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[alsa-devel] [PATCH] ASoC: cs42l52: fix default value for MASTERA_VOL.
by Nicolas Schichan 28 May '13

28 May '13

The default register value for MASTERA_VOL is 0x00, the same as MASTERB_VOL. Signed-off-by: Nicolas Schichan <nschichan(a)freebox.fr> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 8465c1fa..030f53c 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = { { CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */ { CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */ { CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */ - { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */ + { CS42L52_MASTERA_VOL, 0x00 }, /* r20 Master A Volume */ { CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */ { CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */ { CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */ -- 1.7.10.4

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Re: [alsa-devel] [PATCH] ASoC: fsi: fixup sparse errors
by Mark Brown 28 May '13

28 May '13

On Fri, May 24, 2013 at 12:30:07AM -0700, Kuninori Morimoto wrote: > > ping ? Don't top post and don't send contentless pings.

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[alsa-devel] [PATCH] ASoC: ux500: Ensure consistent configuration between DAIs
by Fabio Baltieri 28 May '13

28 May '13

Current implementation of mop500_ab8500 allows for inconsistent sample rate and channel count configuration between the playback and recording interfaces, through in the hardware the two MSP controllers share common clock and frame sync signals. This patch adds the necessary code to ensure that the two device are configure consistently. The check is added at machine driver level, as how to lock DAI configuration depend of the actual hardware implementation. Signed-off-by: Fabio Baltieri <fabio.baltieri(a)linaro.org> --- sound/soc/ux500/mop500_ab8500.c | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 5e0f146..7e923ec 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -16,6 +16,7 @@ #include <linux/device.h> #include <linux/io.h> #include <linux/clk.h> +#include <linux/mutex.h> #include <sound/soc.h> #include <sound/soc-dapm.h> @@ -44,6 +45,12 @@ static unsigned int tx_slots = DEF_TX_SLOTS; static unsigned int rx_slots = DEF_RX_SLOTS; +/* Configuration consistency parameters */ +static DEFINE_MUTEX(mop500_ab8500_params_lock); +static unsigned long mop500_ab8500_usage; +static int mop500_ab8500_rate; +static int mop500_ab8500_channels; + /* Clocks */ static const char * const enum_mclk[] = { "SYSCLK", @@ -231,6 +238,21 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, substream->name, substream->number); + /* Ensure configuration consistency between DAIs */ + mutex_lock(&mop500_ab8500_params_lock); + if (mop500_ab8500_usage) { + if (mop500_ab8500_rate != params_rate(params) || + mop500_ab8500_channels != params_channels(params)) { + mutex_unlock(&mop500_ab8500_params_lock); + return -EBUSY; + } + } else { + mop500_ab8500_rate = params_rate(params); + mop500_ab8500_channels = params_channels(params); + } + __set_bit(cpu_dai->id, &mop500_ab8500_usage); + mutex_unlock(&mop500_ab8500_params_lock); + channels = params_channels(params); switch (params_format(params)) { @@ -329,9 +351,22 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, return 0; } +static int mop500_ab8500_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + mutex_lock(&mop500_ab8500_params_lock); + __clear_bit(cpu_dai->id, &mop500_ab8500_usage); + mutex_unlock(&mop500_ab8500_params_lock); + + return 0; +} + struct snd_soc_ops mop500_ab8500_ops[] = { { .hw_params = mop500_ab8500_hw_params, + .hw_free = mop500_ab8500_hw_free, .startup = mop500_ab8500_startup, .shutdown = mop500_ab8500_shutdown, } -- 1.8.2

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[alsa-devel] [PATCH] ARM: dts: wm8994: Add wm8994 regulator support on smdk5250.
by Padmavathi Venna 28 May '13

28 May '13

This patch adds the required regulator supplies and properties for wm8994 codec on smdk5250 board. Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- arch/arm/boot/dts/exynos5250-smdk5250.dts | 53 +++++++++++++++++++++++++++- 1 files changed, 51 insertions(+), 2 deletions(-) diff --git a/arch/arm/boot/dts/exynos5250-smdk5250.dts b/arch/arm/boot/dts/exynos5250-smdk5250.dts index f2a025e..42fc79e 100644 --- a/arch/arm/boot/dts/exynos5250-smdk5250.dts +++ b/arch/arm/boot/dts/exynos5250-smdk5250.dts @@ -37,6 +37,46 @@ }; }; + avdd2:fixed-regulator@0 { + compatible = "regulator-fixed"; + regulator-name = "avdd2-supply"; + regulator-min-microvolt = <1800000>; + regulator-max-microvolt = <1800000>; + regulator-always-on; + }; + + cpvdd:fixed-regulator@1 { + compatible = "regulator-fixed"; + regulator-name = "cpvdd-supply"; + regulator-min-microvolt = <1800000>; + regulator-max-microvolt = <1800000>; + regulator-always-on; + }; + + dbvdd:fixed-regulator@2 { + compatible = "regulator-fixed"; + regulator-name = "dbvdd-supply"; + regulator-min-microvolt = <3300000>; + regulator-max-microvolt = <3300000>; + regulator-always-on; + }; + + spkvdd1:fixed-regulator@3 { + compatible = "regulator-fixed"; + regulator-name = "spkvdd1-supply"; + regulator-min-microvolt = <5000000>; + regulator-max-microvolt = <5000000>; + regulator-always-on; + }; + + spkvdd2:fixed-regulator@4 { + compatible = "regulator-fixed"; + regulator-name = "spkvdd2-supply"; + regulator-min-microvolt = <5000000>; + regulator-max-microvolt = <5000000>; + regulator-always-on; + }; + i2c@12C70000 { samsung,i2c-sda-delay = <100>; samsung,i2c-max-bus-freq = <20000>; @@ -47,8 +87,17 @@ }; wm8994: wm8994@1a { - compatible = "wlf,wm8994"; - reg = <0x1a>; + compatible = "wlf,wm8994"; + reg = <0x1a>; + + gpio-controller; + #gpio-cells = <2>; + + AVDD2-supply = <&avdd2>; + CPVDD-supply = <&cpvdd>; + DBVDD-supply = <&dbvdd>; + SPKVDD1-supply = <&spkvdd1>; + SPKVDD2-supply = <&spkvdd2>; }; }; -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: wm8994: use the correct pointer to get the control value
by Vinod Koul 28 May '13

28 May '13

Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index f1c54af..49b2b00 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1032,7 +1032,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; int i; int dac; -- 1.7.0.4

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[alsa-devel] [PATCH] ALSA: hda - Enable mic-mute LED on more HP laptops
by Takashi Iwai 28 May '13

28 May '13

The newer HP laptops have SSID 103c:20xx and 103c:21xx, and these usually have the mic-mute LED on Fn-F8. Let's enable it, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1d9d642..9b6cb27 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2233,6 +2233,10 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP Folio", STAC_92HD83XXX_HP_MIC_LED), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x1900, "HP", STAC_92HD83XXX_HP_MIC_LED), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x2000, + "HP", STAC_92HD83XXX_HP_MIC_LED), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x2100, + "HP", STAC_92HD83XXX_HP_MIC_LED), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3388, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3389, -- 1.8.2.3

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[alsa-devel] [PATCH V3 0/4] clk: Samsung: audss: Register audio subsytem clocks using common clk framework
by Padmavathi Venna 28 May '13

28 May '13

Samsung S5PV210 and Exynos SoC has a separate subsystem for audio. This subsystem has a internal clock controller which controls i2s0 and pcm0 clocks. This patch series adds the Samsung audio subsytem clock to the common clock framework and provides the I2S controllers clock information in the dtsi file. This patch series is made based on Kukjin Kim for-next branch Changes since V2: - Removed s5pv210 compatible name from driver as it is not yet supported which is different from Exynos series audio subsystem clock conroller. - Removed clkdev lookup support and added alias names in the i2s0 controller node. Changes since V1: - Reworked on all review comments by Sylwester Nawrocki - Added a header file for all clock indexes as requested by Sylwester - Added different compatible names for s5pv210, exynos4 and exynos5 - Registered the pcm clocks with common clock framework Padmavathi Venna (4): ARM: samsung: use #include for all device trees clk: samsung: register audio subsystem clocks using common clock framework ARM: dts: add Exynos audio subsystem clock controller node ARM: dts: add clock provider information for i2s controllers in Exynos5250 .../bindings/clock/clk-samsung-audss.txt | 64 ++++++++++ arch/arm/boot/dts/exynos4.dtsi | 2 +- arch/arm/boot/dts/exynos4210-origen.dts | 2 +- arch/arm/boot/dts/exynos4210-smdkv310.dts | 2 +- arch/arm/boot/dts/exynos4210-trats.dts | 2 +- arch/arm/boot/dts/exynos4210-universal_c210.dts | 2 +- arch/arm/boot/dts/exynos4210.dtsi | 4 +- arch/arm/boot/dts/exynos4212.dtsi | 2 +- arch/arm/boot/dts/exynos4412-odroidx.dts | 2 +- arch/arm/boot/dts/exynos4412-origen.dts | 2 +- arch/arm/boot/dts/exynos4412-smdk4412.dts | 2 +- arch/arm/boot/dts/exynos4412.dtsi | 2 +- arch/arm/boot/dts/exynos4x12.dtsi | 4 +- arch/arm/boot/dts/exynos5250-arndale.dts | 2 +- arch/arm/boot/dts/exynos5250-smdk5250.dts | 2 +- arch/arm/boot/dts/exynos5250-snow.dts | 4 +- arch/arm/boot/dts/exynos5250.dtsi | 23 +++- arch/arm/boot/dts/exynos5440-sd5v1.dts | 2 +- arch/arm/boot/dts/exynos5440-ssdk5440.dts | 2 +- arch/arm/boot/dts/exynos5440.dtsi | 2 +- arch/arm/boot/dts/s3c2416-smdk2416.dts | 2 +- arch/arm/boot/dts/s3c2416.dtsi | 4 +- arch/arm/boot/dts/s3c24xx.dtsi | 2 +- drivers/clk/samsung/Makefile | 1 + drivers/clk/samsung/clk-samsung-audss.c | 133 ++++++++++++++++++++ include/dt-bindings/clk/samsung-audss-clk.h | 25 ++++ 26 files changed, 269 insertions(+), 27 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/clk-samsung-audss.txt create mode 100644 drivers/clk/samsung/clk-samsung-audss.c create mode 100644 include/dt-bindings/clk/samsung-audss-clk.h -- 1.7.4.4

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[alsa-devel] Problem opening default device in haswell system using aplay
by Stefano Panella 28 May '13

28 May '13

Hi everyone, sorry if this email has been sent twice... ( I sent it once before subscribing) I am working on some haswell systems running Ubuntu 12.04 and I have a problem if I run aplay -D sysdefault. when I do: cat /proc/asound/pcm I get: 00-03: HDMI 0 : HDMI 0 : playback 1 00-07: HDMI 1 : HDMI 1 : playback 1 00-08: HDMI 2 : HDMI 2 : playback 1 01-00: ALC662 rev3 Analog : ALC662 rev3 Analog : playback 1 : capture 1 01-02: ALC662 rev3 Analog : ALC662 rev3 Analog : capture 1 If I do: aplay -L | grep sysdefault I get: sysdefault:CARD=PCH where PCH is card number 1 (id = 1) on most of my systems (non haswell laptops) the default card is number 0 If I run aplay -D sysdefault I get: ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave aplay: main:682: audio open error: No such file or directory if I strace it I get: open("/dev/snd/pcmC0D0p", O_RDWR|O_NONBLOCK|O_CLOEXEC) = -1 ENOENT (No such file or directory) If I run: aplay -D hw:PCH,0 all works fine... Any idea why hw:PCH,0 works and sysdefault does not? I guess there could be a problem in the mapping of devices in hw systems? Has anybody else observed a similar issue? Or has this been a problem in the past already fixed upstream? If so, where was it fixed (kernel, lib, tools, utils, etc.) ? Thanks for all your help on this. In case this is a real problem I am willing to help debugging it and submit patches. Stefano

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[alsa-devel] [PATCH] Remove an obsolete macro in configure.in
by olivier＠trillion01.com 28 May '13

28 May '13

From: Olivier Langlois <olivier(a)trillion01.com> Replaced AM_CONFIG_HEADER with AC_CONFIG_HEADERS lano1106@whippet2 ~/dev/alsa-lib $ ./gitcompile aclocal: warning: autoconf input should be named 'configure.ac', not 'configure.in' configure.in:56: error: 'AM_CONFIG_HEADER': this macro is obsolete. You should use the 'AC_CONFIG_HEADERS' macro instead. /usr/share/aclocal-1.13/obsolete-err.m4:12: AM_CONFIG_HEADER is expanded from... configure.in:56: the top level autom4te: /usr/bin/m4 failed with exit status: 1 aclocal: error: echo failed with exit status: 1 Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> diff --git a/configure.in b/configure.in index ed3fe04..207e154 100644 --- a/configure.in +++ b/configure.in @@ -53,7 +53,7 @@ CC_NOUNDEFINED dnl Checks for header files. AC_HEADER_STDC -AM_CONFIG_HEADER(include/config.h) +AC_CONFIG_HEADERS(include/config.h) dnl Checks for typedefs, structures, and compiler characteristics. AC_C_CONST -- 1.8.2.3

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[alsa-devel] [PATCH] ASoC: wm8994: Ensure lambda is zeroed for WM8994
by Mark Brown 27 May '13

27 May '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5f51127..f5eb644 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1,3 +1,4 @@ +#define DEBUG /* * wm8994.c -- WM8994 ALSA SoC Audio driver * @@ -2078,6 +2079,7 @@ static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll, /* Move down to proper range now rounding is done */ fll->k = K / 10; + fll->lambda = 0; pr_debug("N=%x K=%x\n", fll->n, fll->k); break; @@ -3129,9 +3131,7 @@ static int wm8994_codec_suspend(struct snd_soc_codec *codec) static int wm8994_codec_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->wm8994; int i, ret; - unsigned int val, mask; for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { if (!wm8994->fll_suspend[i].out) @@ -3986,7 +3986,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case 0: case 1: wm8994->hubs.no_series_update = 1; - wm8994->hubs.no_runtime_dcs = true; + //wm8994->hubs.no_runtime_dcs = true; break; default: break; -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: bells: Hookup DMICs for Bells
by Charles Keepax 27 May '13

27 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/bells.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index ceed466..29e2468 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -350,8 +350,16 @@ static struct snd_soc_codec_conf bells_codec_conf[] = { }, }; +static struct snd_soc_dapm_widget bells_widgets[] = { + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + static struct snd_soc_dapm_route bells_routes[] = { { "Sub CLK_SYS", NULL, "OPCLK" }, + + { "DMIC", NULL, "MICBIAS2" }, + { "IN2L", NULL, "DMIC" }, + { "IN2R", NULL, "DMIC" }, }; static struct snd_soc_card bells_cards[] = { @@ -365,6 +373,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), @@ -383,6 +393,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), @@ -401,6 +413,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: wm8994: Fix reporting of accessory removal on WM8958
by Mark Brown 27 May '13

27 May '13

During recent refactoring the code to report removal when MICDET reports an absent microphone was removed, causing problems for systems which rely solely on the MICDET for this functionality. Restore it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0bcbd12..5f51127 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3856,6 +3856,11 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_dbg(codec->dev, "Ignoring removed jack\n"); return IRQ_HANDLED; } + } else if (!(reg & WM8958_MICD_STS)) { + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + SND_JACK_MECHANICAL | SND_JACK_HEADSET | + wm8994->btn_mask); + goto out; } if (wm8994->mic_detecting) -- 1.7.10.4

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[alsa-devel] Haswell: Ensuring HDA codec pins refer to physical outputs
by David Henningsson 27 May '13

27 May '13

Hi, I want to take this problem up again, because it's important we get this right. The HDA driver assumes that a codec pin widget node always refers to the same physical output. With Haswell, it seems like this is not guaranteed to be true. I would like to see this fixed on the graphics side. If not, I don't know how to work around it on the audio side. The problems that occur on the audio side are: 1) Some BIOSes set default pin config. E g, if the machine has a single HDMI out, it can set two of the codec pins to "not connected" and let the third remain "jack". As a result, the HDA driver will ignore the two codec pins and only enable the third. As such, HDMI audio will not work correctly, unless it's the third codec pin that is connected to the physical output. 2) Saving and restoring mutes, volumes etc is done on a per-pin basis. E g, imagine that a user has a dual monitor setup and always wants audio output from the left side monitor, and keep the right side monitor silent. If it is not reliable which codec pin refers to which physical output, one day suddenly the sound might come out on the right side monitor instead. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH] alsactl init: ca0106 file was omitted
by Gabriel M. Beddingfield 27 May '13

27 May '13

The file alsactl/init/00main includes the line: CARDINFO{driver}=="CA0106", INCLUDE="ca0106", GOTO="init_end" However, the file alsactl/init/ca0106 is not making it into the release tarballs (nor to user installations). This patch adds the ca0106 file to the build. Signed-off-by: Gabriel M. Beddingfield <gabrbedd(a)gmail.com> --- CAVEAT EMPTOR: I found this issue while helping someone in the #alsa channel. I have no way to test this patch. alsactl/init/Makefile.am | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/alsactl/init/Makefile.am b/alsactl/init/Makefile.am index 9e2b83d..64dac41 100644 --- a/alsactl/init/Makefile.am +++ b/alsactl/init/Makefile.am @@ -1,7 +1,7 @@ init_files = \ 00main default help info test \ - hda + hda ca0106 EXTRA_DIST = $(init_files) alsainitdir = $(datadir)/alsa/init alsainit_DATA = $(init_files) -- 1.7.9.5

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[alsa-devel] [PATCH: TinyCompress 1/2] add utils file with rate conversion helper
by Vinod Koul 26 May '13

