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

[alsa-devel] [RFC 0/4] DT support for rx51-audio
by Sebastian Reichel 30 Nov '13

30 Nov '13

Hi, This patchset adds DT support in rx51-audio. I tested it on the Nokia N900 and was able to play sound with aplay using earphones and earspeaker. The Loudspeakers did not work. I don't know the reason. The patchset consists of 4 patches: 1. convert rx51-audio to snd_soc_register_card() 2. init audio in rx51 platform file 3. add some DT helpers to ASoC core 4. add DT support to rx51-audio Please note, that the patch is RFC level for now. For example non DT boot does not work, because I did not yet add pdata in boardcode. Proposed DTS node would look like this: sound: rx51-audio { compatible = "nokia,rx51-audio"; ti,mcbsp = <&mcbsp2>; ti,codec = <&tlv320aic3x>; ti,codec_aux = <&tlv320aic3x_aux>; ti,headset_amp = <&tpa6130a2>; tvout-selection-gpio = <&gpio2 8 GPIO_ACTIVE_HIGH>; /* 40 */ jack-detection-gpio = <&gpio6 17 GPIO_ACTIVE_HIGH>; /* 177 */ eci-sw-gpio = <&gpio6 22 GPIO_ACTIVE_HIGH>; /* 182 */ speaker-amp-gpio = <&twl_gpio 7 GPIO_ACTIVE_HIGH>; }; -- Sebastian Pali Rohár (2): ASoC: omap: rx51: Use snd_soc_register_card ARM: OMAP: rx51: Register audio device Sebastian Reichel (2): ASoC: Allow Aux Codecs to be specified using DT ASoC: RX-51: Add DT support to sound driver .../devicetree/bindings/sound/nokia,rx51.txt | 28 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + arch/arm/mach-omap2/board-rx51-peripherals.c | 11 ++ include/sound/rx51.h | 18 ++ include/sound/soc.h | 13 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/rx51.c | 212 +++++++++++++++------ sound/soc/soc-core.c | 68 +++++-- 9 files changed, 277 insertions(+), 81 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nokia,rx51.txt create mode 100644 include/sound/rx51.h -- 1.8.4.rc3

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[alsa-devel] [PATCH] ALSA: hda - Fix silent output on ASUS W7J laptop
by Takashi Iwai 29 Nov '13

29 Nov '13

The recent kernels got regressions on ASUS W7J with ALC660 codec where no sound comes out. After a long debugging session, we found out that setting the pin control on the unused NID 0x10 is mandatory for the outputs. And, it was found out that another magic of NID 0x0f that is required for other ASUS laptops isn't needed on this machine. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66081 Reported-and-tested-by: Andrey Lipaev <lipaev(a)mail.ru> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6366a6683e22..a98f3296b891 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4494,6 +4494,7 @@ enum { ALC861_FIXUP_AMP_VREF_0F, ALC861_FIXUP_NO_JACK_DETECT, ALC861_FIXUP_ASUS_A6RP, + ALC660_FIXUP_ASUS_W7J, }; /* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */ @@ -4543,10 +4544,21 @@ static const struct hda_fixup alc861_fixups[] = { .v.func = alc861_fixup_asus_amp_vref_0f, .chained = true, .chain_id = ALC861_FIXUP_NO_JACK_DETECT, + }, + [ALC660_FIXUP_ASUS_W7J] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* ASUS W7J needs a magic pin setup on unused NID 0x10 + * for enabling outputs + */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + { } + }, } }; static const struct snd_pci_quirk alc861_fixup_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x1253, "ASUS W7J", ALC660_FIXUP_ASUS_W7J), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), -- 1.8.4.4

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[alsa-devel] [PATCH] ASoC: core: Use consistent byte ordering in snd_soc_bytes_get
by Charles Keepax 29 Nov '13

29 Nov '13

snd_soc_bytes_put treats the data in the binary control as big endian words, however snd_soc_bytes_get uses the endian of the host machine. This causes the two functions to be inconsistant with how the mask is applied on little endian machines. This patch applies the big_endian format used in snd_soc_bytes_put to snd_soc_bytes_get. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e53d87..a66783e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3212,11 +3212,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, break; case 2: ((u16 *)(&ucontrol->value.bytes.data))[0] - &= ~params->mask; + &= cpu_to_be16(~params->mask); break; case 4: ((u32 *)(&ucontrol->value.bytes.data))[0] - &= ~params->mask; + &= cpu_to_be32(~params->mask); break; default: return -EINVAL; -- 1.7.2.5

