 include/sound/spear_dma.h create mode 100644 include/sound/spear_spdif.h create mode 100644 sound/soc/codecs/spdif_receiver.c create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/dwc/Kconfig create mode 100644 sound/soc/dwc/Makefile create mode 100644 sound/soc/dwc/designware_i2s.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/spear/spdif_in.c create mode 100644 sound/soc/spear/spdif_in_regs.h create mode 100644 sound/soc/spear/spdif_out.c create mode 100644 sound/soc/spear/spdif_out_regs.h create mode 100644 sound/soc/spear/spear_evb.c create mode 100644 sound/soc/spear/spear_pcm.c

2 26

[alsa-devel] [PATCH] ALSA: snd-usb-caiaq: initialize card pointer
by Daniel Mack 27 Jun '12

27 Jun '12

Fixes the following warning: CC [M] sound/usb/caiaq/device.o sound/usb/caiaq/device.c: In function ‘snd_probe’: sound/usb/caiaq/device.c:500:16: warning: ‘card’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/caiaq/device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 64aed43..7da0d0a 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -485,7 +485,7 @@ static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; - struct snd_card *card; + struct snd_card *card = NULL; struct usb_device *device = interface_to_usbdev(intf); ret = create_card(device, intf, &card); -- 1.7.10.2

2 2

[alsa-devel] [Regression] No sound on HP Pavilion dv6, STAC92xx
by Luis Henriques 27 Jun '12

27 Jun '12

Hi, After commit b0791dda813c179e539b0fc1ecd3f5f30f2571e2 ("ALSA: hda/idt - Fix power-map for speaker-pins with some HP laptops") we have several bug reports about not having sound on HP Pavilion laptops. Here's a link to one of these bug reports: http://bugs.launchpad.net/bugs/1013183 After reverting the referred commit, everything seems to be working fine again. Any ideas/fixes? Cheers, -- Luis

2 9

[alsa-devel] [PATCH] ASoC: tlv320aic3x: add missing registers and bits
by Prchal Jiří 26 Jun '12

26 Jun '12

This patch adds missing registers and bits for TLV320AIC3106. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.new @@ -97,32 +97,20 @@ * There is no point in caching the reset register */ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { - 0x00, 0x00, 0x00, 0x10, /* 0 */ - 0x04, 0x00, 0x00, 0x00, /* 4 */ - 0x00, 0x00, 0x00, 0x01, /* 8 */ - 0x00, 0x00, 0x00, 0x80, /* 12 */ - 0x80, 0xff, 0xff, 0x78, /* 16 */ - 0x78, 0x78, 0x78, 0x78, /* 20 */ - 0x78, 0x00, 0x00, 0xfe, /* 24 */ - 0x00, 0x00, 0xfe, 0x00, /* 28 */ - 0x18, 0x18, 0x00, 0x00, /* 32 */ - 0x00, 0x00, 0x00, 0x00, /* 36 */ - 0x00, 0x00, 0x00, 0x80, /* 40 */ - 0x80, 0x00, 0x00, 0x00, /* 44 */ - 0x00, 0x00, 0x00, 0x04, /* 48 */ - 0x00, 0x00, 0x00, 0x00, /* 52 */ - 0x00, 0x00, 0x04, 0x00, /* 56 */ - 0x00, 0x00, 0x00, 0x00, /* 60 */ - 0x00, 0x04, 0x00, 0x00, /* 64 */ - 0x00, 0x00, 0x00, 0x00, /* 68 */ - 0x04, 0x00, 0x00, 0x00, /* 72 */ - 0x00, 0x00, 0x00, 0x00, /* 76 */ - 0x00, 0x00, 0x00, 0x00, /* 80 */ - 0x00, 0x00, 0x00, 0x00, /* 84 */ - 0x00, 0x00, 0x00, 0x00, /* 88 */ - 0x00, 0x00, 0x00, 0x00, /* 92 */ - 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, /* 100 */ + 0x00, 0x00, 0x00, 0x10, 0x04, 0x00, 0x00, 0x00, /* 0 */ + 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x80, /* 8 */ + 0x80, 0xff, 0xff, 0x78, 0x78, 0x78, 0x78, 0x78, /* 16 */ + 0x78, 0x00, 0x00, 0xfe, 0x00, 0x00, 0xfe, 0x00, /* 24 */ + 0x18, 0x18, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 32 */ + 0x00, 0x00, 0x00, 0x80, 0x80, 0x00, 0x00, 0x00, /* 40 */ + 0x00, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, /* 48 */ + 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, /* 56 */ + 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 64 */ + 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 72 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 80 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 88 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 96 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 104 */ }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h.new @@ -13,7 +13,7 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_CACHEREGNUM 110 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -74,6 +74,8 @@ #define HPLCOM_CFG 37 /* Right High Power Output control registers */ #define HPRCOM_CFG 38 +/* High Power Output Stage Control Register */ +#define HPOUT_SC 40 /* DAC Output Switching control registers */ #define DAC_LINE_MUX 41 /* High Power Output Driver Pop Reduction registers */ @@ -148,6 +150,17 @@ #define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 +/* New AGC registers */ +#define LAGCN_ATTACK 103 +#define LAGCN_DECAY 104 +#define RAGCN_ATTACK 105 +#define RAGCN_DECAY 106 +/* New Programmable ADC Digital Path and I2C Bus Condition Register */ +#define NEW_ADC_DIGITALPATH 107 +/* Passive Analog Signal Bypass Selection During Powerdown Register */ +#define PASSIVE_BYPASS 108 +/* DAC Quiescent Current Adjustment Register */ +#define DAC_ICC_ADJ 109 /* Page select register bits */ #define PAGE0_SELECT 0 @@ -163,6 +176,10 @@ #define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) #define LDAC2LCH (0x1 << 3) #define RDAC2RCH (0x1 << 1) +#define LDAC2RCH (0x2 << 3) +#define RDAC2LCH (0x2 << 1) +#define LDAC2MONOMIX (0x3 << 3) +#define RDAC2MONOMIX (0x3 << 1) /* PLL registers bitfields */ #define PLLP_SHIFT 0