26 May '13

Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Android.mk | 2 +- include/tinycompress/tinycompress.h | 2 + makefile.linux | 2 +- utils.c | 96 +++++++++++++++++++++++++++++++++++ 4 files changed, 100 insertions(+), 2 deletions(-) create mode 100644 utils.c diff --git a/Android.mk b/Android.mk index b9c52e2..104dfef 100644 --- a/Android.mk +++ b/Android.mk @@ -2,7 +2,7 @@ LOCAL_PATH:= $(call my-dir) include $(CLEAR_VARS) LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include -LOCAL_SRC_FILES:= compress.c +LOCAL_SRC_FILES:= compress.c utils.c LOCAL_MODULE := libtinycompress LOCAL_SHARED_LIBRARIES:= libcutils libutils LOCAL_MODULE_TAGS := optional diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 9867afc..e947322 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -267,4 +267,6 @@ const char *compress_get_error(struct compress *compress); #define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ #define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ +/* utility functions */ +unsigned int compress_get_alsa_rate(unsigned int rate); #endif diff --git a/makefile.linux b/makefile.linux index 60d6c04..677878d 100644 --- a/makefile.linux +++ b/makefile.linux @@ -2,7 +2,7 @@ LIB = libtinycompress BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c +SRC = cplay.c utils.c LIBOBJ = ${LIBSRC:.c=.o} OBJ = ${SRC:.c=.o} diff --git a/utils.c b/utils.c new file mode 100644 index 0000000..fa37f23 --- /dev/null +++ b/utils.c @@ -0,0 +1,96 @@ +/* + * BSD LICENSE + * + * tinycompress utility functions + * Copyright (c) 2011-2013, Intel Corporation + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)intel.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * tinycompress utility functions + * Copyright (c) 2011-2013, Intel Corporation + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ +#include <stdio.h> +#include <stdlib.h> +#include <stdbool.h> +#include <linux/types.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "tinycompress/tinycompress.h" + + +unsigned int compress_get_alsa_rate(unsigned int rate) +{ + switch (rate) { + case 5512: + return SNDRV_PCM_RATE_5512; + case 8000: + return SNDRV_PCM_RATE_8000; + case 11025: + return SNDRV_PCM_RATE_11025; + case 16000: + return SNDRV_PCM_RATE_16000; + case 220500: + return SNDRV_PCM_RATE_22050; + case 32000: + return SNDRV_PCM_RATE_32000; + case 44100: + return SNDRV_PCM_RATE_44100; + case 48000: + return SNDRV_PCM_RATE_48000; + case 64000: + return SNDRV_PCM_RATE_64000; + case 88200: + return SNDRV_PCM_RATE_88200; + case 96000: + return SNDRV_PCM_RATE_96000; + case 176400: + return SNDRV_PCM_RATE_176400; + case 192000: + return SNDRV_PCM_RATE_192000; + default: + return 0; + } +} -- 1.7.0.4

1 2

[alsa-devel] [PATCH: TinyCompress] fix error reporting in tinycompress
by Vinod Koul 26 May '13

26 May '13

From: Lakshmi N Vinnakota <lakshmi.n.vinnakota(a)intel.com> The compress API are expected to return a negative value on an error with errno indicating the type of error. But in some cases, this is not happening. The fix is the oops function always returns -1 and the actual error number is determined by the global variable errno. Signed-off-by: Lakshmi N Vinnakota <lakshmi.n.vinnakota(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- compress.c | 61 ++++++++++++++++++++++++++++++----------------------------- 1 files changed, 31 insertions(+), 30 deletions(-) diff --git a/compress.c b/compress.c index 8753758..7f56c78 100644 --- a/compress.c +++ b/compress.c @@ -102,10 +102,11 @@ static int oops(struct compress *compress, int e, const char *fmt, ...) va_end(ap); sz = strlen(compress->error); - if (errno) - snprintf(compress->error + sz, COMPR_ERR_MAX - sz, - ": %s", strerror(e)); - return e; + snprintf(compress->error + sz, COMPR_ERR_MAX - sz, + ": %s", strerror(e)); + errno = e; + + return -1; } const char *compress_get_error(struct compress *compress) @@ -151,27 +152,27 @@ static bool _is_codec_supported(struct compress *compress, struct compr_config * } } if (codec == false) { - oops(compress, -ENXIO, "this codec is not supported"); + oops(compress, ENXIO, "this codec is not supported"); return false; } if (config->fragment_size < caps->min_fragment_size) { - oops(compress, -EINVAL, "requested fragment size %d is below min supported %d", + oops(compress, EINVAL, "requested fragment size %d is below min supported %d", config->fragment_size, caps->min_fragment_size); return false; } if (config->fragment_size > caps->max_fragment_size) { - oops(compress, -EINVAL, "requested fragment size %d is above max supported %d", + oops(compress, EINVAL, "requested fragment size %d is above max supported %d", config->fragment_size, caps->max_fragment_size); return false; } if (config->fragments < caps->min_fragments) { - oops(compress, -EINVAL, "requested fragments %d are below min supported %d", + oops(compress, EINVAL, "requested fragments %d are below min supported %d", config->fragments, caps->min_fragments); return false; } if (config->fragments > caps->max_fragments) { - oops(compress, -EINVAL, "requested fragments %d are above max supported %d", + oops(compress, EINVAL, "requested fragments %d are above max supported %d", config->fragments, caps->max_fragments); return false; } @@ -219,7 +220,7 @@ struct compress *compress_open(unsigned int card, unsigned int device, char fn[256]; if (!config) { - oops(&bad_compress, -EINVAL, "passed bad config"); + oops(&bad_compress, EINVAL, "passed bad config"); return &bad_compress; } @@ -242,7 +243,7 @@ struct compress *compress_open(unsigned int card, unsigned int device, compress->flags = flags; if (!((flags & COMPRESS_OUT) || (flags & COMPRESS_IN))) { - oops(&bad_compress, -EINVAL, "can't deduce device direction from given flags"); + oops(&bad_compress, EINVAL, "can't deduce device direction from given flags"); goto config_fail; } @@ -315,12 +316,12 @@ int compress_get_hpointer(struct compress *compress, __u64 time; if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &kavail)) return oops(compress, errno, "cannot get avail"); if (0 == kavail.tstamp.sampling_rate) - return oops(compress, -ENODATA, "sample rate unknown"); + return oops(compress, ENODATA, "sample rate unknown"); *avail = (unsigned int)kavail.avail; time = kavail.tstamp.pcm_io_frames / kavail.tstamp.sampling_rate; tstamp->tv_sec = time; @@ -335,7 +336,7 @@ int compress_get_tstamp(struct compress *compress, struct snd_compr_tstamp ktstamp; if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_TSTAMP, &ktstamp)) return oops(compress, errno, "cannot get tstamp"); @@ -355,9 +356,9 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size const unsigned int frag_size = compress->config->fragment_size; if (!(compress->flags & COMPRESS_IN)) - return oops(compress, -EINVAL, "Invalid flag set"); + return oops(compress, EINVAL, "Invalid flag set"); if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); fds.fd = compress->fd; fds.events = POLLOUT; @@ -383,7 +384,7 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size continue; } if (fds.revents & POLLERR) { - return oops(compress, -EIO, "poll returned error!"); + return oops(compress, EIO, "poll returned error!"); } } /* write avail bytes */ @@ -415,9 +416,9 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) const unsigned int frag_size = compress->config->fragment_size; if (!(compress->flags & COMPRESS_OUT)) - return oops(compress, -EINVAL, "Invalid flag set"); + return oops(compress, EINVAL, "Invalid flag set"); if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); fds.fd = compress->fd; fds.events = POLLIN; @@ -440,7 +441,7 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) continue; } if (fds.revents & POLLERR) { - return oops(compress, -EIO, "poll returned error!"); + return oops(compress, EIO, "poll returned error!"); } } /* read avail bytes */ @@ -466,7 +467,7 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) int compress_start(struct compress *compress) { if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_START)) return oops(compress, errno, "cannot start the stream"); compress->running = 1; @@ -477,7 +478,7 @@ int compress_start(struct compress *compress) int compress_stop(struct compress *compress) { if (!is_compress_running(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_STOP)) return oops(compress, errno, "cannot stop the stream"); return 0; @@ -486,7 +487,7 @@ int compress_stop(struct compress *compress) int compress_pause(struct compress *compress) { if (!is_compress_running(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_PAUSE)) return oops(compress, errno, "cannot pause the stream"); return 0; @@ -502,7 +503,7 @@ int compress_resume(struct compress *compress) int compress_drain(struct compress *compress) { if (!is_compress_running(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_DRAIN)) return oops(compress, errno, "cannot drain the stream"); return 0; @@ -511,10 +512,10 @@ int compress_drain(struct compress *compress) int compress_partial_drain(struct compress *compress) { if (!is_compress_running(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (!compress->next_track) - return oops(compress, -EPERM, "next track not signalled"); + return oops(compress, EPERM, "next track not signalled"); if (ioctl(compress->fd, SNDRV_COMPRESS_PARTIAL_DRAIN)) return oops(compress, errno, "cannot drain the stream\n"); compress->next_track = 0; @@ -524,10 +525,10 @@ int compress_partial_drain(struct compress *compress) int compress_next_track(struct compress *compress) { if (!is_compress_running(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (!compress->gapless_metadata) - return oops(compress, -EPERM, "metadata not set"); + return oops(compress, EPERM, "metadata not set"); if (ioctl(compress->fd, SNDRV_COMPRESS_NEXT_TRACK)) return oops(compress, errno, "cannot set next track\n"); compress->next_track = 1; @@ -542,14 +543,14 @@ int compress_set_gapless_metadata(struct compress *compress, int version; if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); version = get_compress_version(compress); if (version <= 0) return -1; if (version < SNDRV_PROTOCOL_VERSION(0, 1, 1)) - return oops(compress, -ENXIO, "gapless apis not supported in kernel"); + return oops(compress, ENXIO, "gapless apis not supported in kernel"); metadata.key = SNDRV_COMPRESS_ENCODER_PADDING; metadata.value[0] = mdata->encoder_padding; -- 1.7.0.4

1 1

[alsa-devel] How to specify data format if playback/capture were different ?
by Kuninori Morimoto 25 May '13

25 May '13

Hi Mark, and ALSA ML Now, I'm creating ak4554 AD/DA. My question is about data format. If my understand is correct, we can specify cpu/codec deivce data format by using snd_soc_dai_set_fmt(dai, SND_SOC_DAIFMT_xxx); But, ak4554 case, data formats are playback : SND_SOC_DAIFMT_RIGHT_J capture : SND_SOC_DAIFMT_LEFT_J and, it can't exchange this data format. CPU driver should set these information for ak4554 when playback/capture. Then, how to specify these data format on ALSA SoC ? Best regards --- Kuninori Morimoto

3 8

[alsa-devel] [PATCH 0/6] More code cleanup for ASoC ux500
by Fabio Baltieri 25 May '13

25 May '13

Hi Mark, this series contains more various code cleanup patches that I put together while working on other stuff this drivers. The first patch is actually a resend, as Linus Walleij agreed to drop that code in the original thread (https://lkml.org/lkml/2013/5/17/487). The others are just a bunch of sparse fixes and generic code cleanup and dead code removal. Thanks, Fabio Fabio Baltieri (6): ASoC: ux500: Drop pinctrl sleep support ASoC: ab8500-codec: Move codec ops on a separate structure ASoC: ux500: Drop dangling struct i2s_controller ASoC: ux500: Drop unused code from msp headers ASoC: ux500: Add missing mop500_ab8500.h include ASoC: ux500: Drop redundant msp id enumerations sound/soc/codecs/ab8500-codec.c | 19 ++++------- sound/soc/ux500/mop500_ab8500.c | 1 + sound/soc/ux500/ux500_msp_dai.h | 2 -- sound/soc/ux500/ux500_msp_i2s.c | 75 ++--------------------------------------- sound/soc/ux500/ux500_msp_i2s.h | 59 ++------------------------------ 5 files changed, 12 insertions(+), 144 deletions(-) -- 1.8.2

3 9

[alsa-devel] hdmi output (switching sample format/rates)
by Sander Jansen 25 May '13

25 May '13

In my audio application when I sometimes need to switch to a different sample format and/or rate, I usually leave the audio device open and merely reconfigure the device to the new parameters. This works with all the regular pcm devices, but it fails when I use the "hdmi" device. It seemed to just silently fail and no audio output is heard. As far as I know, no error condition is triggered (but I'll admit, my knowledge of the alsa api is rather limited). My question, Is this a bug or is it supposed to work this way and if so are there ways to detect whether a devices needs to be reopened or not. Thanks, Sander

2 2

[alsa-devel] [PATCH v2] ASoC: codecs: adau1701: add DT bindings
by Daniel Mack 25 May '13

25 May '13

Apart from pure matching, the bindings also support setting the the reset gpio line. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Lars, thanks for the quick review. Here's v2 of that patch. .../devicetree/bindings/sound/adi,adau1701.txt | 23 ++++++++++++++ sound/soc/codecs/adau1701.c | 35 +++++++++++++++++++++- 2 files changed, 57 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1701.txt diff --git a/Documentation/devicetree/bindings/sound/adi,adau1701.txt b/Documentation/devicetree/bindings/sound/adi,adau1701.txt new file mode 100644 index 0000000..3afeda7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1701.txt @@ -0,0 +1,23 @@ +Analog Devices ADAU1701 + +Required properties: + + - compatible: Should contain "adi,adau1701" + - reg: The i2c address. Value depends on the state of ADDR0 + and ADDR1, as wired in hardware. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + +Examples: + + i2c_bus { + adau1701@34 { + compatible = "adi,adau1701"; + reg = <0x34>; + reset-gpio = <&gpio 23 0>; + }; + }; diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 95e1677..3fc1763 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -13,6 +13,9 @@ #include <linux/i2c.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -452,6 +455,14 @@ static struct snd_soc_dai_driver adau1701_dai = { .symmetric_rates = 1, }; +#ifdef CONFIG_OF +static const struct of_device_id adau1701_dt_ids[] = { + { .compatible = "adi,adau1701", }, + { } +}; +MODULE_DEVICE_TABLE(of, adau1701_dt_ids); +#endif + static int adau1701_probe(struct snd_soc_codec *codec) { int ret; @@ -494,12 +505,33 @@ static int adau1701_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct adau1701 *adau1701; + struct device *dev = &client->dev; + int gpio_nreset = -EINVAL; int ret; - adau1701 = devm_kzalloc(&client->dev, sizeof(*adau1701), GFP_KERNEL); + adau1701 = devm_kzalloc(dev, sizeof(*adau1701), GFP_KERNEL); if (!adau1701) return -ENOMEM; + if (dev->of_node) { + gpio_nreset = of_get_named_gpio(dev->of_node, "reset-gpio", 0); + if (gpio_nreset < 0 && gpio_nreset != -ENOENT) + return gpio_nreset; + } + + if (gpio_is_valid(gpio_nreset)) { + ret = devm_gpio_request_one(dev, gpio_nreset, GPIOF_OUT_INIT_LOW, + "ADAU1701 Reset"); + if (ret < 0) + return ret; + + /* minimum reset time is 20ns */ + udelay(1); + gpio_set_value(gpio_nreset, 1); + /* power-up time may be as long as 85ms */ + mdelay(85); + } + i2c_set_clientdata(client, adau1701); ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, &adau1701_dai, 1); @@ -522,6 +554,7 @@ static struct i2c_driver adau1701_i2c_driver = { .driver = { .name = "adau1701", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(adau1701_dt_ids), }, .probe = adau1701_i2c_probe, .remove = adau1701_i2c_remove, -- 1.8.1.4

3 2

[alsa-devel] [PATCH v2] ASoC: Add ssm2518 support
by Lars-Peter Clausen 25 May '13