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[alsa-devel] [PATCH 1/2] ASoC: atmel: sam9x5_wm8731: fix oops when unload module
by Bo Shen 29 Nov '13

29 Nov '13

As the priv is not assigned to card->drvdata, it is NULL, so when unload module, it will cause NULL pointer oops. Assign priv to card->drvdata to fix this issue. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/atmel/sam9x5_wm8731.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 992ae38..6dc33ac 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -99,6 +99,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) card->dev = &pdev->dev; card->owner = THIS_MODULE; + card->drvdata = priv; card->dai_link = dai; card->num_links = 1; card->dapm_widgets = sam9x5_dapm_widgets; -- 1.7.9.5

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[alsa-devel] [PATCH 1/4] dt: Add bindings documentation for the ADI AXI-I2S controller
by Lars-Peter Clausen 29 Nov '13

29 Nov '13

This patch adds the devicetree documentation for the ADI AXI-SPDIF audio controller. The controller has: * One set of memory mapped register * Two clocks, one for the memory mapped register interface, one used as the audio reference clock * One DMA interface each for the transmit and receive data Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org --- .../devicetree/bindings/sound/adi,axi-i2s.txt | 29 ++++++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,axi-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt new file mode 100644 index 0000000..d9c0e53 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt @@ -0,0 +1,29 @@ +ADI AXI-I2S controller + +Required properties: + - compatible : Must be "adi,axi-i2s-1.00.a" + - reg : Must contain I2S registers location and length + - clocks : Clock specifier providing a handle to the controller's clocks. The + controller expects two clocks, the clock used for the AXI interface and the + clock used as thei sampling rate reference clock sample. + - clock-names : "axi" for the clock to the AXI interface, "ref" for the sample + rate reference clock. + - dma : Specifiers for the DMA channels that are used by the core. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + i2s: i2s@0x77600000 { + compatible = "adi,axi-i2s-1.00.a"; + reg = <0x77600000 0x1000>; + clocks = <&clk 15 &audio_clock>; + clock-names = "axi", "ref"; + dmas = <&ps7_dma 0 &ps7_dma 1>; + dma-names = "tx", "rx"; + }; -- 1.8.0

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[alsa-devel] [PATCH] ALSA: hda - allow a codec to define its own depop delay time
by mengdong.lin＠intel.com 29 Nov '13

29 Nov '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds 'depop_delay' to struct hda_codec, to indicate a depop delay time in ms when power-down, in function set_power_state() to D3. Default value is -1, for a default delay time. Machine fixup can set a suitable value according to the codec chip and HW audio design. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 1e04a20..75ec91e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1444,6 +1444,7 @@ int snd_hda_codec_new(struct hda_bus *bus, INIT_LIST_HEAD(&codec->conn_list); INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); + codec->depop_delay = -1; #ifdef CONFIG_PM spin_lock_init(&codec->power_lock); @@ -3901,8 +3902,10 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, /* this delay seems necessary to avoid click noise at power-down */ if (power_state == AC_PWRST_D3) { - /* transition time less than 10ms for power down */ - msleep(codec->epss ? 10 : 100); + if (codec->depop_delay < 0) + msleep(codec->epss ? 10 : 100); + else if (codec->depop_delay > 0) + msleep(codec->depop_delay); flags = HDA_RW_NO_RESPONSE_FALLBACK; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index bedb436..efd0edc 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -943,6 +943,8 @@ struct hda_codec { struct snd_array jacks; #endif + int depop_delay; /* depop delay in ms, -1 for default delay time */ + /* fix-up list */ int fixup_id; const struct hda_fixup *fixup_list; -- 1.8.1.2

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[alsa-devel] [RFC] ALSA: add new alsa control byte extended
by Vinod Koul 29 Nov '13