3 3

Re: [alsa-devel] [PATCH V2] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 26 Jun '12

26 Jun '12

On Tue, Jun 26, 2012 at 14:31:42, Prchal Jiří wrote: > Hi Gururaja, > shouldn't be better to use: > > snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p); > > instead of "read mask write" ? Sure will resend the patch. Thanks for the review > Even at this place you don't need to keep rest of register (PLL_Q and PLL_enable), so yust writing is OK. > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, pll_p << PLLP_SHIFT); > > Dne 26.6.2012 08:03, Hebbar, Gururaja napsal(a): > > In sound/soc/codecs/tlv320aic3x.c > > > > data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, > > data | (pll_p<< PLLP_SHIFT)); > > > > In the above code, pll-p value is OR'ed with previous value without > > clearing it. Bug is not seen if pll-p value doesn't change across > > Sampling frequency. > > > > However on some platforms (like AM335x EVM-SK), pll-p may have different > > values across different sampling frequencies. In such case, above code > > configures the pll with a wrong value. > > Because of this bug, when a audio stream is played with pll value > > different from previous stream, audio is heard as differently. > > > > Signed-off-by: Hebbar, Gururaja<gururaja.hebbar(a)ti.com> > > --- > > chnages in V2: > > modify subject to indicate ASOC and codec as tlv320aic3x > > > > sound/soc/codecs/tlv320aic3x.c | 2 +- > > sound/soc/codecs/tlv320aic3x.h | 1 + > > 2 files changed, 2 insertions(+), 1 deletions(-) > > > > diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c > > index d0dbac1..8c1977b 100644 > > --- a/sound/soc/codecs/tlv320aic3x.c > > +++ b/sound/soc/codecs/tlv320aic3x.c > > @@ -963,7 +963,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, > > } > > > > found: > > - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); > > + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG)& ~PLLP_MASK; > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, > > data | (pll_p<< PLLP_SHIFT)); > > snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, > > diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h > > index 06a1978..16d9999 100644 > > --- a/sound/soc/codecs/tlv320aic3x.h > > +++ b/sound/soc/codecs/tlv320aic3x.h > > @@ -166,6 +166,7 @@ > > > > /* PLL registers bitfields */ > > #define PLLP_SHIFT 0 > > +#define PLLP_MASK 7 > > #define PLLQ_SHIFT 3 > > #define PLLR_SHIFT 0 > > #define PLLJ_SHIFT 2 > > Regards, Gururaja