25 May '13

This patch adds a ASoC CODEC driver for the SSM2516. The SSM2516 is a stereo Class-D audio amplifier with an I2S interface for audio in and a built-in dynamic range control processor. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Power off the codec in the I2C probe function * Allow a sysclk of 0 which won't but any constraints on the supported sample rates. This can be used by systems which are able to change the sysclk on demand. * Remove {} around multi-line single statements --- .../devicetree/bindings/sound/ssm2518.txt | 20 + include/linux/platform_data/ssm2518.h | 22 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2518.c | 856 +++++++++++++++++++++ sound/soc/codecs/ssm2518.h | 20 + 6 files changed, 924 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ssm2518.txt create mode 100644 include/linux/platform_data/ssm2518.h create mode 100644 sound/soc/codecs/ssm2518.c create mode 100644 sound/soc/codecs/ssm2518.h diff --git a/Documentation/devicetree/bindings/sound/ssm2518.txt b/Documentation/devicetree/bindings/sound/ssm2518.txt new file mode 100644 index 0000000..59381a7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ssm2518.txt @@ -0,0 +1,20 @@ +SSM2518 audio amplifier + +This device supports I2C only. + +Required properties: + - compatible : Must be "adi,ssm2518" + - reg : the I2C address of the device. This will either be 0x34 (ADDR pin low) + or 0x35 (ADDR pin high) + +Optional properties: + - gpios : GPIO connected to the nSD pin. If the property is not present it is + assumed that the nSD pin is hardwired to always on. + +Example: + + ssm2518: ssm2518@34 { + compatible = "adi,ssm2518"; + reg = <0x34>; + gpios = <&gpio 5 0>; + }; diff --git a/include/linux/platform_data/ssm2518.h b/include/linux/platform_data/ssm2518.h new file mode 100644 index 0000000..9a8e3ea --- /dev/null +++ b/include/linux/platform_data/ssm2518.h @@ -0,0 +1,22 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __LINUX_PLATFORM_DATA_SSM2518_H__ +#define __LINUX_PLATFORM_DATA_SSM2518_H__ + +/** + * struct ssm2518_platform_data - Platform data for the ssm2518 driver + * @enable_gpio: GPIO connected to the nSD pin. Set to -1 if the nSD pin is + * hardwired. + */ +struct ssm2518_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..d76609a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF + select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C @@ -313,6 +314,9 @@ config SND_SOC_SN95031 config SND_SOC_SPDIF tristate +config SND_SOC_SSM2518 + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dae0aa6..7204c24 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -52,6 +52,7 @@ snd-soc-si476x-objs := si476x.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o @@ -176,6 +177,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c new file mode 100644 index 0000000..3139a1b --- /dev/null +++ b/sound/soc/codecs/ssm2518.c @@ -0,0 +1,856 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/platform_data/ssm2518.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ssm2518.h" + +#define SSM2518_REG_POWER1 0x00 +#define SSM2518_REG_CLOCK 0x01 +#define SSM2518_REG_SAI_CTRL1 0x02 +#define SSM2518_REG_SAI_CTRL2 0x03 +#define SSM2518_REG_CHAN_MAP 0x04 +#define SSM2518_REG_LEFT_VOL 0x05 +#define SSM2518_REG_RIGHT_VOL 0x06 +#define SSM2518_REG_MUTE_CTRL 0x07 +#define SSM2518_REG_FAULT_CTRL 0x08 +#define SSM2518_REG_POWER2 0x09 +#define SSM2518_REG_DRC_1 0x0a +#define SSM2518_REG_DRC_2 0x0b +#define SSM2518_REG_DRC_3 0x0c +#define SSM2518_REG_DRC_4 0x0d +#define SSM2518_REG_DRC_5 0x0e +#define SSM2518_REG_DRC_6 0x0f +#define SSM2518_REG_DRC_7 0x10 +#define SSM2518_REG_DRC_8 0x11 +#define SSM2518_REG_DRC_9 0x12 + +#define SSM2518_POWER1_RESET BIT(7) +#define SSM2518_POWER1_NO_BCLK BIT(5) +#define SSM2518_POWER1_MCS_MASK (0xf << 1) +#define SSM2518_POWER1_MCS_64FS (0x0 << 1) +#define SSM2518_POWER1_MCS_128FS (0x1 << 1) +#define SSM2518_POWER1_MCS_256FS (0x2 << 1) +#define SSM2518_POWER1_MCS_384FS (0x3 << 1) +#define SSM2518_POWER1_MCS_512FS (0x4 << 1) +#define SSM2518_POWER1_MCS_768FS (0x5 << 1) +#define SSM2518_POWER1_MCS_100FS (0x6 << 1) +#define SSM2518_POWER1_MCS_200FS (0x7 << 1) +#define SSM2518_POWER1_MCS_400FS (0x8 << 1) +#define SSM2518_POWER1_SPWDN BIT(0) + +#define SSM2518_CLOCK_ASR BIT(0) + +#define SSM2518_SAI_CTRL1_FMT_MASK (0x3 << 5) +#define SSM2518_SAI_CTRL1_FMT_I2S (0x0 << 5) +#define SSM2518_SAI_CTRL1_FMT_LJ (0x1 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_24BIT (0x2 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_16BIT (0x3 << 5) + +#define SSM2518_SAI_CTRL1_SAI_MASK (0x7 << 2) +#define SSM2518_SAI_CTRL1_SAI_I2S (0x0 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_2 (0x1 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_4 (0x2 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_8 (0x3 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_16 (0x4 << 2) +#define SSM2518_SAI_CTRL1_SAI_MONO (0x5 << 2) + +#define SSM2518_SAI_CTRL1_FS_MASK (0x3) +#define SSM2518_SAI_CTRL1_FS_8000_12000 (0x0) +#define SSM2518_SAI_CTRL1_FS_16000_24000 (0x1) +#define SSM2518_SAI_CTRL1_FS_32000_48000 (0x2) +#define SSM2518_SAI_CTRL1_FS_64000_96000 (0x3) + +#define SSM2518_SAI_CTRL2_BCLK_INTERAL BIT(7) +#define SSM2518_SAI_CTRL2_LRCLK_PULSE BIT(6) +#define SSM2518_SAI_CTRL2_LRCLK_INVERT BIT(5) +#define SSM2518_SAI_CTRL2_MSB BIT(4) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK (0x3 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_32 (0x0 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_24 (0x1 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_16 (0x2 << 2) +#define SSM2518_SAI_CTRL2_BCLK_INVERT BIT(1) + +#define SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET 4 +#define SSM2518_CHAN_MAP_RIGHT_SLOT_MASK 0xf0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET 0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_MASK 0x0f + +#define SSM2518_MUTE_CTRL_ANA_GAIN BIT(5) +#define SSM2518_MUTE_CTRL_MUTE_MASTER BIT(0) + +#define SSM2518_POWER2_APWDN BIT(0) + +#define SSM2518_DAC_MUTE BIT(6) +#define SSM2518_DAC_FS_MASK 0x07 +#define SSM2518_DAC_FS_8000 0x00 +#define SSM2518_DAC_FS_16000 0x01 +#define SSM2518_DAC_FS_32000 0x02 +#define SSM2518_DAC_FS_64000 0x03 +#define SSM2518_DAC_FS_128000 0x04 + +struct ssm2518 { + struct regmap *regmap; + bool right_j; + + unsigned int sysclk; + const struct snd_pcm_hw_constraint_list *constraints; + + int enable_gpio; +}; + +static const struct reg_default ssm2518_reg_defaults[] = { + { 0x00, 0x05 }, + { 0x01, 0x00 }, + { 0x02, 0x02 }, + { 0x03, 0x00 }, + { 0x04, 0x10 }, + { 0x05, 0x40 }, + { 0x06, 0x40 }, + { 0x07, 0x81 }, + { 0x08, 0x0c }, + { 0x09, 0x99 }, + { 0x0a, 0x7c }, + { 0x0b, 0x5b }, + { 0x0c, 0x57 }, + { 0x0d, 0x89 }, + { 0x0e, 0x8c }, + { 0x0f, 0x77 }, + { 0x10, 0x26 }, + { 0x11, 0x1c }, + { 0x12, 0x97 }, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(ssm2518_vol_tlv, -7125, 2400); +static const DECLARE_TLV_DB_SCALE(ssm2518_compressor_tlv, -3400, 200, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_expander_tlv, -8100, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_noise_gate_tlv, -9600, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_post_drc_tlv, -2400, 300, 0); + +static const DECLARE_TLV_DB_RANGE(ssm2518_limiter_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-2200, 200, 0), + 7, 15, TLV_DB_SCALE_ITEM(-800, 100, 0), +); + +static const char * const ssm2518_drc_peak_detector_attack_time_text[] = { + "0 ms", "0.1 ms", "0.19 ms", "0.37 ms", "0.75 ms", "1.5 ms", "3 ms", + "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", "192 ms", "384 ms", + "768 ms", "1536 ms", +}; + +static const char * const ssm2518_drc_peak_detector_release_time_text[] = { + "0 ms", "1.5 ms", "3 ms", "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", + "192 ms", "384 ms", "768 ms", "1536 ms", "3072 ms", "6144 ms", + "12288 ms", "24576 ms" +}; + +static const char * const ssm2518_drc_hold_time_text[] = { + "0 ms", "0.67 ms", "1.33 ms", "2.67 ms", "5.33 ms", "10.66 ms", + "21.32 ms", "42.64 ms", "85.28 ms", "170.56 ms", "341.12 ms", + "682.24 ms", "1364 ms", +}; + +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum, + SSM2518_REG_DRC_2, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum, + SSM2518_REG_DRC_2, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum, + SSM2518_REG_DRC_6, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum, + SSM2518_REG_DRC_6, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum, + SSM2518_REG_DRC_7, 4, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum, + SSM2518_REG_DRC_7, 0, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum, + SSM2518_REG_DRC_9, 0, ssm2518_drc_peak_detector_release_time_text); + +static const struct snd_kcontrol_new ssm2518_snd_controls[] = { + SOC_SINGLE("Playback De-emphasis Switch", SSM2518_REG_MUTE_CTRL, + 4, 1, 0), + SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2518_REG_LEFT_VOL, + SSM2518_REG_RIGHT_VOL, 0, 0xff, 1, ssm2518_vol_tlv), + SOC_DOUBLE("Master Playback Switch", SSM2518_REG_MUTE_CTRL, 2, 1, 1, 1), + + SOC_SINGLE("Amp Low Power Mode Switch", SSM2518_REG_POWER2, 4, 1, 0), + SOC_SINGLE("DAC Low Power Mode Switch", SSM2518_REG_POWER2, 3, 1, 0), + + SOC_SINGLE("DRC Limiter Switch", SSM2518_REG_DRC_1, 5, 1, 0), + SOC_SINGLE("DRC Compressor Switch", SSM2518_REG_DRC_1, 4, 1, 0), + SOC_SINGLE("DRC Expander Switch", SSM2518_REG_DRC_1, 3, 1, 0), + SOC_SINGLE("DRC Noise Gate Switch", SSM2518_REG_DRC_1, 2, 1, 0), + SOC_DOUBLE("DRC Switch", SSM2518_REG_DRC_1, 0, 1, 1, 0), + + SOC_SINGLE_TLV("DRC Limiter Threshold Volume", + SSM2518_REG_DRC_3, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Compressor Lower Threshold Volume", + SSM2518_REG_DRC_3, 0, 15, 1, ssm2518_compressor_tlv), + SOC_SINGLE_TLV("DRC Expander Upper Threshold Volume", SSM2518_REG_DRC_4, + 4, 15, 1, ssm2518_expander_tlv), + SOC_SINGLE_TLV("DRC Noise Gate Threshold Volume", + SSM2518_REG_DRC_4, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Upper Output Threshold Volume", + SSM2518_REG_DRC_5, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Lower Output Threshold Volume", + SSM2518_REG_DRC_5, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Post Volume", SSM2518_REG_DRC_8, + 2, 15, 1, ssm2518_post_drc_tlv), + + SOC_ENUM("DRC Peak Detector Attack Time", + ssm2518_drc_peak_detector_attack_time_enum), + SOC_ENUM("DRC Peak Detector Release Time", + ssm2518_drc_peak_detector_release_time_enum), + SOC_ENUM("DRC Attack Time", ssm2518_drc_attack_time_enum), + SOC_ENUM("DRC Decay Time", ssm2518_drc_decay_time_enum), + SOC_ENUM("DRC Hold Time", ssm2518_drc_hold_time_enum), + SOC_ENUM("DRC Noise Gate Hold Time", + ssm2518_drc_noise_gate_hold_time_enum), + SOC_ENUM("DRC RMS Averaging Time", ssm2518_drc_rms_averaging_time_enum), +}; + +static const struct snd_soc_dapm_widget ssm2518_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", SSM2518_REG_POWER2, 1, 1), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", SSM2518_REG_POWER2, 2, 1), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route ssm2518_routes[] = { + { "OUTL", NULL, "DACL" }, + { "OUTR", NULL, "DACR" }, +}; + +struct ssm2518_mcs_lut { + unsigned int rate; + const unsigned int *sysclks; +}; + +static const unsigned int ssm2518_sysclks_2048000[] = { + 2048000, 4096000, 8192000, 12288000, 16384000, 24576000, + 3200000, 6400000, 12800000, 0 +}; + +static const unsigned int ssm2518_sysclks_2822000[] = { + 2822000, 5644800, 11289600, 16934400, 22579200, 33868800, + 4410000, 8820000, 17640000, 0 +}; + +static const unsigned int ssm2518_sysclks_3072000[] = { + 3072000, 6144000, 12288000, 16384000, 24576000, 38864000, + 4800000, 9600000, 19200000, 0 +}; + +static const struct ssm2518_mcs_lut ssm2518_mcs_lut[] = { + { 8000, ssm2518_sysclks_2048000, }, + { 11025, ssm2518_sysclks_2822000, }, + { 12000, ssm2518_sysclks_3072000, }, + { 16000, ssm2518_sysclks_2048000, }, + { 24000, ssm2518_sysclks_3072000, }, + { 22050, ssm2518_sysclks_2822000, }, + { 32000, ssm2518_sysclks_2048000, }, + { 44100, ssm2518_sysclks_2822000, }, + { 48000, ssm2518_sysclks_3072000, }, + { 96000, ssm2518_sysclks_3072000, }, +}; + +static const unsigned int ssm2518_rates_2048000[] = { + 8000, 16000, 32000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2048000 = { + .list = ssm2518_rates_2048000, + .count = ARRAY_SIZE(ssm2518_rates_2048000), +}; + +static const unsigned int ssm2518_rates_2822000[] = { + 11025, 22050, 44100, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2822000 = { + .list = ssm2518_rates_2822000, + .count = ARRAY_SIZE(ssm2518_rates_2822000), +}; + +static const unsigned int ssm2518_rates_3072000[] = { + 12000, 24000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_3072000 = { + .list = ssm2518_rates_3072000, + .count = ARRAY_SIZE(ssm2518_rates_3072000), +}; + +static const unsigned int ssm2518_rates_12288000[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_12288000 = { + .list = ssm2518_rates_12288000, + .count = ARRAY_SIZE(ssm2518_rates_12288000), +}; + +static unsigned int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, + unsigned int rate) +{ + const unsigned int *sysclks = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(ssm2518_mcs_lut); i++) { + if (ssm2518_mcs_lut[i].rate == rate) { + sysclks = ssm2518_mcs_lut[i].sysclks; + break; + } + } + + if (!sysclks) + return -EINVAL; + + for (i = 0; sysclks[i]; i++) { + if (sysclks[i] == ssm2518->sysclk) + return i; + } + + return -EINVAL; +} + +static int ssm2518_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int ctrl1, ctrl1_mask; + int mcs; + int ret; + + mcs = ssm2518_lookup_mcs(ssm2518, rate); + if (mcs < 0) + return mcs; + + ctrl1_mask = SSM2518_SAI_CTRL1_FS_MASK; + + if (rate >= 8000 && rate <= 12000) + ctrl1 = SSM2518_SAI_CTRL1_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + ctrl1 = SSM2518_SAI_CTRL1_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + ctrl1 = SSM2518_SAI_CTRL1_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + ctrl1 = SSM2518_SAI_CTRL1_FS_64000_96000; + else + return -EINVAL; + + if (ssm2518->right_j) { + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_16BIT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + break; + default: + return -EINVAL; + } + ctrl1_mask |= SSM2518_SAI_CTRL1_FMT_MASK; + } + + /* Disable auto samplerate detection */ + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_CLOCK, + SSM2518_CLOCK_ASR, SSM2518_CLOCK_ASR); + if (ret < 0) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + ctrl1_mask, ctrl1); + if (ret < 0) + return ret; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_MCS_MASK, mcs << 1); +} + +static int ssm2518_mute(struct snd_soc_dai *dai, int mute) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (mute) + val = SSM2518_MUTE_CTRL_MUTE_MASTER; + else + val = 0; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_MUTE_CTRL, + SSM2518_MUTE_CTRL_MUTE_MASTER, val); +} + +static int ssm2518_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1 = 0, ctrl2 = 0; + bool invert_fclk; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_fclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_fclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = true; + break; + default: + return -EINVAL; + } + + ssm2518->right_j = false; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + ssm2518->right_j = true; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = false; + break; + default: + return -EINVAL; + } + + if (invert_fclk) + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_INVERT; + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, ctrl1); + if (ret) + return ret; + + return regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, ctrl2); +} + +static int ssm2518_set_power(struct ssm2518 *ssm2518, bool enable) +{ + int ret = 0; + + if (!enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN, SSM2518_POWER1_SPWDN); + regcache_mark_dirty(ssm2518->regmap); + } + + if (gpio_is_valid(ssm2518->enable_gpio)) + gpio_set_value(ssm2518->enable_gpio, enable); + + regcache_cache_only(ssm2518->regmap, !enable); + + if (enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN | SSM2518_POWER1_RESET, 0x00); + regcache_sync(ssm2518->regmap); + } + + return ret; +} + +static int ssm2518_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = ssm2518_set_power(ssm2518, true); + break; + case SND_SOC_BIAS_OFF: + ret = ssm2518_set_power(ssm2518, false); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + + return 0; +} + +static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1, ctrl2; + int left_slot, right_slot; + int ret; + + if (slots == 0) + return regmap_update_bits(ssm2518->regmap, + SSM2518_REG_SAI_CTRL1, SSM2518_SAI_CTRL1_SAI_MASK, + SSM2518_SAI_CTRL1_SAI_I2S); + + if (tx_mask == 0 || tx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + /* We assume the left channel < right channel */ + left_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + switch (width) { + case 16: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_16; + break; + case 24: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_24; + break; + case 32: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_32; + break; + default: + return -EINVAL; + } + + switch (slots) { + case 1: + ctrl1 = SSM2518_SAI_CTRL1_SAI_MONO; + break; + case 2: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_2; + break; + case 4: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_4; + break; + case 8: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_8; + break; + case 16: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_16; + break; + default: + return -EINVAL; + } + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_CHAN_MAP, + (left_slot << SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET) | + (right_slot << SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + SSM2518_SAI_CTRL1_SAI_MASK, ctrl1); + if (ret) + return ret; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, + SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK, ctrl2); +} + +static int ssm2518_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + + if (ssm2518->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, ssm2518->constraints); + + return 0; +} + +#define SSM2518_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ssm2518_dai_ops = { + .startup = ssm2518_startup, + .hw_params = ssm2518_hw_params, + .digital_mute = ssm2518_mute, + .set_fmt = ssm2518_set_dai_fmt, + .set_tdm_slot = ssm2518_set_tdm_slot, +}; + +static struct snd_soc_dai_driver ssm2518_dai = { + .name = "ssm2518-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SSM2518_FORMATS, + }, + .ops = &ssm2518_dai_ops, +}; + +static int ssm2518_probe(struct snd_soc_codec *codec) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = ssm2518->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int ssm2518_remove(struct snd_soc_codec *codec) +{ + ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (clk_id != SSM2518_SYSCLK) + return -EINVAL; + + switch (source) { + case SSM2518_SYSCLK_SRC_MCLK: + val = 0; + break; + case SSM2518_SYSCLK_SRC_BCLK: + /* In this case the bitclock is used as the system clock, and + * the bitclock signal needs to be connected to the MCLK pin and + * the BCLK pin is left unconnected */ + val = SSM2518_POWER1_NO_BCLK; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 0: + ssm2518->constraints = NULL; + break; + case 2048000: + case 4096000: + case 8192000: + case 3200000: + case 6400000: + case 12800000: + ssm2518->constraints = &ssm2518_constraints_2048000; + break; + case 2822000: + case 5644800: + case 11289600: + case 16934400: + case 22579200: + case 33868800: + case 4410000: + case 8820000: + case 17640000: + ssm2518->constraints = &ssm2518_constraints_2822000; + break; + case 3072000: + case 6144000: + case 38864000: + case 4800000: + case 9600000: + case 19200000: + ssm2518->constraints = &ssm2518_constraints_3072000; + break; + case 12288000: + case 16384000: + case 24576000: + ssm2518->constraints = &ssm2518_constraints_12288000; + break; + default: + return -EINVAL; + } + + ssm2518->sysclk = freq; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_NO_BCLK, val); +} + +static struct snd_soc_codec_driver ssm2518_codec_driver = { + .probe = ssm2518_probe, + .remove = ssm2518_remove, + .set_bias_level = ssm2518_set_bias_level, + .set_sysclk = ssm2518_set_sysclk, + .idle_bias_off = true, + + .controls = ssm2518_snd_controls, + .num_controls = ARRAY_SIZE(ssm2518_snd_controls), + .dapm_widgets = ssm2518_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2518_dapm_widgets), + .dapm_routes = ssm2518_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2518_routes), +}; + +static bool ssm2518_register_volatile(struct device *dev, unsigned int reg) +{ + return false; +} + +static const struct regmap_config ssm2518_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = SSM2518_REG_DRC_9, + .volatile_reg = ssm2518_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ssm2518_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ssm2518_reg_defaults), +}; + +static int ssm2518_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm2518_platform_data *pdata = i2c->dev.platform_data; + struct ssm2518 *ssm2518; + int ret; + + ssm2518 = devm_kzalloc(&i2c->dev, sizeof(*ssm2518), GFP_KERNEL); + if (ssm2518 == NULL) + return -ENOMEM; + + if (pdata) { + ssm2518->enable_gpio = pdata->enable_gpio; + } else if (i2c->dev.of_node) { + ssm2518->enable_gpio = of_get_gpio(i2c->dev.of_node, 0); + if (ssm2518->enable_gpio < 0 && ssm2518->enable_gpio != -ENOENT) + return ssm2518->enable_gpio; + } else { + ssm2518->enable_gpio = -1; + } + + if (gpio_is_valid(ssm2518->enable_gpio)) { + ret = devm_gpio_request_one(&i2c->dev, ssm2518->enable_gpio, + GPIOF_OUT_INIT_HIGH, "SSM2518 nSD"); + if (ret) + return ret; + } + + i2c_set_clientdata(i2c, ssm2518); + + ssm2518->regmap = devm_regmap_init_i2c(i2c, &ssm2518_regmap_config); + if (IS_ERR(ssm2518->regmap)) + return PTR_ERR(ssm2518->regmap); + + /* + * The reset bit is obviously volatile, but we need to be able to cache + * the other bits in the register, so we can't just mark the whole + * register as volatile. Since this is the only place where we'll ever + * touch the reset bit just bypass the cache for this operation. + */ + regcache_cache_bypass(ssm2518->regmap, true); + ret = regmap_write(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_RESET); + regcache_cache_bypass(ssm2518->regmap, false); + if (ret) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER2, + SSM2518_POWER2_APWDN, 0x00); + if (ret) + return ret; + + ret = ssm2518_set_power(ssm2518, false); + if (ret) + return ret; + + return snd_soc_register_codec(&i2c->dev, &ssm2518_codec_driver, + &ssm2518_dai, 1); +} + +static int ssm2518_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm2518_i2c_ids[] = { + { "ssm2518", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids); + +static struct i2c_driver ssm2518_driver = { + .driver = { + .name = "ssm2518", + .owner = THIS_MODULE, + }, + .probe = ssm2518_i2c_probe, + .remove = ssm2518_i2c_remove, + .id_table = ssm2518_i2c_ids, +}; +module_i2c_driver(ssm2518_driver); + +MODULE_DESCRIPTION("ASoC SSM2518 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2518.h b/sound/soc/codecs/ssm2518.h new file mode 100644 index 0000000..62511d8 --- /dev/null +++ b/sound/soc/codecs/ssm2518.h @@ -0,0 +1,20 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __SND_SOC_CODECS_SSM2518_H__ +#define __SND_SOC_CODECS_SSM2518_H__ + +#define SSM2518_SYSCLK 0 + +enum ssm2518_sysclk_src { + SSM2518_SYSCLK_SRC_MCLK = 0, + SSM2518_SYSCLK_SRC_BCLK = 1, +}; + +#endif -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: wm5110: Correct DSP4R Mixer control name
by Charles Keepax 25 May '13