29 Nov '13

As discussed in the last Audio uConf we need to improve byte controls to allow larger size data to be sent to DSPs The modern DSPs require alsa byte controls size which far exceeds the today 512bytes limit. In order for usermode to send larger sizes (few 100s of KBs) along with size information we add extended byte control interface which sends any size bytes parameter buffer for DSPs to use Obviosly the size must be supported by the device and would be required to inform the max size allowed for the control. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/asound.h | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 9fc6219..8c10359 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -782,7 +782,8 @@ typedef int __bitwise snd_ctl_elem_type_t; #define SNDRV_CTL_ELEM_TYPE_BYTES ((__force snd_ctl_elem_type_t) 4) /* byte array */ #define SNDRV_CTL_ELEM_TYPE_IEC958 ((__force snd_ctl_elem_type_t) 5) /* IEC958 (S/PDIF) setup */ #define SNDRV_CTL_ELEM_TYPE_INTEGER64 ((__force snd_ctl_elem_type_t) 6) /* 64-bit integer type */ -#define SNDRV_CTL_ELEM_TYPE_LAST SNDRV_CTL_ELEM_TYPE_INTEGER64 +#define SNDRV_CTL_ELEM_TYPE_BYTES_EXT ((__force snd_ctl_elem_type_t) 7) /* bytes array extended */ +#define SNDRV_CTL_ELEM_TYPE_LAST SNDRV_CTL_ELEM_TYPE_BYTES_EXT typedef int __bitwise snd_ctl_elem_iface_t; #define SNDRV_CTL_ELEM_IFACE_CARD ((__force snd_ctl_elem_iface_t) 0) /* global control */ @@ -891,6 +892,11 @@ struct snd_ctl_elem_value { unsigned char data[512]; unsigned char *data_ptr; /* obsoleted */ } bytes; + struct { + unsigned long size; /* size of buffer */ + __u64 data; i /* user data pointer */ + } bytes_ext; + struct snd_aes_iec958 iec958; } value; /* RO */ struct timespec tstamp; -- 1.7.0.4

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[alsa-devel] How to properly restart after config change
by Dolevo Jay 29 Nov '13

29 Nov '13

Hi experts, In my application, I some parameters in the asound.conf at runtime. However, after my software edits and changes the necessary parameters, the application still runs with the old configuration. If I run aplay, it works fine and runs with the new config but not my application. So, could anyone tell me what I should do extra after changing the parameters? Perhaps a small restart of something? thanks.

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Re: [alsa-devel] ALSA: Add ALSA driver for Atmel Audio Bitstream DAC
by Dan Carpenter 29 Nov '13

29 Nov '13

Hello Hans-Christian Egtvedt, The patch e4967d6016b7: "ALSA: Add ALSA driver for Atmel Audio Bitstream DAC" from Feb 5, 2009, leads to the following static checker warning: "sound/atmel/abdac.c:373 set_sample_rates() error: buffer overflow 'dac->rates' 6 <= 6" sound/atmel/abdac.c 354 /* we start at 192 kHz and work our way down to 5112 Hz */ 355 while (new_rate >= RATE_MIN && index < (MAX_NUM_RATES + 1)) { index == MAX_NUM_RATES + 1 so index is 7. 356 new_rate = clk_round_rate(dac->sample_clk, 256 * new_rate); 357 if (new_rate < 0) 358 break; 359 /* make sure we are below the ABDAC clock */ 360 if (new_rate <= clk_get_rate(dac->pclk)) { 361 dac->rates[index] = new_rate / 256; index == MAX_NUM_RATES is off by one. 362 index++; 363 } 364 /* divide by 256 and then by two to get next rate */ 365 new_rate /= 256 * 2; 366 } 367 368 if (index) { 369 int i; 370 371 /* reverse array, smallest go first */ 372 for (i = 0; i < (index / 2); i++) { 373 unsigned int tmp = dac->rates[index - 1 - i]; 7 - 1 - 0 is 6, but dac->rates[] only has 6 elements so we are potentially reading beyond the end of the array here. 374 dac->rates[index - 1 - i] = dac->rates[i]; 375 dac->rates[i] = tmp; 376 } 377 378 dac->constraints_rates.count = index; 379 dac->constraints_rates.list = dac->rates; 380 dac->constraints_rates.mask = 0; 381 dac->rates_num = index; 382 383 retval = 0; 384 } regards, dan carpenter

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[alsa-devel] Remove depop delay for suspend and resume
by Kailang 29 Nov '13

29 Nov '13

Hi Takashi, Intel test alc262. They find the suspend and resume time were > 50ms. So, I delete the delay for alc_eapd_shutup and alc_resume. If we want to have depop functions, we will follow like alc283. Thanks. Kailang

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