1 0

[alsa-devel] [PATCH V2] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 26 Jun '12

26 Jun '12

In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as differently. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- chnages in V2: modify subject to indicate ASOC and codec as tlv320aic3x sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d0dbac1..8c1977b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -963,7 +963,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG) & ~PLLP_MASK; snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 06a1978..16d9999 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 -- 1.7.1

2 2

[alsa-devel] [PATCH] ASoC: tlv320aic3x: add input clock selection
by Prchal Jiří 26 Jun '12

26 Jun '12

This patch adds input selection of main codec clock - from what pin. Tested with TLV320AIC3106 on BCLK and MCLK. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.orig +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c @@ -972,6 +972,10 @@ struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + /* set clock on MCLK or GPIO2 or BCLK */ + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK, clk_id); + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK, clk_id); + aic3x->sysclk = freq; return 0; } --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h.orig +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h @@ -178,6 +178,13 @@ #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 +#define PLLCLK_IN_MASK 0x30 +#define CLKDIV_IN_MASK 0xc0 +/* clock in source */ +#define CLKIN_MCLK 0 +#define CLKIN_GPIO2 1 +#define CLKIN_BCLK 2 + /* Software reset register bits */ #define SOFT_RESET 0x80

2 2

[alsa-devel] [PATCH] ASoC: wm8994: Don't suspend accessory detection
by Mark Brown 26 Jun '12

26 Jun '12

Leave it up to the machine driver to disable accessory detection if desired, the common pattern is to have accessory detection be a wake source. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 37 ------------------------------------- 1 file changed, 37 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6474bae..a4aafdb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2967,23 +2967,8 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_codec_suspend(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->wm8994; int i, ret; - switch (control->type) { - case WM8994: - snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); - break; - case WM1811: - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, 0); - /* Fall through */ - case WM8958: - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, 0); - break; - } - for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], sizeof(struct wm8994_fll_config)); @@ -3033,28 +3018,6 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) i + 1, ret); } - switch (control->type) { - case WM8994: - if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) - snd_soc_update_bits(codec, WM8994_MICBIAS, - WM8994_MICD_ENA, WM8994_MICD_ENA); - break; - case WM1811: - if (wm8994->jackdet && wm8994->jack_cb) { - /* Restart from idle */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, - WM1811_JACKDET_MODE_JACK); - break; - } - break; - case WM8958: - if (wm8994->jack_cb) - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, WM8958_MICD_ENA); - break; - } - return 0; } #else -- 1.7.10

1 0

[alsa-devel] [PATCH] davinci:mcasp: Fix codec pll configure bug
by Hebbar, Gururaja 26 Jun '12

26 Jun '12

In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as stretched. Fix the same by clearing the mask and then writing the value. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d0dbac1..8c1977b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -963,7 +963,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG) & ~PLLP_MASK; snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 06a1978..16d9999 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 -- 1.7.1

1 1

[alsa-devel] [PATCH] ALSA: hda - Add Realtek ALC280 codec support
by David Henningsson 26 Jun '12

26 Jun '12

This chip looks very similar to ALC269 and ALC27* variants. The bug reporter has verified that sound was working after this patch had been applied. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/1017017 Tested-by: Richard Crossley <richardcrossley(a)o2.co.uk> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) Alsa info available at: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1017017/+attachment/32… diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 41475ae..a5534b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6800,6 +6800,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 }, { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 }, { .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 }, + { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, -- 1.7.9.5