25 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c00480b..ba38f06 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), -- 1.7.2.5

2 1

[alsa-devel] Handling HDA FIFO error
by Daniel Drake 25 May '13

25 May '13

Hi, On the OLPC XO-1.5 (VIA VX855 SoC with HDA) we sometimes see a problem where audio playback fails. Easy to reproduce by playing a short wav file in a loop, it will fail after a minute or two. The playback app gets confused here: snd_pcm_wait() returns success, but snd_pcm_writei() returns EAGAIN. Digging into the HDA code, I can see that normally, when the stream gets started, we get interrupts with SD_STS as 0x24. In the failure case, as soon as we start the stream, we get a flood of interrupts with SD_STS 0x8 - which indicates FIFO error. Disabling interrupt generation upon FIFO error does not help much - it avoids the flood of interrupts, but doesn't change the fact that audio playing apps hang or loop infinitely. I have tried the various position_fix options with no luck. Are there any things I could check that would explain why a FIFO error might be occurring? After a quick read of the HDA spec I don't really understand what this FIFO actually is and what it does, how does it relate to the BDL, CORB and RIRB? Failing that... any suggestions for how to improve the hda_intel driver to handle FIFO error gracefully? Right now there is no code to handle that. Thanks Daniel

3 3

[alsa-devel] [PATCH] ASoC: Add support for BCM2708
by Florian Meier 24 May '13

24 May '13

This driver adds support for digital audio (I2S) for the BCM2708 SoC that is used by the Raspberry Pi. External audio codecs can be connected to the Raspberry Pi via P5 header. It relies on cyclic DMA engine support for BCM2708 that is not included in this patch. The currently supported audio codecs are TLV320AIC23 and WM8731. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> --- A Raspberry Pi kernel featuring this driver can be found at https://github.com/koalo/linux rpi-3.8.y-asocdev You can find more information about how to use it at http://blog.koalo.de/2013/05/i2s-support-for-raspberry-pi.html sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/bcm2708/Kconfig | 29 ++ sound/soc/bcm2708/Makefile | 12 + sound/soc/bcm2708/bcm2708-i2s.c | 964 +++++++++++++++++++++++++++++++++++++++ sound/soc/bcm2708/bcm2708-pcm.c | 303 ++++++++++++ sound/soc/bcm2708/bcm2708-pcm.h | 40 ++ sound/soc/bcm2708/rpi-mbed.c | 113 +++++ sound/soc/bcm2708/rpi-proto.c | 131 ++++++ 9 files changed, 1594 insertions(+) create mode 100644 sound/soc/bcm2708/Kconfig create mode 100644 sound/soc/bcm2708/Makefile create mode 100644 sound/soc/bcm2708/bcm2708-i2s.c create mode 100644 sound/soc/bcm2708/bcm2708-pcm.c create mode 100644 sound/soc/bcm2708/bcm2708-pcm.h create mode 100644 sound/soc/bcm2708/rpi-mbed.c create mode 100644 sound/soc/bcm2708/rpi-proto.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 9e675c7..d3852f8 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM # All the supported SoCs source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" +source "sound/soc/bcm2708/Kconfig" source "sound/soc/blackfin/Kconfig" source "sound/soc/cirrus/Kconfig" source "sound/soc/davinci/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 197b6ae..eef1a7b 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -14,6 +14,7 @@ obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ +obj-$(CONFIG_SND_SOC) += bcm2708/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += cirrus/ obj-$(CONFIG_SND_SOC) += davinci/ diff --git a/sound/soc/bcm2708/Kconfig b/sound/soc/bcm2708/Kconfig new file mode 100644 index 0000000..f47c2c0 --- /dev/null +++ b/sound/soc/bcm2708/Kconfig @@ -0,0 +1,29 @@ +config SND_BCM2708_SOC_I2S + tristate + +config SND_BCM2708_SOC + tristate "SoC Audio support for the Broadcom BCM2708 I2S module" + depends on MACH_BCM2708 + select SND_SOC_DMAENGINE_PCM + select DMA_BCM2708 + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for codecs attached to + the BCM2708 I2S interface. You will also need + to select the audio interfaces to support below. + +config SND_BCM2708_SOC_RPI_CODEC_MBED + tristate "Support for AudioCODEC for mbed (TLV320AIC32B)" + depends on SND_BCM2708_SOC + select SND_BCM2708_SOC_I2S + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for AudioCODEC for mbed (TLV320AIC32B) + +config SND_BCM2708_SOC_RPI_CODEC_PROTO + tristate "Support for Audio Codec Board - PROTO (WM8731)" + depends on SND_BCM2708_SOC + select SND_BCM2708_SOC_I2S + select SND_SOC_WM8731 + help + Say Y if you want to add support for Audio Codec Board - PROTO (WM8731) diff --git a/sound/soc/bcm2708/Makefile b/sound/soc/bcm2708/Makefile new file mode 100644 index 0000000..284d76d --- /dev/null +++ b/sound/soc/bcm2708/Makefile @@ -0,0 +1,12 @@ +# BCM2708 Platform Support +snd-soc-bcm2708-objs := bcm2708-pcm.o +snd-soc-bcm2708-i2s-objs := bcm2708-i2s.o +snd-soc-rpi-mbed-objs := rpi-mbed.o +snd-soc-rpi-proto-objs := rpi-proto.o + +obj-$(CONFIG_SND_BCM2708_SOC) += snd-soc-bcm2708.o +obj-$(CONFIG_SND_BCM2708_SOC_I2S) += snd-soc-bcm2708-i2s.o + +# BCM2708 Machine Support +obj-$(CONFIG_SND_BCM2708_SOC_RPI_CODEC_MBED) += snd-soc-rpi-mbed.o +obj-$(CONFIG_SND_BCM2708_SOC_RPI_CODEC_PROTO) += snd-soc-rpi-proto.o diff --git a/sound/soc/bcm2708/bcm2708-i2s.c b/sound/soc/bcm2708/bcm2708-i2s.c new file mode 100644 index 0000000..a8e995f --- /dev/null +++ b/sound/soc/bcm2708/bcm2708-i2s.c @@ -0,0 +1,964 @@ +/* + * ALSA SoC I2S Audio Layer for Broadcom BCM2708 SoC + * + * Author: Florian Meier, <florian.meier(a)koalo.de> + * Copyright 2013 + * + * based on + * Raspberry Pi PCM I2S ALSA Driver + * Copyright (c) by Phil Poole 2013 + * + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * Vladimir Barinov, <vbarinov(a)embeddedalley.com> + * Copyright (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> + * + * OMAP ALSA SoC DAI driver using McBSP port + * Copyright (C) 2008 Nokia Corporation + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Peter Ujfalusi <peter.ujfalusi(a)ti.com> + * + * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver + * Author: Timur Tabi <timur(a)freescale.com> + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "bcm2708-pcm.h" + +/* Clock registers */ +#define BCM2708_CLK_PCMCTL_REG 0x00 +#define BCM2708_CLK_PCMDIV_REG 0x04 + +/* Clock register settings */ +#define BCM2708_CLK_PASSWD (0x5a000000) +#define BCM2708_CLK_MASH(v) ((v) << 9) +#define BCM2708_CLK_FLIP (1 << 8) +#define BCM2708_CLK_BUSY (1 << 7) +#define BCM2708_CLK_KILL (1 << 5) +#define BCM2708_CLK_ENAB (1 << 4) +#define BCM2708_CLK_SRC(v) (v) + +#define BCM2708_CLK_DIVI(v) ((v) << 12) +#define BCM2708_CLK_DIVF(v) (v) + +enum { + BCM2708_CLK_MASH_0 = 0, + BCM2708_CLK_MASH_1, + BCM2708_CLK_MASH_2, + BCM2708_CLK_MASH_3, +}; + +enum { + BCM2708_CLK_SRC_GND = 0, + BCM2708_CLK_SRC_OSC, + BCM2708_CLK_SRC_DBG0, + BCM2708_CLK_SRC_DBG1, + BCM2708_CLK_SRC_PLLA, + BCM2708_CLK_SRC_PLLC, + BCM2708_CLK_SRC_PLLD, + BCM2708_CLK_SRC_HDMI, +}; + +/* Most clocks are not useable (freq = 0) */ +static const unsigned int bcm2708_clk_freq[BCM2708_CLK_SRC_HDMI+1] = { + [BCM2708_CLK_SRC_GND] = 0, + [BCM2708_CLK_SRC_OSC] = 19200000, + [BCM2708_CLK_SRC_DBG0] = 0, + [BCM2708_CLK_SRC_DBG1] = 0, + [BCM2708_CLK_SRC_PLLA] = 0, + [BCM2708_CLK_SRC_PLLC] = 0, + [BCM2708_CLK_SRC_PLLD] = 500000000, + [BCM2708_CLK_SRC_HDMI] = 0, +}; + + +/* I2S registers */ +#define BCM2708_I2S_CS_A_REG 0x00 +#define BCM2708_I2S_FIFO_A_REG 0x04 +#define BCM2708_I2S_MODE_A_REG 0x08 +#define BCM2708_I2S_RXC_A_REG 0x0c +#define BCM2708_I2S_TXC_A_REG 0x10 +#define BCM2708_I2S_DREQ_A_REG 0x14 +#define BCM2708_I2S_INTEN_A_REG 0x18 +#define BCM2708_I2S_INTSTC_A_REG 0x1c +#define BCM2708_I2S_GRAY_REG 0x20 + +/* I2S register settings */ +#define BCM2708_I2S_STBY (1 << 25) +#define BCM2708_I2S_SYNC (1 << 24) +#define BCM2708_I2S_RXSEX (1 << 23) +#define BCM2708_I2S_RXF (1 << 22) +#define BCM2708_I2S_TXE (1 << 21) +#define BCM2708_I2S_RXD (1 << 20) +#define BCM2708_I2S_TXD (1 << 19) +#define BCM2708_I2S_RXR (1 << 18) +#define BCM2708_I2S_TXW (1 << 17) +#define BCM2708_I2S_CS_RXERR (1 << 16) +#define BCM2708_I2S_CS_TXERR (1 << 15) +#define BCM2708_I2S_RXSYNC (1 << 14) +#define BCM2708_I2S_TXSYNC (1 << 13) +#define BCM2708_I2S_DMAEN (1 << 9) +#define BCM2708_I2S_RXTHR(v) ((v) << 7) +#define BCM2708_I2S_TXTHR(v) ((v) << 5) +#define BCM2708_I2S_RXCLR (1 << 4) +#define BCM2708_I2S_TXCLR (1 << 3) +#define BCM2708_I2S_TXON (1 << 2) +#define BCM2708_I2S_RXON (1 << 1) +#define BCM2708_I2S_EN (1) + +#define BCM2708_I2S_CLKDIS (1 << 28) +#define BCM2708_I2S_PDMN (1 << 27) +#define BCM2708_I2S_PDME (1 << 26) +#define BCM2708_I2S_FRXP (1 << 25) +#define BCM2708_I2S_FTXP (1 << 24) +#define BCM2708_I2S_CLKM (1 << 23) +#define BCM2708_I2S_CLKI (1 << 22) +#define BCM2708_I2S_FSM (1 << 21) +#define BCM2708_I2S_FSI (1 << 20) +#define BCM2708_I2S_FLEN(v) ((v) << 10) +#define BCM2708_I2S_FSLEN(v) (v) + +#define BCM2708_I2S_CHWEX (1 << 15) +#define BCM2708_I2S_CHEN (1 << 14) +#define BCM2708_I2S_CHPOS(v) ((v) << 4) +#define BCM2708_I2S_CHWID(v) (v) +#define BCM2708_I2S_CH1(v) ((v) << 16) +#define BCM2708_I2S_CH2(v) (v) + +#define BCM2708_I2S_TX_PANIC(v) ((v) << 24) +#define BCM2708_I2S_RX_PANIC(v) ((v) << 16) +#define BCM2708_I2S_TX(v) ((v) << 8) +#define BCM2708_I2S_RX(v) (v) + +#define BCM2708_I2S_INT_RXERR (1 << 3) +#define BCM2708_I2S_INT_TXERR (1 << 2) +#define BCM2708_I2S_INT_RXR (1 << 1) +#define BCM2708_I2S_INT_TXW (1 << 0) + +/* I2S DMA interface */ +#define BCM2708_I2S_FIFO_PHYSICAL_ADDR 0x7E203004 +#define BCM2708_DMA_DREQ_PCM_TX 2 +#define BCM2708_DMA_DREQ_PCM_RX 3 + +/* General device struct */ +struct bcm2708_i2s_dev { + struct device *dev; + struct bcm2708_pcm_dma_data dma_params[2]; + void __iomem *i2s_base; + void __iomem *clk_base; + unsigned int fmt; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; +}; + +static inline void bcm2708_i2s_write_reg(struct bcm2708_i2s_dev *dev, + int reg, u32 val) +{ + dev_dbg(dev->dev, "I2S write to register %p = %x\n", + dev->clk_base + reg, val); + __raw_writel(val, dev->i2s_base + reg); +} + +static inline u32 bcm2708_i2s_read_reg(struct bcm2708_i2s_dev *dev, int reg) +{ + return __raw_readl(dev->i2s_base + reg); +} + +static inline void bcm2708_i2s_set_bits(struct bcm2708_i2s_dev *dev, + int reg, u32 val) +{ + u32 oldval = __raw_readl(dev->i2s_base + reg); + bcm2708_i2s_write_reg(dev, reg, oldval | val); +} + +static inline void bcm2708_i2s_clear_bits(struct bcm2708_i2s_dev *dev, + int reg, u32 val) +{ + u32 oldval = __raw_readl(dev->i2s_base + reg); + bcm2708_i2s_write_reg(dev, reg, oldval & ~val); +} + +static inline void bcm2708_clk_write_reg(struct bcm2708_i2s_dev *dev, + int reg, u32 val) +{ + dev_dbg(dev->dev, "PCM clk write to register %p = %x\n", + dev->clk_base + reg, val); + __raw_writel(val, dev->clk_base + reg); +} + +static inline u32 bcm2708_clk_read_reg(struct bcm2708_i2s_dev *dev, int reg) +{ + return __raw_readl(dev->clk_base + reg); +} + +static void bcm2708_i2s_start_clock(struct bcm2708_i2s_dev *dev) +{ + /* + * Start the clock if in master mode. + */ + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + if (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBS_CFM) { + unsigned int clkreg = bcm2708_clk_read_reg(dev, + BCM2708_CLK_PCMCTL_REG); + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMCTL_REG, + BCM2708_CLK_PASSWD | clkreg | BCM2708_CLK_ENAB); + } +} + +static void bcm2708_i2s_stop_clock(struct bcm2708_i2s_dev *dev) +{ + int timeout = 1000; + + /* stop clock */ + unsigned int clkreg = bcm2708_clk_read_reg(dev, BCM2708_CLK_PCMCTL_REG); + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMCTL_REG, ~(BCM2708_CLK_ENAB) + & (BCM2708_CLK_PASSWD | clkreg)); + + /* wait for the BUSY flag going down */ + while ((bcm2708_clk_read_reg(dev, BCM2708_CLK_PCMCTL_REG) + & BCM2708_CLK_BUSY) + && timeout > 0) { + timeout--; + } + + if (timeout <= 0) { + /* KILL the clock */ + dev_err(dev->dev, "I2S clock didn't stop. Kill the clock!\n"); + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMCTL_REG, + BCM2708_CLK_KILL | BCM2708_CLK_PASSWD + | clkreg); + } +} + +static void bcm2708_i2s_clear_fifos(struct bcm2708_i2s_dev *dev) +{ + int timeout = 1000; + unsigned int syncval; + + /* Backup the current state */ + unsigned int i2s_active_state = bcm2708_i2s_read_reg(dev, + BCM2708_I2S_CS_A_REG) + & (BCM2708_I2S_RXON | BCM2708_I2S_TXON); + + unsigned int clk_active_state = bcm2708_clk_read_reg(dev, + BCM2708_CLK_PCMCTL_REG) & BCM2708_CLK_ENAB; + + /* Start clock if not running */ + if (!clk_active_state) { + unsigned int clkreg = bcm2708_clk_read_reg(dev, + BCM2708_CLK_PCMCTL_REG); + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMCTL_REG, + BCM2708_CLK_PASSWD | clkreg | BCM2708_CLK_ENAB); + } + + /* Stop I2S module */ + bcm2708_i2s_clear_bits(dev, BCM2708_I2S_CS_A_REG, BCM2708_I2S_RXON + | BCM2708_I2S_TXON); + + /* + * Clear the FIFOs + * Requires at least 2 PCM clock cycles to take effect + */ + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, BCM2708_I2S_RXCLR + | BCM2708_I2S_TXCLR); + + /* Wait for 2 PCM clock cycles */ + + /* + * Toggle the SYNC flag - after 2 PCM clock cycles it can be read back + * FIXME: This does not seem to work for slave mode! + */ + syncval = bcm2708_i2s_read_reg(dev, BCM2708_I2S_CS_A_REG) + & BCM2708_I2S_SYNC; + if (syncval) + bcm2708_i2s_clear_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_SYNC); + else + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_SYNC); + + /* Wait for the SYNC flag changing it's state */ + while (((bcm2708_i2s_read_reg(dev, BCM2708_I2S_CS_A_REG) + & BCM2708_I2S_SYNC) == syncval) + && timeout > 0) { + timeout--; + } + + if (timeout <= 0) + dev_err(dev->dev, "I2S SYNC error!\n"); + + /* Stop clock if it was not running before */ + if (!clk_active_state) + bcm2708_i2s_stop_clock(dev); + + /* Restore I2S state */ + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, i2s_active_state); +} + +static void bcm2708_i2s_start(struct bcm2708_i2s_dev *dev, + struct snd_pcm_substream *substream) +{ + bcm2708_i2s_start_clock(dev); + + /* + * Enable the stream. + */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_RXON); + } else { + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_TXON); + } +} + +static void bcm2708_i2s_stop(struct bcm2708_i2s_dev *dev, + struct snd_pcm_substream *substream) +{ + unsigned int still_running; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + bcm2708_i2s_clear_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_RXON); + } else { + bcm2708_i2s_clear_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_TXON); + } + + still_running = bcm2708_i2s_read_reg(dev, BCM2708_I2S_CS_A_REG) + & (BCM2708_I2S_TXON | BCM2708_I2S_RXON); + + /* Stop also the clock when not SND_SOC_DAIFMT_CONT */ + if (!still_running && !(dev->fmt & SND_SOC_DAIFMT_CONT)) + bcm2708_i2s_stop_clock(dev); + +} + +static int bcm2708_i2s_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + dev->fmt = fmt; + return 0; +} + +static bool bcm2708_i2s_check_other_stream(struct bcm2708_i2s_dev *dev, + struct snd_pcm_substream *substream, + struct snd_pcm_substream *other_stream, + struct snd_pcm_hw_params *params) +{ + if (other_stream->runtime->format && + (other_stream->runtime->format != params_format(params))) { + dev_err(dev->dev, + "Sample formats of streams are different. %i (%s) != %i (%s) Initialization failed!\n", + other_stream->runtime->format, + (other_stream->stream == + SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"), + params_format(params), + (substream->stream == + SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture")); + return false; + } + + if (other_stream->runtime->rate && + (other_stream->runtime->rate != params_rate(params))) { + dev_err(dev->dev, + "Sampling rates of streams are different. %i (%s) != %i (%s) Initialization failed!\n", + other_stream->runtime->rate, + (other_stream->stream == + SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"), + params_rate(params), + (substream->stream == + SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture")); + return false; + } + + return true; +} + +static int bcm2708_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + unsigned int sampling_rate = params_rate(params); + unsigned int data_length, data_delay, half_frame; + unsigned int ch1pos, ch2pos, mode, format; + unsigned int mash = BCM2708_CLK_MASH_1; + unsigned int divi, divf, target_frequency; + int clk_src = -1; + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + unsigned int bit_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBS_CFM); + + unsigned int frame_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBM_CFS); + + /* Ensure, that both streams have the same settings */ + struct snd_pcm_substream *other_stream = dev->first_stream; + if (other_stream == substream) + other_stream = dev->second_stream; + + if (other_stream != NULL + && !bcm2708_i2s_check_other_stream(dev, + substream, other_stream, params)) + return -EINVAL; + + /* + * If a stream is already enabled, + * the registers are already set properly. + */ + if (bcm2708_i2s_read_reg(dev, BCM2708_I2S_CS_A_REG) + & (BCM2708_I2S_TXON | BCM2708_I2S_RXON)) + return 0; + + /* + * Adjust the data length according to the format. + * We prefill the half frame length with an integer + * divider of 2400 as explained at the clock settings. + * Maybe it is overwritten there, if the Integer mode + * does not apply. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + data_length = 16; + half_frame = 20; + break; + case SNDRV_PCM_FORMAT_S32_LE: + data_length = 32; + half_frame = 40; + break; + default: + return -EINVAL; + } + + /* + * Clock Settings + * + * The target frequency of the bit clock is + * sampling rate * frame length + * + * Integer mode: + * Sampling rates that are multiples of 8000 kHz + * can be driven by the oscillator of 19.2 MHz + * with an integer divider as long as the frame length + * is an integer divider of 19200000/8000=2400 as set up above. + * This is no longer possible if the sampling rate + * is too high (e.g. 192 kHz), because the oscillator is too slow. + * + * MASH mode: + * For all other sampling rates, it is not possible to + * have an integer divider. Approximate the clock + * with the MASH module that induces a slight frequency + * variance. To minimize that it is best to have the fastest + * clock here. That is PLLD with 500 MHz. + */ + target_frequency = sampling_rate*half_frame*2; + clk_src = BCM2708_CLK_SRC_OSC; + mash = BCM2708_CLK_MASH_0; + + if (bcm2708_clk_freq[clk_src] % target_frequency == 0 + && bit_master && frame_master) { + divi = bcm2708_clk_freq[clk_src]/target_frequency; + divf = 0; + } else { + uint64_t dividend; + + /* + * Overwrite half frame length, because the + * above trick is not needed. + * This is fixed, because a bit clock of 64*fs + * seems to be what most codecs want. + * Is it necessary to have this dynamic? + */ + half_frame = 32; + target_frequency = sampling_rate*half_frame*2; + + clk_src = BCM2708_CLK_SRC_PLLD; + mash = BCM2708_CLK_MASH_1; + + dividend = bcm2708_clk_freq[clk_src]; + dividend *= 1024; + do_div(dividend, target_frequency); + divi = dividend / 1024; + divf = dividend % 1024; + } + + /* Set clock divider */ + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMDIV_REG, BCM2708_CLK_PASSWD + | BCM2708_CLK_DIVI(divi) + | BCM2708_CLK_DIVF(divf)); + + /* Setup clock, but don't start it yet */ + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMCTL_REG, BCM2708_CLK_PASSWD + | BCM2708_CLK_MASH(mash) + | BCM2708_CLK_SRC(clk_src)); + + + /* Setup the frame format */ + format = BCM2708_I2S_CHEN; + + if (data_length > 24) + format |= BCM2708_I2S_CHWEX; + + format |= BCM2708_I2S_CHWID((data_length-8)&0xf); + + switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + data_delay = 1; + break; + default: + /* TODO + * Others are possible but are not implemented at the moment. + */ + dev_err(dev->dev, "%s:bad format\n", __func__); + return -EINVAL; + } + + ch1pos = data_delay; + ch2pos = half_frame+data_delay; + + switch (params_channels(params)) { + case 2: + format = BCM2708_I2S_CH1(format) | BCM2708_I2S_CH2(format); + format |= BCM2708_I2S_CH1(BCM2708_I2S_CHPOS(ch1pos)); + format |= BCM2708_I2S_CH2(BCM2708_I2S_CHPOS(ch2pos)); + break; + default: + return -EINVAL; + } + + /* + * Set format for both streams. + * We cannot set another frame length + * (and therefore word length) anyway, + * so the format will be the same. + */ + bcm2708_i2s_write_reg(dev, BCM2708_I2S_RXC_A_REG, format); + bcm2708_i2s_write_reg(dev, BCM2708_I2S_TXC_A_REG, format); + + + /* Setup the I2S mode */ + mode = 0; + + if (data_length <= 16) { + /* + * Use frame packed mode (2 channels per 32 bit word) + * We cannot set another frame length in the second stream + * (and therefore word length) anyway, + * so the format will be the same. + */ + mode |= BCM2708_I2S_FTXP | BCM2708_I2S_FRXP; + } + + mode |= BCM2708_I2S_FLEN(half_frame*2-1); + mode |= BCM2708_I2S_FSLEN(half_frame); + + /* master or slave? */ + switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* cpu is master */ + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* + * codec is bit clock master + * cpu is frame master + */ + mode |= BCM2708_I2S_CLKM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + /* + * codec is frame master + * cpu is bit clock master + */ + mode |= BCM2708_I2S_FSM; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is master */ + mode |= BCM2708_I2S_CLKM; + mode |= BCM2708_I2S_FSM; + break; + default: + dev_err(dev->dev, "%s:bad master\n", __func__); + return -EINVAL; + } + + /* + * Invert clocks? + * + * The BCM approach seems to be inverted to the classical I2S approach. + */ + switch (dev->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* none - therefore, both for BCM */ + mode |= BCM2708_I2S_CLKI; + mode |= BCM2708_I2S_FSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* both - therefore, none for BCM*/ + break; + case SND_SOC_DAIFMT_NB_IF: + /* + * invert only frame sync - therefore, + * invert only bit clock for BCM + */ + mode |= BCM2708_I2S_CLKI; + break; + case SND_SOC_DAIFMT_IB_NF: + /* + * invert only bit clock - therefore, + * invert only frame sync for BCM + */ + mode |= BCM2708_I2S_FSI; + break; + default: + return -EINVAL; + } + + bcm2708_i2s_write_reg(dev, BCM2708_I2S_MODE_A_REG, mode); + + + /* Setup the DMA parameters */ + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, BCM2708_I2S_RXTHR(1) + | BCM2708_I2S_TXTHR(1) + | BCM2708_I2S_DMAEN); + + bcm2708_i2s_write_reg(dev, BCM2708_I2S_DREQ_A_REG, + BCM2708_I2S_TX_PANIC(0x10) + | BCM2708_I2S_RX_PANIC(0x30) + | BCM2708_I2S_TX(0x30) + | BCM2708_I2S_RX(0x20)); + + /* Clear FIFOs */ + bcm2708_i2s_clear_fifos(dev); + + return 0; +} + + +static int bcm2708_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + unsigned int cs_reg; + + bcm2708_i2s_start_clock(dev); + + /* + * Clear both FIFOs if the one that should be started + * is not empty at the moment. This should only happen + * after overrun. Otherwise, hw_params would have cleared + * the FIFO. + */ + cs_reg = bcm2708_i2s_read_reg(dev, BCM2708_I2S_CS_A_REG); + + switch (substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + if (cs_reg & BCM2708_I2S_TXE) + break; + case SNDRV_PCM_STREAM_CAPTURE: + if (!(cs_reg & BCM2708_I2S_RXD)) + break; + + bcm2708_i2s_clear_fifos(dev); + } + + return 0; +} + +static int bcm2708_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + bcm2708_i2s_start(dev, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + bcm2708_i2s_stop(dev, substream); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int bcm2708_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + if (!dev->first_stream) { + dev->first_stream = substream; + + /* should this still be running stop it */ + bcm2708_i2s_stop_clock(dev); + + /* enable PCM block */ + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, BCM2708_I2S_EN); + + /* + * Disable STBY + * Requires at least 4 PCM clock cycles to take effect + */ + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_STBY); + } else { + struct snd_pcm_runtime *first_runtime = + dev->first_stream->runtime; + + /* + * This is the second stream open, so we need to impose + * sample size and sampling rate constraints. + * This is because frame length and clock cannot be specified + * seperately. + * + * Note that this can cause a race condition if the + * second stream is opened before the first stream is + * fully initialized. We provide some protection by + * checking to make sure the first stream is + * initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample + * rate or size. If the second stream is opened + * before the first stream has received its final + * parameters, then the second stream may be + * constrained to the wrong sample rate or size. + * + * We will continue in case of failure and recheck the + * constraint in hw_params. + */ + if (!first_runtime->format) { + dev_err(substream->pcm->card->dev, + "Set format in %s stream first! " + "Initialization may fail.\n", + substream->stream == + SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + } else { + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, + first_runtime->format, + first_runtime->format); + } + + if (!first_runtime->rate) { + dev_err(substream->pcm->card->dev, + "Set sampling rate in %s stream first! " + "Initialization may fail!\n", + substream->stream == + SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + } else { + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + first_runtime->rate, + first_runtime->rate); + } + + dev->second_stream = substream; + } + + snd_soc_dai_set_dma_data(dai, substream, + &dev->dma_params[substream->stream]); + + return 0; +} + +static void bcm2708_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + bcm2708_i2s_stop(dev, substream); + + if (dev->first_stream == substream) + dev->first_stream = dev->second_stream; + + dev->second_stream = NULL; + + /* If both streams are stopped, disable module and clock */ + if (!dev->first_stream) { + /* Disable the module */ + bcm2708_i2s_clear_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_EN); + + /* + * Stopping clock is necessary, because stop does + * not stop the clock when SND_SOC_DAIFMT_CONT + */ + bcm2708_i2s_stop_clock(dev); + } +} + +static const struct snd_soc_dai_ops bcm2708_i2s_dai_ops = { + .startup = bcm2708_i2s_startup, + .shutdown = bcm2708_i2s_shutdown, + .prepare = bcm2708_i2s_prepare, + .trigger = bcm2708_i2s_trigger, + .hw_params = bcm2708_i2s_hw_params, + .set_fmt = bcm2708_i2s_set_dai_fmt, +}; + +static struct snd_soc_dai_driver bcm2708_i2s_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .ops = &bcm2708_i2s_dai_ops, + .symmetric_rates = 1 +}; + +static void bcm2708_i2s_setup_gpio(void) +{ + /* + * This is the common way to handle the GPIO pins for + * the Raspberry Pi. + * TODO Better way would be to handle + * this in the device tree! + */ +#define INP_GPIO(g) *(gpio+((g)/10)) &= ~(7<<(((g)%10)*3)) +#define SET_GPIO_ALT(g,a) *(gpio+(((g)/10))) |= (((a)<=3?(a)+4:(a)==4?3:2)<<(((g)%10)*3)) + + unsigned int *gpio; + int pin; + gpio = ioremap(GPIO_BASE, SZ_16K); + + /* SPI is on GPIO 7..11 */ + for (pin = 28; pin <= 31; pin++) { + INP_GPIO(pin); /* set mode to GPIO input first */ + SET_GPIO_ALT(pin, 2); /* set mode to ALT 0 */ + } +#undef INP_GPIO +#undef SET_GPIO_ALT +} + +static int bcm2708_i2s_probe(struct platform_device *pdev) +{ + struct bcm2708_i2s_dev *dev; + int i; + void __iomem *base[2]; + + /* request both ioareas */ + for (i = 0; i <= 1; i++) { + struct resource *mem, *ioarea; + mem = platform_get_resource(pdev, IORESOURCE_MEM, i); + if (!mem) { + dev_err(&pdev->dev, "I2S probe: Memory resource could not be found.\n"); + return -ENODEV; + } + + ioarea = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), + pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "I2S probe: Memory region already claimed.\n"); + return -EBUSY; + } + + base[i] = devm_ioremap(&pdev->dev, mem->start, + resource_size(mem)); + if (!base[i]) { + dev_err(&pdev->dev, "I2S probe: ioremap failed.\n"); + return -ENOMEM; + } + } + + dev = devm_kzalloc(&pdev->dev, sizeof(struct bcm2708_i2s_dev), + GFP_KERNEL); + if (!dev) { + dev_err(&pdev->dev, "I2S probe: kzalloc failed.\n"); + return -ENOMEM; + } + + dev->i2s_base = base[0]; + dev->clk_base = base[1]; + + bcm2708_i2s_setup_gpio(); + + /* Set the appropriate DMA parameters */ + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].port_addr = + (dma_addr_t)BCM2708_I2S_FIFO_PHYSICAL_ADDR; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].port_addr = + (dma_addr_t)BCM2708_I2S_FIFO_PHYSICAL_ADDR; + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_req = + BCM2708_DMA_DREQ_PCM_TX; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_req = + BCM2708_DMA_DREQ_PCM_RX; + + /* Store the pdev */ + dev->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, dev); + + return snd_soc_register_dai(&pdev->dev, &bcm2708_i2s_dai); +} + +static int bcm2708_i2s_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&pdev->dev); + return 0; +} + +static struct platform_driver bcm2708_i2s_driver = { + .probe = bcm2708_i2s_probe, + .remove = bcm2708_i2s_remove, + .driver = { + .name = "bcm2708-i2s", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(bcm2708_i2s_driver); + +MODULE_AUTHOR("Florian Meier"); +MODULE_DESCRIPTION("BCM2708 I2S Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/bcm2708/bcm2708-pcm.c b/sound/soc/bcm2708/bcm2708-pcm.c new file mode 100644 index 0000000..4b3e688 --- /dev/null +++ b/sound/soc/bcm2708/bcm2708-pcm.c @@ -0,0 +1,303 @@ +/* + * ALSA PCM interface for Broadcom BCM2708 SoC + * + * Author: Florian Meier, <florian.meier(a)koalo.de> + * Copyright 2013 + * + * based on + * ALSA PCM interface for the OMAP SoC + * Copyright (C) 2008 Nokia Corporation + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Peter Ujfalusi <peter.ujfalusi(a)ti.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * TODO This should be transferred to the generic DMA engine driver + * as soon as the official BCM2708 kernel migrates to 3.10. + * Otherwise, it would be difficult to test. + */ + +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/bcm2708-dmaengine.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> +#include <sound/soc.h> + +#include "bcm2708-pcm.h" + +static const struct snd_pcm_hardware bcm2708_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_JOINT_DUPLEX | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .period_bytes_min = 32, + .period_bytes_max = 64 * 1024, + .periods_min = 2, + .periods_max = 255, + .buffer_bytes_max = 128 * 1024, +}; + +static int bcm2708_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct bcm2708_pcm_dma_data *dma_data; + struct bcm2708_dma_slave_config config; + struct dma_chan *chan; + int err = 0; + + /* get DMA data (e.g. FIFO address and DREQ) */ + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + /* + * There seems to be no use for bufferless + * transfer with this SoC. + */ + if (!dma_data) + return 0; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + chan = snd_dmaengine_pcm_get_chan(substream); + if (!chan) + return -EINVAL; + + /* fills in addr_width and direction */ + err = snd_hwparams_to_dma_slave_config(substream, params, &config.cfg); + if (err) + return err; + + /* Override the *_dma addr_width if requested by the DAI driver */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + config.cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + else + config.cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + + config.cfg.src_addr = dma_data->port_addr; + config.cfg.dst_addr = dma_data->port_addr; + config.cfg.src_maxburst = dma_data->packet_size; + config.cfg.dst_maxburst = dma_data->packet_size; + config.sync_dreq = dma_data->dma_req; + + return dmaengine_slave_config(chan, &config.cfg); +} + +static int bcm2708_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static int bcm2708_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct bcm2708_pcm_dma_data *dma_data; + int ret = 0; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + default: + ret = -EINVAL; + } + + if (ret == 0) + ret = snd_dmaengine_pcm_trigger(substream, cmd); + + return ret; +} + +static snd_pcm_uframes_t bcm2708_pcm_pointer( + struct snd_pcm_substream *substream) +{ + return snd_dmaengine_pcm_pointer(substream); +} + +static int bcm2708_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct bcm2708_pcm_dma_data *dma_data; + int ret; + + snd_soc_set_runtime_hwparams(substream, &bcm2708_pcm_hardware); + + /* Ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + return snd_dmaengine_pcm_open(substream, NULL, NULL); +} + +static int bcm2708_pcm_close(struct snd_pcm_substream *substream) +{ + snd_dmaengine_pcm_close(substream); + return 0; +} + +static int bcm2708_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops bcm2708_pcm_ops = { + .open = bcm2708_pcm_open, + .close = bcm2708_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = bcm2708_pcm_hw_params, + .hw_free = bcm2708_pcm_hw_free, + .trigger = bcm2708_pcm_trigger, + .pointer = bcm2708_pcm_pointer, + .mmap = bcm2708_pcm_mmap, +}; + +static u64 bcm2708_pcm_dmamask = DMA_BIT_MASK(64); + +static int bcm2708_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = bcm2708_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static void bcm2708_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static int bcm2708_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &bcm2708_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(64); + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = bcm2708_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = bcm2708_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + /* free preallocated buffers in case of error */ + if (ret) + bcm2708_pcm_free_dma_buffers(pcm); + + return ret; +} + +static struct snd_soc_platform_driver bcm2708_soc_platform = { + .ops = &bcm2708_pcm_ops, + .pcm_new = bcm2708_pcm_new, + .pcm_free = bcm2708_pcm_free_dma_buffers, +}; + +static int bcm2708_pcm_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, + &bcm2708_soc_platform); +} + +static int bcm2708_pcm_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver bcm2708_pcm_driver = { + .driver = { + .name = "bcm2708-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = bcm2708_pcm_probe, + .remove = bcm2708_pcm_remove, +}; + +module_platform_driver(bcm2708_pcm_driver); + +MODULE_AUTHOR("Florian Meier"); +MODULE_DESCRIPTION("BCM2708 PCM DMA module"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bcm2708-pcm-audio"); diff --git a/sound/soc/bcm2708/bcm2708-pcm.h b/sound/soc/bcm2708/bcm2708-pcm.h new file mode 100644 index 0000000..e6a6acb --- /dev/null +++ b/sound/soc/bcm2708/bcm2708-pcm.h @@ -0,0 +1,40 @@ +/* + * ALSA PCM interface for Broadcom BCM2708 SoC + * + * Author: Florian Meier, <florian.meier(a)koalo.de> + * Copyright 2013 + * + * based on + * ALSA PCM interface for the OMAP SoC + * Copyright (C) 2008 Nokia Corporation + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Peter Ujfalusi <peter.ujfalusi(a)ti.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __BCM2708_PCM_H__ +#define __BCM2708_PCM_H__ + +struct snd_pcm_substream; + +struct bcm2708_pcm_dma_data { + char *name; /* stream identifier */ + int dma_req; /* DMA request line */ + unsigned long port_addr; /* transmit/receive register */ + int packet_size; +}; + +#endif diff --git a/sound/soc/bcm2708/rpi-mbed.c b/sound/soc/bcm2708/rpi-mbed.c new file mode 100644 index 0000000..cb8d25f --- /dev/null +++ b/sound/soc/bcm2708/rpi-mbed.c @@ -0,0 +1,113 @@ +/* + * ASoC driver for mbed AudioCODEC (with a TLV320AIC23B) + * connected to a Raspberry Pi + * + * Author: Florian Meier, <florian.meier(a)koalo.de> + * Copyright 2013 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "../codecs/tlv320aic23.h" + +static int snd_rpi_mbed_init(struct snd_soc_pcm_runtime *rtd) +{ + return 0; +} + +static int snd_rpi_mbed_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int sysclk; + + sysclk = 12000000; /* this is fixed on this board */ + + /* set tlv320aic23 sysclk */ + snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, 0); + + return 0; +} + +/* machine stream operations */ +static struct snd_soc_ops snd_rpi_mbed_ops = { + .hw_params = snd_rpi_mbed_hw_params, +}; + +static struct snd_soc_dai_link snd_rpi_mbed_dai[] = { +{ + .name = "TLV320AIC23", + .stream_name = "TLV320AIC23 HiFi", + .cpu_dai_name = "bcm2708-i2s.0", + .codec_dai_name = "tlv320aic23-hifi", + .platform_name = "bcm2708-pcm-audio.0", + .codec_name = "tlv320aic23-codec.1-001b", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &snd_rpi_mbed_ops, + .init = snd_rpi_mbed_init, +}, +}; + +/* audio machine driver */ +static struct snd_soc_card snd_rpi_mbed = { + .name = "snd_rpi_mbed", + .dai_link = snd_rpi_mbed_dai, + .num_links = ARRAY_SIZE(snd_rpi_mbed_dai), +}; + +static int snd_rpi_mbed_probe(struct platform_device *pdev) +{ + int ret = 0; + + snd_rpi_mbed.dev = &pdev->dev; + ret = snd_soc_register_card(&snd_rpi_mbed); + if (ret) { + dev_err(&pdev->dev, + "snd_soc_register_card() failed: %d\n", ret); + } + + return ret; +} + + +static int snd_rpi_mbed_remove(struct platform_device *pdev) +{ + return snd_soc_unregister_card(&snd_rpi_mbed); +} + +static struct platform_driver snd_rpi_mbed_driver = { + .driver = { + .name = "snd-rpi-mbed", + .owner = THIS_MODULE, + }, + .probe = snd_rpi_mbed_probe, + .remove = snd_rpi_mbed_remove, +}; + +module_platform_driver(snd_rpi_mbed_driver); + +MODULE_AUTHOR("Florian Meier"); +MODULE_DESCRIPTION("ASoC Driver for Raspberry Pi connected to mbed AudioCODEC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/bcm2708/rpi-proto.c b/sound/soc/bcm2708/rpi-proto.c new file mode 100644 index 0000000..25be85a --- /dev/null +++ b/sound/soc/bcm2708/rpi-proto.c @@ -0,0 +1,131 @@ +/* + * ASoC driver for PROTO AudioCODEC (with a WM8731) + * connected to a Raspberry Pi + * + * Author: Florian Meier, <florian.meier(a)koalo.de> + * Copyright 2013 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "../codecs/wm8731.h" + +static const unsigned int wm8731_rates_12288000[] = { + 8000, 32000, 48000, 96000, +}; + +static struct snd_pcm_hw_constraint_list wm8731_constraints_12288000 = { + .list = wm8731_rates_12288000, + .count = ARRAY_SIZE(wm8731_rates_12288000), +}; + +static int snd_rpi_proto_startup(struct snd_pcm_substream *substream) +{ + /* Setup constraints, because there is a 12.288 MHz XTAL on the board */ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &wm8731_constraints_12288000); + return 0; +} + +static int snd_rpi_proto_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int sysclk = 12288000; /* This is fixed on this board */ + + /* Set proto sysclk */ + int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, + sysclk, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(substream->pcm->dev, + "Failed to set WM8731 SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +/* machine stream operations */ +static struct snd_soc_ops snd_rpi_proto_ops = { + .startup = snd_rpi_proto_startup, + .hw_params = snd_rpi_proto_hw_params, +}; + +static struct snd_soc_dai_link snd_rpi_proto_dai[] = { +{ + .name = "WM8731", + .stream_name = "WM8731 HiFi", + .cpu_dai_name = "bcm2708-i2s.0", + .codec_dai_name = "wm8731-hifi", + .platform_name = "bcm2708-pcm-audio.0", + .codec_name = "wm8731.1-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &snd_rpi_proto_ops, +}, +}; + +/* audio machine driver */ +static struct snd_soc_card snd_rpi_proto = { + .name = "snd_rpi_proto", + .dai_link = snd_rpi_proto_dai, + .num_links = ARRAY_SIZE(snd_rpi_proto_dai), +}; + +static int snd_rpi_proto_probe(struct platform_device *pdev) +{ + int ret = 0; + + snd_rpi_proto.dev = &pdev->dev; + ret = snd_soc_register_card(&snd_rpi_proto); + if (ret) { + dev_err(&pdev->dev, + "snd_soc_register_card() failed: %d\n", ret); + } + + return ret; +} + + +static int snd_rpi_proto_remove(struct platform_device *pdev) +{ + return snd_soc_unregister_card(&snd_rpi_proto); +} + +static struct platform_driver snd_rpi_proto_driver = { + .driver = { + .name = "snd-rpi-proto", + .owner = THIS_MODULE, + }, + .probe = snd_rpi_proto_probe, + .remove = snd_rpi_proto_remove, +}; + +module_platform_driver(snd_rpi_proto_driver); + +MODULE_AUTHOR("Florian Meier"); +MODULE_DESCRIPTION("ASoC Driver for Raspberry Pi connected to PROTO board (WM8731)"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