2 1

Re: [alsa-devel] Using UCM with PulseAudio
by Arun Raghavan 25 Jun '12

25 Jun '12

On Thu, 2012-06-21 at 16:30 +0800, Feng Wei wrote: > 2012/6/20 Arun Raghavan <arun.raghavan(a)collabora.co.uk>: > > Liam, > > Thanks for clearing things up. > > > > On Fri, 2012-06-15 at 18:08 +0100, Liam Girdwood wrote: > > [...] > >> > The first problem is mutual exclusivity of verbs. From what I can > >> > understand, verbs are intended to be mutually exclusive -- if you have a > >> > HiFi verb and a VoiceCall verb, only one may be used at a time. We have > >> > mapped verbs to card profiles, which offer the same guarantee. However, > >> > on Android (which is a fair example of the kind of audio policy we might > >> > want), the HiFi verb PCMs maybe used while the VoiceCall PCMs are open. > >> > This is done, for example, to play an end-of-call tone from the CPU > >> > while the modem PCMs are still held open. Is there some way to do this > >> > with UCM? > >> > > >> > >> A verb is not tied to any specific PCM device here. It's intended to be > >> the highest level of audio use case where it can configure any audio > >> resource (including multiple cards) to enable the use case action. > >> > >> So for OMAP4 we would have the HiFi verb for all use case where we are > >> playing or capturing HiFi quality and the voice call verb where we are > >> making a telephone voice call. > >> > >> UCM provides the "modifier" to allow ad-hoc modifications to the audio > >> use case like above where we want to play an end of call tone. In this > >> case the verb is still VoiceCall, but PA would enable the "PlayTone" > >> modifer to play the tone (UCM can also tell Pulseaudio the sink PCM and > >> volume control for the tone data). > > > > Just as an observation, I think the Android HAL just uses PCM 0 and not > > the PCM 3, but in the PA case, I'll do it the way you describe since > > that makes more sense. > > > >> > The second problem is having separate PCMs for modifiers. In the OMAP4 > >> > profile, ringtone playback is exposed via a PlayTone modifier which > >> > corresponds to a separate PCM from regular HiFi playback. In the UCM-PA > >> > mapping we decided on, modifiers were implemented as device intended > >> > roles on a sink, so that when a stream with that role came in, we could > >> > enable the modifier, and disable it when such a stream ends. However, > >> > this doesn't account for switching the PCM on which playback is > >> > occurring. Should we be creating a separate sink for such modifiers > >> > (with lower priority, so they're not routed to unless there's a stream > >> > with the required role coming in)? Or should we be reopening the PCM for > >> > this? > >> > >> The intention for modifiers that use separate ALSA PCM sinks/sources > >> from the verb is to keep the main stream on the verb PCM source/sink and > >> the modifier stream will use the modifier PCM sink/source (this can be > >> the same PCM for some hardware). > >> > >> e.g. MP3 will be played to pcm 0 sink and ringtone to pcm 1 sink. The HW > >> will then mix both streams before they are rendered. > > > > Okay, so what this means is that we need to extend the current UCM work > > to create a second, lower priority sink for each modifier that has a > > distinct PlaybackPCM, and let the role-based routing pick that in cases > > where it makes sense. It's good that this doesn't change how we've > > mapped existing concepts -- just adds to it. > Sounds fine to me. The only problem is can we always open the modifier > pcm when modifier is not enabled. If not, I prefer to create/destroy > the sink by events. The alternative is to always add the sink and enable/disable the modifier in the I/O thread since we will get a message when a sink-input is added/removed. I'm just hacking this bit up, and will post patches if it all works out. There is another question here -- whether the modifier sink should have any ports or not. If I understand this correctly, we could let the actual routing be handled by the "main" sink's port and have the modifier sinks not touch that at all by not having any ports. In the ideal world though, it would be nice to have the ability to mark ports as being "dependent" somehow, so that port changes on the main sink would reflect in all others. This would be particularly useful for sinks and sources that have correlated paths like they do in the OMAP4 case. -- Arun

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Re: [alsa-devel] Possible message repeat
by Clemens Ladisch 25 Jun '12

25 Jun '12

Brent Weatherall wrote: > our goal is to specify a special XU that WOULD require vendors > supporting it to modify their firmware. We fully expect a vendor to > modify their firmware to support our specification. That being said, > is there a simple way to use ALSA to communicate with a vendor > specific XU as long as the software understands the XU? For example, > in the video4linux2 world, we can specify a UVC XU with a set of > controls for special processing, then vendors that want to support it > modify their firmware to support it. Then anyone can use our custom > user-space drivers to easily talk to the video device. V4L2 has > controls made for using XUs in this manner. The audio class specification does not define any mechanism for a device to specify what those controls would be. And because nobody has ever needed to use XUs so far, the current ALSA driver has no other way to create such controls. If you insist on not changing ALSA, you must bypass it. Regards, Clemens

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