4 9

[alsa-devel] [PATCH alsa-lib] Allow specifying the max number of cards
by Takashi Iwai 24 May '13

24 May '13

Add --with-max-cards option to specify the max number of cards in configure script, when the support for more than 32 cards is required. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- configure.in | 14 ++++++++++++++ src/confmisc.c | 2 +- src/control/cards.c | 6 +++--- src/control/control_hw.c | 2 +- src/hwdep/hwdep_hw.c | 2 +- 5 files changed, 20 insertions(+), 6 deletions(-) diff --git a/configure.in b/configure.in index 35b8e84..3d394fc 100644 --- a/configure.in +++ b/configure.in @@ -632,6 +632,20 @@ for t in $CTL_PLUGIN_LIST; do fi done +dnl Max number of cards +AC_MSG_CHECKING(for max number of cards) +AC_ARG_WITH(max-cards, + AS_HELP_STRING([--with-max-cards], [Specify the max number of cards (default = 32)]), + [ max_cards="$withval" ], [ max_cards="32" ]) +AC_MSG_RESULT([$max_cards]) + +if test "$max_cards" -lt 1; then + AC_ERROR([Invalid max cards $max_cards]) +elif test "$max_cards" -gt 256; then + AC_ERROR([Invalid max cards $max_cards]) +fi +AC_DEFINE_UNQUOTED(SND_MAX_CARDS, $max_cards, [Max number of cards]) + dnl Make a symlink for inclusion of alsa/xxx.h if test ! -L "$srcdir"/include/alsa ; then echo "Making a symlink include/alsa" diff --git a/src/confmisc.c b/src/confmisc.c index 80b0027..af686be 100644 --- a/src/confmisc.c +++ b/src/confmisc.c @@ -668,7 +668,7 @@ int snd_determine_driver(int card, char **driver) char *res = NULL; int err; - assert(card >= 0 && card <= 32); + assert(card >= 0 && card <= SND_MAX_CARDS); err = open_ctl(card, &ctl); if (err < 0) { SNDERR("could not open control for card %i", card); diff --git a/src/control/cards.c b/src/control/cards.c index b528e33..5d7376c 100644 --- a/src/control/cards.c +++ b/src/control/cards.c @@ -103,7 +103,7 @@ int snd_card_next(int *rcard) return -EINVAL; card = *rcard; card = card < 0 ? 0 : card + 1; - for (; card < 32; card++) { + for (; card < SND_MAX_CARDS; card++) { if (snd_card_load(card)) { *rcard = card; return 0; @@ -134,7 +134,7 @@ int snd_card_get_index(const char *string) (isdigit(*string) && isdigit(*(string + 1)) && *(string + 2) == 0)) { if (sscanf(string, "%i", &card) != 1) return -EINVAL; - if (card < 0 || card > 31) + if (card < 0 || card >= SND_MAX_CARDS) return -EINVAL; err = snd_card_load1(card); if (err >= 0) @@ -143,7 +143,7 @@ int snd_card_get_index(const char *string) } if (string[0] == '/') /* device name */ return snd_card_load2(string); - for (card = 0; card < 32; card++) { + for (card = 0; card < SND_MAX_CARDS; card++) { #ifdef SUPPORT_ALOAD if (! snd_card_load(card)) continue; diff --git a/src/control/control_hw.c b/src/control/control_hw.c index 90c4ba7..148097f 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -382,7 +382,7 @@ int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, int mode) *handle = NULL; - if (CHECK_SANITY(card < 0 || card >= 32)) { + if (CHECK_SANITY(card < 0 || card >= SND_MAX_CARDS)) { SNDMSG("Invalid card index %d", card); return -EINVAL; } diff --git a/src/hwdep/hwdep_hw.c b/src/hwdep/hwdep_hw.c index e4fcdc3..4314e32 100644 --- a/src/hwdep/hwdep_hw.c +++ b/src/hwdep/hwdep_hw.c @@ -112,7 +112,7 @@ int snd_hwdep_hw_open(snd_hwdep_t **handle, const char *name, int card, int devi *handle = NULL; - if (card < 0 || card >= 32) + if (card < 0 || card >= SND_MAX_CARDS) return -EINVAL; sprintf(filename, SNDRV_FILE_HWDEP, card, device); fd = snd_open_device(filename, mode); -- 1.8.2.3

1 0

[alsa-devel] [PATCH 1/2] ALSA: Fix the default suffix string with high card number
by Takashi Iwai 24 May '13

24 May '13

ALSA core tries to add a suffix as "_1" automatically when the given id string conflicts. The current code assumes implicitly that the max card number is 16 so that the single hex "_X" suffix can be put. However, with the dynamic device management, the card can be at most 32, so it can put even a non-hex character there. Also, when the max card number is increased in future, this would result in worse. This patch rewrites the code to add the suffix string in a simpler (thus cleaner) way. It can support up to three digits, so it should suffice for most requirements. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 30 +++++++++++++----------------- 1 file changed, 13 insertions(+), 17 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 6ef0640..ed4a481 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -549,7 +549,6 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, const char *nid) { int len, loops; - bool with_suffix; bool is_default = false; char *id; @@ -565,26 +564,23 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, is_default = true; } - with_suffix = false; + len = strlen(id); for (loops = 0; loops < SNDRV_CARDS; loops++) { + char *spos; + char sfxstr[5]; /* "_012" */ + int sfxlen; + if (card_id_ok(card, id)) return; /* OK */ - len = strlen(id); - if (!with_suffix) { - /* add the "_X" suffix */ - char *spos = id + len; - if (len > sizeof(card->id) - 3) - spos = id + sizeof(card->id) - 3; - strcpy(spos, "_1"); - with_suffix = true; - } else { - /* modify the existing suffix */ - if (id[len - 1] != '9') - id[len - 1]++; - else - id[len - 1] = 'A'; - } + /* Add _XYZ suffix */ + sprintf(sfxstr, "_%X", loops + 1); + sfxlen = strlen(sfxstr); + if (len + sfxlen >= sizeof(card->id)) + spos = id + sizeof(card->id) - sfxlen - 1; + else + spos = id + len; + strcpy(spos, sfxstr); } /* fallback to the default id */ if (!is_default) { -- 1.8.2.3

1 1

[alsa-devel] [PATCH] ALSA: Constify the snd_pcm_substream struct ops field
by Lars-Peter Clausen 24 May '13

24 May '13

The ops field of the snd_pcm_substream struct is never modified inside the ALSA core. Making it const allows drivers to declare their snd_pcm_ops struct as const. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/pcm.h | 5 +++-- sound/core/pcm_lib.c | 3 ++- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index b48792f..84b10f9 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -384,7 +384,7 @@ struct snd_pcm_substream { unsigned int dma_buf_id; size_t dma_max; /* -- hardware operations -- */ - struct snd_pcm_ops *ops; + const struct snd_pcm_ops *ops; /* -- runtime information -- */ struct snd_pcm_runtime *runtime; /* -- timer section -- */ @@ -871,7 +871,8 @@ const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format); int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int frames); snd_pcm_format_t snd_pcm_build_linear_format(int width, int unsigned, int big_endian); -void snd_pcm_set_ops(struct snd_pcm * pcm, int direction, struct snd_pcm_ops *ops); +void snd_pcm_set_ops(struct snd_pcm * pcm, int direction, + const struct snd_pcm_ops *ops); void snd_pcm_set_sync(struct snd_pcm_substream *substream); int snd_pcm_lib_interleave_len(struct snd_pcm_substream *substream); int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 41b3dfe..82bb029 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -568,7 +568,8 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) * * Sets the given PCM operators to the pcm instance. */ -void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, struct snd_pcm_ops *ops) +void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, + const struct snd_pcm_ops *ops) { struct snd_pcm_str *stream = &pcm->streams[direction]; struct snd_pcm_substream *substream; -- 1.8.0

2 1

[alsa-devel] My ALSA amidicat program
by Josh Lehan 24 May '13

24 May '13

Hello! A few years ago, we talked on the alsa-devel mailing list. I had written a program "amidicat" which was well received. It's been a while, and I just now remembered it again. http://mailman.alsa-project.org/pipermail/alsa-devel/2010-June/028446.html You mentioned that such a program might be useful to have in alsa-utils. Is this still the case? If so, I would be honored to get my program in there. http://mailman.alsa-project.org/pipermail/alsa-devel/2010-January/024413.ht… Thanks! Josh Lehan

2 1

[alsa-devel] [PATCH] ucm: Fix segfault when device argument is missing from _sw{dev, mod}
by Juho Hämäläinen 24 May '13

24 May '13

Signed-off-by: Juho Hämäläinen <juho.hamalainen(a)tieto.com> --- src/ucm/main.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/src/ucm/main.c b/src/ucm/main.c index d38b7fa..74e1530 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1649,6 +1649,10 @@ int snd_use_case_set(snd_use_case_mgr_t *uc_mgr, } else { str = NULL; } + if (!str) { + err = -EINVAL; + goto __end; + } if (check_identifier(identifier, "_swdev")) err = switch_device(uc_mgr, str, value); else if (check_identifier(identifier, "_swmod")) -- 1.7.10.4

2 3

[alsa-devel] [PATCH] ASoC: wm5110: Correct DSP4R Mixer control name
by Charles Keepax 24 May '13

24 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c00480b..ba38f06 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), -- 1.7.2.5

1 0

[alsa-devel] [PATCH 4/4 V2] ALSA: hda - Continue probe in work context to avoid request_module deadlock
by Wang Xingchao 24 May '13

24 May '13

There's deadlock when request_module(i915) in azx_probe. It looks like: device_lock(audio pci device) -> azx_probe -> module_request (or symbol_request) -> modprobe (userspace) -> i915 init -> drm_pci_init -> pci_register_driver -> bus_add_driver -> driver_attach -> which in turn tries all locks on pci bus, and when it tries the one on the audio device, it will deadlock. This patch introduce a work to store remaining probe stuff, and let request_module run in safe work context. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/hda_i915.c | 13 ++++-- sound/pci/hda/hda_intel.c | 105 +++++++++++++++++++++++++++------------------- 2 files changed, 71 insertions(+), 47 deletions(-) diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c index 76c13d5..7547b20 100644 --- a/sound/pci/hda/hda_i915.c +++ b/sound/pci/hda/hda_i915.c @@ -42,13 +42,18 @@ int hda_i915_init(void) { int err = 0; - get_power = symbol_request(i915_request_power_well); + get_power = symbol_get(i915_request_power_well); if (!get_power) { - snd_printk(KERN_WARNING "hda-i915: get_power symbol get fail\n"); - return -ENODEV; + request_module("i915"); + get_power = symbol_get(i915_request_power_well); + if (!get_power) { + snd_printk(KERN_WARNING "hda-i915: get_power symbol get fail\n"); + return -ENODEV; + } + snd_printdd("hda-i915: get_power symbol get successful\n"); } - put_power = symbol_request(i915_release_power_well); + put_power = symbol_get(i915_release_power_well); if (!put_power) { symbol_put(i915_request_power_well); get_power = NULL; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f20a88c..1bc7c3b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -76,6 +76,7 @@ static int probe_only[SNDRV_CARDS]; static int jackpoll_ms[SNDRV_CARDS]; static bool single_cmd; static int enable_msi = -1; +static int dev; #ifdef CONFIG_SND_HDA_PATCH_LOADER static char *patch[SNDRV_CARDS]; #endif @@ -542,6 +543,8 @@ struct azx { /* for pending irqs */ struct work_struct irq_pending_work; + struct delayed_work probe_work; + /* reboot notifier (for mysterious hangup problem at power-down) */ struct notifier_block reboot_notifier; @@ -3670,58 +3673,22 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) } #endif -static int azx_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) +static void azx_probe_work(struct work_struct *work) { - static int dev; - struct snd_card *card; - struct azx *chip; + struct azx *chip = + container_of(work, struct azx, probe_work.work); + struct snd_card *card = chip->card; + struct pci_dev *pci = chip->pci; bool probe_now; int err; - if (dev >= SNDRV_CARDS) - return -ENODEV; - if (!enable[dev]) { - dev++; - return -ENOENT; - } - - err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); - if (err < 0) { - snd_printk(KERN_ERR "hda-intel: Error creating card!\n"); - return err; - } - - snd_card_set_dev(card, &pci->dev); - - err = azx_create(card, pci, dev, pci_id->driver_data, &chip); - if (err < 0) - goto out_free; - card->private_data = chip; - - pci_set_drvdata(pci, card); - - err = register_vga_switcheroo(chip); - if (err < 0) { - snd_printk(KERN_ERR SFX - "%s: Error registering VGA-switcheroo client\n", pci_name(pci)); - goto out_free; - } - - if (check_hdmi_disabled(pci)) { - snd_printk(KERN_INFO SFX "%s: VGA controller is disabled\n", - pci_name(pci)); - snd_printk(KERN_INFO SFX "%s: Delaying initialization\n", pci_name(pci)); - chip->disabled = true; - } - /* Request power well for Haswell HDA controller and codec */ if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { #ifdef CONFIG_SND_HDA_I915 err = hda_i915_init(); if (err < 0) { snd_printk(KERN_ERR SFX "Error request power-well from i915\n"); - return err; + goto out_free; } hda_display_power(true); #else @@ -3760,7 +3727,7 @@ static int azx_probe(struct pci_dev *pci, dev++; complete_all(&chip->probe_wait); - return 0; + return; out_free_power: if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(false); @@ -3769,6 +3736,58 @@ out_free_power: out_free: snd_card_free(card); pci_set_drvdata(pci, NULL); +} + +static int azx_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct azx *chip; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) { + snd_printk(KERN_ERR "hda-intel: Error creating card!\n"); + return err; + } + + snd_card_set_dev(card, &pci->dev); + + err = azx_create(card, pci, dev, pci_id->driver_data, &chip); + if (err < 0) + goto out_free; + card->private_data = chip; + + pci_set_drvdata(pci, card); + + err = register_vga_switcheroo(chip); + if (err < 0) { + snd_printk(KERN_ERR SFX + "%s: Error registering VGA-switcheroo client\n", pci_name(pci)); + goto out_free; + } + + if (check_hdmi_disabled(pci)) { + snd_printk(KERN_INFO SFX "%s: VGA controller is disabled\n", + pci_name(pci)); + snd_printk(KERN_INFO SFX "%s: Delaying initialization\n", pci_name(pci)); + chip->disabled = true; + } + + /* continue probing in work context as may trigger request module */ + INIT_DELAYED_WORK(&chip->probe_work, azx_probe_work); + schedule_delayed_work(&chip->probe_work, 0); + return 0; +out_free: + snd_card_free(card); + pci_set_drvdata(pci, NULL); return err; } -- 1.8.1.2

4 6

Re: [alsa-devel] [PATCH 00/33] devm improvement series, part 1, take 2
by Thierry Reding 23 May '13

23 May '13

On Thu, May 16, 2013 at 01:15:28PM +0200, Wolfram Sang wrote: > Lately, I have been experimenting how to improve the devm interface to make > writing device drivers easier and less error prone while also getting rid of > its subtle issues. I think it has more potential but still needs work and > definately conistency, especiall in its usage. > > The first thing I come up with is a low hanging fruit regarding > devm_ioremap_resouce(). This function already checks if the passed resource is > valid and gives an error message if not. So, we can remove similar checks from > the drivers and get rid of a bit of code and a number of inconsistent error > strings. Sorry for jumping in so late. I generally like the idea. One small inconvenience is that devm_ioremap_resource() returns -EINVAL if res == NULL, which means that drivers will now also return -EINVAL in cases where no resource was returned. Typically drivers handle this by returning something like -ENODEV, -ENXIO, -ENOENT. Some do return -EINVAL but perhaps having a separate error code (and maybe error message as well) for a missing resource would be helpful. Doing this would be rather easy now that you've paved the way by making devm_ioremap_resource() usage consistent across drivers. Thierry

1 0

[alsa-devel] [PATCH 0/4 V5] Power-well API implementation for Haswell
by Wang Xingchao 23 May '13

23 May '13

Hi all, This is V5 and here're some changes notes: change from V4-->V5: - fix reference count bug - new patch on general runtime pm support for audio pci device - new patch to avoid request_module() deadlock change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (4): i915/drm: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA ALSA: hda - Fix runtime PM check ALSA: hda - Continue probe in work context to avoid request_module deadlock drivers/gpu/drm/i915/i915_dma.c | 6 ++ drivers/gpu/drm/i915/i915_drv.h | 12 ++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 92 ++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 +++++++++++ sound/pci/hda/Kconfig | 10 +++ sound/pci/hda/Makefile | 3 + sound/pci/hda/hda_i915.c | 75 ++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 +++++++++++ sound/pci/hda/hda_intel.c | 132 ++++++++++++++++++++++++++++----------- 10 files changed, 360 insertions(+), 45 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.8.1.2

4 12

[alsa-devel] [PATCH 1/2] ASoC: codecs: adau1701: refactor firmware loading function
by Daniel Mack 23 May '13

23 May '13

Pass a struct i2c_client * to adau1701_load_firmware directly to make the code more readable. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/adau1701.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index dafdbe8..95e1677 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -180,9 +180,9 @@ static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) return value; } -static int adau1701_load_firmware(struct snd_soc_codec *codec) +static int adau1701_load_firmware(struct i2c_client *client) { - return process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE); + return process_sigma_firmware(client, ADAU1701_FIRMWARE); } static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, @@ -455,10 +455,11 @@ static struct snd_soc_dai_driver adau1701_dai = { static int adau1701_probe(struct snd_soc_codec *codec) { int ret; + struct i2c_client *client = to_i2c_client(codec->dev); - codec->control_data = to_i2c_client(codec->dev); + codec->control_data = client; - ret = adau1701_load_firmware(codec); + ret = adau1701_load_firmware(client); if (ret) dev_warn(codec->dev, "Failed to load firmware\n"); -- 1.8.1.4

3 4

[alsa-devel] [PATCH 0/4] CS42L52 fixes.
by Nicolas Schichan 23 May '13

23 May '13

Hi, In the following emails, you will find the fixes I currently have for the CS42L52 driver. The fixes are: - Set both Master Volume A & B init value to 0x88 (otherwise left channel and right channel master volume would not be the same after init). - Use IFACE_CTL2 register for Microphone Bias enum (IFACE_CTL1 is clearly not controling microphone bias according to the datasheet). - Various controls would use a bogus shift value, resulting in only the high order bits being updated when changing them. - The wrong mask was used when muting/unmuting the codec, resutling in the wrong part of the PB_CTL1 register being cleared, especially the headphone analog gain part. All those patches are against the latest linus tree. Regards, -- Nicolas Schichan

4 15

[alsa-devel] [PATCH] pcm: Fix typo in documentation: PREPARE → PREPARED.
by Jordi Mallach 23 May '13

23 May '13

Signed-off-by: Jordi Mallach <jordi(a)debian.org> --- src/pcm/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 438fb3f..7ec1f0e 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -437,7 +437,7 @@ to #SND_PCM_STATE_SETUP if successfully finishes, otherwise the state #SND_PCM_STATE_OPEN is entered. When it is brought to SETUP state, this function automatically -calls #snd_pcm_prepare() function to bring to the PREPARE state +calls #snd_pcm_prepare() function to bring to the PREPARED state as below. \par snd_pcm_prepare -- 1.7.10.4

2 1

[alsa-devel] [PATCH alsa-utils] Require alsa-lib 1.0.27.
by Jordi Mallach 23 May '13

23 May '13

Both aplay and speaker-test use the new snd_pcm_abort() function, which was introduced in alsa-lib 1.0.27. Signed-off-by: Jordi Mallach <jordi(a)debian.org> --- configure.in | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.in b/configure.in index aae3e66..0023e4d 100644 --- a/configure.in +++ b/configure.in @@ -32,7 +32,7 @@ AC_PROG_MKDIR_P AC_PROG_LN_S AC_PROG_SED PKG_PROG_PKG_CONFIG -AM_PATH_ALSA(1.0.24) +AM_PATH_ALSA(1.0.27) if test "x$enable_alsatest" = "xyes"; then AC_CHECK_FUNC([snd_ctl_elem_add_enumerated], , [AC_ERROR([No user enum control support in alsa-lib])]) -- 1.7.10.4

2 1

[alsa-devel] [PATCH] usb-6fire: Modify firmware version check
by Torsten Schenk 23 May '13

23 May '13

Check only the uppermost 16 bits instead of the whole 32 bits of the version information. Do this because all firmware version tested with this version information worked correctly and the strict check causes problems for several users. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c --- a/sound/usb/6fire/firmware.c 2013-05-23 13:32:39.432545897 +0200 +++ b/sound/usb/6fire/firmware.c 2013-05-23 13:33:21.998193012 +0200 @@ -42,8 +42,8 @@ 0x94, 0x01, 0x5c, 0x02 /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */ }; -static const u8 known_fw_versions[][4] = { - { 0x03, 0x01, 0x0b, 0x00 } +static const u8 known_fw_versions[][2] = { + { 0x03, 0x01 } }; struct ihex_record { @@ -343,7 +343,7 @@ int i; for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++) - if (!memcmp(version, known_fw_versions + i, 4)) + if (!memcmp(version, known_fw_versions + i, 2)) return 0; snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "

2 1

[alsa-devel] [PATCH 01/11] ALSA: pxa2xx-ac97: Remove redundant platform_set_drvdata()
by Sachin Kamat 23 May '13

23 May '13

Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> Cc: Eric Miao <eric.y.miao(a)gmail.com> --- sound/arm/pxa2xx-ac97.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index ec54be4..ce431e6 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -230,7 +230,6 @@ static int pxa2xx_ac97_remove(struct platform_device *dev) if (card) { snd_card_free(card); - platform_set_drvdata(dev, NULL); pxa2xx_ac97_hw_remove(dev); } -- 1.7.9.5

3 12

[alsa-devel] [PATCH] ASoC: fsl: expand the size of the name in fsl_ssi_private struct
by Nicolin Chen 23 May '13

23 May '13

strcpy(ssi_private->name, p) in probe() sets "name" by "p", gotten from dts, while the length of "p", if the devicetree node name of SSI is commonly set, would always be larger than 1 char size, so need a larger size for "name". Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837..b6a5f94 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -152,5 +152,5 @@ struct fsl_ssi_private { } stats; - char name[1]; + char name[32]; }; -- 1.7.1

3 2

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] crec: Initial version of a compressed capture utility
by Richard Fitzgerald 23 May '13

23 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- Android.mk | 10 ++ crec.c | 339 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ makefile.linux | 20 ++- 3 files changed, 362 insertions(+), 7 deletions(-) create mode 100644 crec.c diff --git a/Android.mk b/Android.mk index b9c52e2..5baccaf 100644 --- a/Android.mk +++ b/Android.mk @@ -20,3 +20,13 @@ LOCAL_MODULE_TAGS := optional include $(BUILD_EXECUTABLE) +include $(CLEAR_VARS) + +LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include +LOCAL_SRC_FILES:= crec.c +LOCAL_MODULE := crec +LOCAL_SHARED_LIBRARIES:= libcutils libutils libtinycompress +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) + diff --git a/crec.c b/crec.c new file mode 100644 index 0000000..4f13878 --- /dev/null +++ b/crec.c @@ -0,0 +1,339 @@ +/* + * BSD LICENSE + * + * tinyplay command line player for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)linux.intel.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * tinyplay command line player for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "sound/compress_params.h" +#include "tinycompress/tinycompress.h" + +static int verbose; + +static void usage(void) +{ + fprintf(stderr, "usage: crec [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n\n" + "-v\tverbose mode\n" + "-h\tPrints this help list\n\n" + "Example:\n" + "\tcrec -c 1 -d 2 test.raw\n" + "\tcrec -f 5 test.raw\n"); + + exit(EXIT_FAILURE); +} + +static int sample_rate_to_alsa(unsigned int *alsa_rate, int rate) +{ + switch (rate) { + case 5512: + *alsa_rate = SNDRV_PCM_RATE_5512; + return 0; + case 8000: + *alsa_rate = SNDRV_PCM_RATE_8000; + return 0; + case 11025: + *alsa_rate = SNDRV_PCM_RATE_11025; + return 0; + case 16000: + *alsa_rate = SNDRV_PCM_RATE_16000; + return 0; + case 220500: + *alsa_rate = SNDRV_PCM_RATE_22050; + return 0; + case 32000: + *alsa_rate = SNDRV_PCM_RATE_32000; + return 0; + case 44100: + *alsa_rate = SNDRV_PCM_RATE_44100; + return 0; + case 48000: + *alsa_rate = SNDRV_PCM_RATE_48000; + return 0; + case 64000: + *alsa_rate = SNDRV_PCM_RATE_64000; + return 0; + case 88200: + *alsa_rate = SNDRV_PCM_RATE_88200; + return 0; + case 96000: + *alsa_rate = SNDRV_PCM_RATE_96000; + return 0; + case 176400: + *alsa_rate = SNDRV_PCM_RATE_176400; + return 0; + case 192000: + *alsa_rate = SNDRV_PCM_RATE_192000; + return 0; + default: + return -EINVAL; + } +} + +static int alsa_to_sample_rate(unsigned int alsa_rate) +{ + switch (alsa_rate) { + case SNDRV_PCM_RATE_5512: + return 5512; + case SNDRV_PCM_RATE_8000: + return 8000; + case SNDRV_PCM_RATE_11025: + return 11025; + case SNDRV_PCM_RATE_16000: + return 16000; + case SNDRV_PCM_RATE_22050: + return 22050; + case SNDRV_PCM_RATE_32000: + return 32000; + case SNDRV_PCM_RATE_44100: + return 44100; + case SNDRV_PCM_RATE_48000: + return 48000; + case SNDRV_PCM_RATE_64000: + return 64000; + case SNDRV_PCM_RATE_88200: + return 88200; + case SNDRV_PCM_RATE_96000: + return 96000; + case SNDRV_PCM_RATE_176400: + return 176400; + case SNDRV_PCM_RATE_192000: + return 192000; + default: + return -EINVAL; + } +} + +static int print_time(struct compress *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (compress_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + return -1; + } else + fprintf(stderr, "DSP played %jd.%jd\n", (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + return 0; +} + +void capture_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag, unsigned int rate) +{ + struct compr_config config; + struct snd_codec codec; + struct compress *compress; + FILE *file; + char *buffer; + int size, num_read, wrote; + + if (verbose) + printf("%s: entry\n", __func__); + file = fopen(name, "wb"); + if (!file) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = 1; + codec.ch_out = 1; + if (sample_rate_to_alsa(&codec.sample_rate, rate) != 0) { + fprintf(stderr, "unknown sample rate %d\n", rate); + goto FILE_EXIT; + } + + codec.bit_rate = 0; + codec.rate_control = 0; + codec.profile = 0; + codec.level = 0; + codec.ch_mode = 0; + codec.format = 0; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } else { + /* use driver defaults */ + config.fragment_size = 0; + config.fragments = 0; + } + config.codec = &codec; + + compress = compress_open(card, device, COMPRESS_OUT, &config); + if (!compress || !is_compress_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto FILE_EXIT; + }; + if (verbose) + printf("%s: Opened compress device\n", __func__); + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto COMP_EXIT; + } + + printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + printf("Format %u Channels %u, %u Hz\n", + codec.id, codec.ch_out, + alsa_to_sample_rate(codec.sample_rate)); + + compress_start(compress); + if (verbose) + printf("%s: Capturing audio NOW!!!\n", __func__); + + do { + num_read = compress_read(compress, buffer, size); + if (num_read > 0) { + wrote = fwrite(buffer, 1, num_read, file); + if (wrote < 0) { + fprintf(stderr, "Error writing output file\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto BUF_EXIT; + } + if (wrote != num_read) { + /* TODO: Buffer pointer needs to be set here */ + fprintf(stderr, "DSP supplied %d, we wrote %d\n", num_read, wrote); + } + if (verbose) { + print_time(compress); + printf("%s: read %d\n", __func__, num_read); + } + } + } while (num_read > 0); + + if (verbose) + printf("%s: exit success\n", __func__); + free(buffer); + fclose(file); + compress_close(compress); + return; +BUF_EXIT: + free(buffer); +COMP_EXIT: + compress_close(compress); +FILE_EXIT: + fclose(file); + if (verbose) + printf("%s: exit failure\n", __func__); + exit(EXIT_FAILURE); +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0, rate = 16000; + + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "rhvb:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'r': + rate = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + capture_samples(file, card, device, buffer_size, frag, rate); + + fprintf(stderr, "Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); +} + diff --git a/makefile.linux b/makefile.linux index 60d6c04..9404fd8 100644 --- a/makefile.linux +++ b/makefile.linux @@ -1,10 +1,13 @@ LIB = libtinycompress -BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c +PLAYBIN = cplay +RECBIN = crec +PLAYSRC = cplay.c +RECSRC = crec.c LIBOBJ = ${LIBSRC:.c=.o} -OBJ = ${SRC:.c=.o} +PLAYOBJ = ${PLAYSRC:.c=.o} +RECOBJ = ${RECSRC:.c=.o} CC = gcc CROSS_COMPILE = @@ -13,23 +16,26 @@ CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress LIBLDFLAGS = -lasound -all: libtinycompress cplay +all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} -$(BIN): ${OBJ} ${LIB} +$(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} +$(RECBIN): ${RECOBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: - @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} .PHONY: all clean -- 1.7.2.5

3 2

[alsa-devel] [PATCH] ASoC: fsl: expand the size of the name in fsl_ssi_private struct
by Nicolin Chen 23 May '13

23 May '13

strcpy(ssi_private->name, p) in probe() sets "name" by "p", gotten from dts, while the length of "p", if the devicetree node name of SSI is commonly set, would always be larger than 1 char size, so need a larger size for "name". Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837..b6a5f94 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -152,5 +152,5 @@ struct fsl_ssi_private { } stats; - char name[1]; + char name[32]; }; -- 1.7.1

1 0

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] compress: fix hpointer error when no sample rate
by Richard Fitzgerald 23 May '13

23 May '13

Fixes the oops() in compress_get_hpointer() to return -ENODATA instead of errno if the sample rate is zero. The value in errno here is not related to the reason we are returning an error. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/compress.c b/compress.c index e80e598..abd0799 100644 --- a/compress.c +++ b/compress.c @@ -315,7 +315,7 @@ int compress_get_hpointer(struct compress *compress, if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &kavail)) return oops(compress, errno, "cannot get avail"); if (0 == kavail.tstamp.sampling_rate) - return oops(compress, errno, "invalid paramter"); + return oops(compress, -ENODATA, "sample rate unknown"); *avail = (unsigned int)kavail.avail; time = kavail.tstamp.pcm_io_frames / kavail.tstamp.sampling_rate; tstamp->tv_sec = time; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: Fix early event callback list iteration
by Lars-Peter Clausen 23 May '13

23 May '13

The power_list field is used when adding a widget to a power sequence list. Use the same field when iterating the list using list_for_each_entry, otherwise we'll see undefined behavior. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 071579b..3507346 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1738,11 +1738,11 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) &async_domain); async_synchronize_full_domain(&async_domain); - list_for_each_entry(w, &down_list, list) { + list_for_each_entry(w, &down_list, power_list) { dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMD); } - list_for_each_entry(w, &up_list, list) { + list_for_each_entry(w, &up_list, power_list) { dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMU); } -- 1.8.2.1

2 1

[alsa-devel] [PATCH] ASoC: Add ssm2518 support
by Lars-Peter Clausen 23 May '13

23 May '13

This patch adds a ASoC CODEC driver for the SSM2516. The SSM2516 is a stereo Class-D audio amplifier with an I2S interface for audio in and a built-in dynamic range control processor. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- .../devicetree/bindings/sound/ssm2518.txt | 20 + include/linux/platform_data/ssm2518.h | 22 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2518.c | 845 +++++++++++++++++++++ sound/soc/codecs/ssm2518.h | 20 + 6 files changed, 913 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ssm2518.txt create mode 100644 include/linux/platform_data/ssm2518.h create mode 100644 sound/soc/codecs/ssm2518.c create mode 100644 sound/soc/codecs/ssm2518.h diff --git a/Documentation/devicetree/bindings/sound/ssm2518.txt b/Documentation/devicetree/bindings/sound/ssm2518.txt new file mode 100644 index 0000000..59381a7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ssm2518.txt @@ -0,0 +1,20 @@ +SSM2518 audio amplifier + +This device supports I2C only. + +Required properties: + - compatible : Must be "adi,ssm2518" + - reg : the I2C address of the device. This will either be 0x34 (ADDR pin low) + or 0x35 (ADDR pin high) + +Optional properties: + - gpios : GPIO connected to the nSD pin. If the property is not present it is + assumed that the nSD pin is hardwired to always on. + +Example: + + ssm2518: ssm2518@34 { + compatible = "adi,ssm2518"; + reg = <0x34>; + gpios = <&gpio 5 0>; + }; diff --git a/include/linux/platform_data/ssm2518.h b/include/linux/platform_data/ssm2518.h new file mode 100644 index 0000000..9a8e3ea --- /dev/null +++ b/include/linux/platform_data/ssm2518.h @@ -0,0 +1,22 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __LINUX_PLATFORM_DATA_SSM2518_H__ +#define __LINUX_PLATFORM_DATA_SSM2518_H__ + +/** + * struct ssm2518_platform_data - Platform data for the ssm2518 driver + * @enable_gpio: GPIO connected to the nSD pin. Set to -1 if the nSD pin is + * hardwired. + */ +struct ssm2518_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..d76609a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF + select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C @@ -313,6 +314,9 @@ config SND_SOC_SN95031 config SND_SOC_SPDIF tristate +config SND_SOC_SSM2518 + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dae0aa6..7204c24 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -52,6 +52,7 @@ snd-soc-si476x-objs := si476x.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o @@ -176,6 +177,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c new file mode 100644 index 0000000..5874ba5 --- /dev/null +++ b/sound/soc/codecs/ssm2518.c @@ -0,0 +1,845 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/platform_data/ssm2518.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ssm2518.h" + +#define SSM2518_REG_POWER1 0x00 +#define SSM2518_REG_CLOCK 0x01 +#define SSM2518_REG_SAI_CTRL1 0x02 +#define SSM2518_REG_SAI_CTRL2 0x03 +#define SSM2518_REG_CHAN_MAP 0x04 +#define SSM2518_REG_LEFT_VOL 0x05 +#define SSM2518_REG_RIGHT_VOL 0x06 +#define SSM2518_REG_MUTE_CTRL 0x07 +#define SSM2518_REG_FAULT_CTRL 0x08 +#define SSM2518_REG_POWER2 0x09 +#define SSM2518_REG_DRC_1 0x0a +#define SSM2518_REG_DRC_2 0x0b +#define SSM2518_REG_DRC_3 0x0c +#define SSM2518_REG_DRC_4 0x0d +#define SSM2518_REG_DRC_5 0x0e +#define SSM2518_REG_DRC_6 0x0f +#define SSM2518_REG_DRC_7 0x10 +#define SSM2518_REG_DRC_8 0x11 +#define SSM2518_REG_DRC_9 0x12 + +#define SSM2518_POWER1_RESET BIT(7) +#define SSM2518_POWER1_NO_BCLK BIT(5) +#define SSM2518_POWER1_MCS_MASK (0xf << 1) +#define SSM2518_POWER1_MCS_64FS (0x0 << 1) +#define SSM2518_POWER1_MCS_128FS (0x1 << 1) +#define SSM2518_POWER1_MCS_256FS (0x2 << 1) +#define SSM2518_POWER1_MCS_384FS (0x3 << 1) +#define SSM2518_POWER1_MCS_512FS (0x4 << 1) +#define SSM2518_POWER1_MCS_768FS (0x5 << 1) +#define SSM2518_POWER1_MCS_100FS (0x6 << 1) +#define SSM2518_POWER1_MCS_200FS (0x7 << 1) +#define SSM2518_POWER1_MCS_400FS (0x8 << 1) +#define SSM2518_POWER1_SPWDN BIT(0) + +#define SSM2518_CLOCK_ASR BIT(0) + +#define SSM2518_SAI_CTRL1_FMT_MASK (0x3 << 5) +#define SSM2518_SAI_CTRL1_FMT_I2S (0x0 << 5) +#define SSM2518_SAI_CTRL1_FMT_LJ (0x1 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_24BIT (0x2 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_16BIT (0x3 << 5) + +#define SSM2518_SAI_CTRL1_SAI_MASK (0x7 << 2) +#define SSM2518_SAI_CTRL1_SAI_I2S (0x0 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_2 (0x1 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_4 (0x2 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_8 (0x3 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_16 (0x4 << 2) +#define SSM2518_SAI_CTRL1_SAI_MONO (0x5 << 2) + +#define SSM2518_SAI_CTRL1_FS_MASK (0x3) +#define SSM2518_SAI_CTRL1_FS_8000_12000 (0x0) +#define SSM2518_SAI_CTRL1_FS_16000_24000 (0x1) +#define SSM2518_SAI_CTRL1_FS_32000_48000 (0x2) +#define SSM2518_SAI_CTRL1_FS_64000_96000 (0x3) + +#define SSM2518_SAI_CTRL2_BCLK_INTERAL BIT(7) +#define SSM2518_SAI_CTRL2_LRCLK_PULSE BIT(6) +#define SSM2518_SAI_CTRL2_LRCLK_INVERT BIT(5) +#define SSM2518_SAI_CTRL2_MSB BIT(4) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK (0x3 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_32 (0x0 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_24 (0x1 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_16 (0x2 << 2) +#define SSM2518_SAI_CTRL2_BCLK_INVERT BIT(1) + +#define SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET 4 +#define SSM2518_CHAN_MAP_RIGHT_SLOT_MASK 0xf0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET 0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_MASK 0x0f + +#define SSM2518_MUTE_CTRL_ANA_GAIN BIT(5) +#define SSM2518_MUTE_CTRL_MUTE_MASTER BIT(0) + +#define SSM2518_POWER2_APWDN BIT(0) + +#define SSM2518_DAC_MUTE BIT(6) +#define SSM2518_DAC_FS_MASK 0x07 +#define SSM2518_DAC_FS_8000 0x00 +#define SSM2518_DAC_FS_16000 0x01 +#define SSM2518_DAC_FS_32000 0x02 +#define SSM2518_DAC_FS_64000 0x03 +#define SSM2518_DAC_FS_128000 0x04 + +struct ssm2518 { + struct regmap *regmap; + bool right_j; + + unsigned int sysclk; + const struct snd_pcm_hw_constraint_list *constraints; + + int enable_gpio; +}; + +static const struct reg_default ssm2518_reg_defaults[] = { + { 0x00, 0x05 }, + { 0x01, 0x00 }, + { 0x02, 0x02 }, + { 0x03, 0x00 }, + { 0x04, 0x10 }, + { 0x05, 0x40 }, + { 0x06, 0x40 }, + { 0x07, 0x81 }, + { 0x08, 0x0c }, + { 0x09, 0x99 }, + { 0x0a, 0x7c }, + { 0x0b, 0x5b }, + { 0x0c, 0x57 }, + { 0x0d, 0x89 }, + { 0x0e, 0x8c }, + { 0x0f, 0x77 }, + { 0x10, 0x26 }, + { 0x11, 0x1c }, + { 0x12, 0x97 }, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(ssm2518_vol_tlv, -7125, 2400); +static const DECLARE_TLV_DB_SCALE(ssm2518_compressor_tlv, -3400, 200, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_expander_tlv, -8100, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_noise_gate_tlv, -9600, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_post_drc_tlv, -2400, 300, 0); + +static const DECLARE_TLV_DB_RANGE(ssm2518_limiter_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-2200, 200, 0), + 7, 15, TLV_DB_SCALE_ITEM(-800, 100, 0), +); + +static const char * const ssm2518_drc_peak_detector_attack_time_text[] = { + "0 ms", "0.1 ms", "0.19 ms", "0.37 ms", "0.75 ms", "1.5 ms", "3 ms", + "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", "192 ms", "384 ms", + "768 ms", "1536 ms", +}; + +static const char * const ssm2518_drc_peak_detector_release_time_text[] = { + "0 ms", "1.5 ms", "3 ms", "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", + "192 ms", "384 ms", "768 ms", "1536 ms", "3072 ms", "6144 ms", + "12288 ms", "24576 ms" +}; + +static const char * const ssm2518_drc_hold_time_text[] = { + "0 ms", "0.67 ms", "1.33 ms", "2.67 ms", "5.33 ms", "10.66 ms", + "21.32 ms", "42.64 ms", "85.28 ms", "170.56 ms", "341.12 ms", + "682.24 ms", "1364 ms", +}; + +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum, + SSM2518_REG_DRC_2, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum, + SSM2518_REG_DRC_2, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum, + SSM2518_REG_DRC_6, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum, + SSM2518_REG_DRC_6, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum, + SSM2518_REG_DRC_7, 4, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum, + SSM2518_REG_DRC_7, 0, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum, + SSM2518_REG_DRC_9, 0, ssm2518_drc_peak_detector_release_time_text); + +static const struct snd_kcontrol_new ssm2518_snd_controls[] = { + SOC_SINGLE("Playback De-emphasis Switch", SSM2518_REG_MUTE_CTRL, + 4, 1, 0), + SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2518_REG_LEFT_VOL, + SSM2518_REG_RIGHT_VOL, 0, 0xff, 1, ssm2518_vol_tlv), + SOC_DOUBLE("Master Playback Switch", SSM2518_REG_MUTE_CTRL, 2, 1, 1, 1), + + SOC_SINGLE("Amp Low Power Mode Switch", SSM2518_REG_POWER2, 4, 1, 0), + SOC_SINGLE("DAC Low Power Mode Switch", SSM2518_REG_POWER2, 3, 1, 0), + + SOC_SINGLE("DRC Limiter Switch", SSM2518_REG_DRC_1, 5, 1, 0), + SOC_SINGLE("DRC Compressor Switch", SSM2518_REG_DRC_1, 4, 1, 0), + SOC_SINGLE("DRC Expander Switch", SSM2518_REG_DRC_1, 3, 1, 0), + SOC_SINGLE("DRC Noise Gate Switch", SSM2518_REG_DRC_1, 2, 1, 0), + SOC_DOUBLE("DRC Switch", SSM2518_REG_DRC_1, 0, 1, 1, 0), + + SOC_SINGLE_TLV("DRC Limiter Threshold Volume", + SSM2518_REG_DRC_3, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Compressor Lower Threshold Volume", + SSM2518_REG_DRC_3, 0, 15, 1, ssm2518_compressor_tlv), + SOC_SINGLE_TLV("DRC Expander Upper Threshold Volume", SSM2518_REG_DRC_4, + 4, 15, 1, ssm2518_expander_tlv), + SOC_SINGLE_TLV("DRC Noise Gate Threshold Volume", + SSM2518_REG_DRC_4, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Upper Output Threshold Volume", + SSM2518_REG_DRC_5, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Lower Output Threshold Volume", + SSM2518_REG_DRC_5, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Post Volume", SSM2518_REG_DRC_8, + 2, 15, 1, ssm2518_post_drc_tlv), + + SOC_ENUM("DRC Peak Detector Attack Time", + ssm2518_drc_peak_detector_attack_time_enum), + SOC_ENUM("DRC Peak Detector Release Time", + ssm2518_drc_peak_detector_release_time_enum), + SOC_ENUM("DRC Attack Time", ssm2518_drc_attack_time_enum), + SOC_ENUM("DRC Decay Time", ssm2518_drc_decay_time_enum), + SOC_ENUM("DRC Hold Time", ssm2518_drc_hold_time_enum), + SOC_ENUM("DRC Noise Gate Hold Time", + ssm2518_drc_noise_gate_hold_time_enum), + SOC_ENUM("DRC RMS Averaging Time", ssm2518_drc_rms_averaging_time_enum), +}; + +static const struct snd_soc_dapm_widget ssm2518_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", SSM2518_REG_POWER2, 1, 1), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", SSM2518_REG_POWER2, 2, 1), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route ssm2518_routes[] = { + { "OUTL", NULL, "DACL" }, + { "OUTR", NULL, "DACR" }, +}; + +struct ssm2518_mcs_lut { + unsigned int rate; + const unsigned int *sysclks; +}; + +static const unsigned int ssm2518_sysclks_2048000[] = { + 2048000, 4096000, 8192000, 12288000, 16384000, 24576000, + 3200000, 6400000, 12800000, 0 +}; + +static const unsigned int ssm2518_sysclks_2822000[] = { + 2822000, 5644800, 11289600, 16934400, 22579200, 33868800, + 4410000, 8820000, 17640000, 0 +}; + +static const unsigned int ssm2518_sysclks_3072000[] = { + 3072000, 6144000, 12288000, 16384000, 24576000, 38864000, + 4800000, 9600000, 19200000, 0 +}; + +static const struct ssm2518_mcs_lut ssm2518_mcs_lut[] = { + { 8000, ssm2518_sysclks_2048000, }, + { 11025, ssm2518_sysclks_2822000, }, + { 12000, ssm2518_sysclks_3072000, }, + { 16000, ssm2518_sysclks_2048000, }, + { 24000, ssm2518_sysclks_3072000, }, + { 22050, ssm2518_sysclks_2822000, }, + { 32000, ssm2518_sysclks_2048000, }, + { 44100, ssm2518_sysclks_2822000, }, + { 48000, ssm2518_sysclks_3072000, }, + { 96000, ssm2518_sysclks_3072000, }, +}; + +static const unsigned int ssm2518_rates_2048000[] = { + 8000, 16000, 32000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2048000 = { + .list = ssm2518_rates_2048000, + .count = ARRAY_SIZE(ssm2518_rates_2048000), +}; + +static const unsigned int ssm2518_rates_2822000[] = { + 11025, 22050, 44100, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2822000 = { + .list = ssm2518_rates_2822000, + .count = ARRAY_SIZE(ssm2518_rates_2822000), +}; + +static const unsigned int ssm2518_rates_3072000[] = { + 12000, 24000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_3072000 = { + .list = ssm2518_rates_3072000, + .count = ARRAY_SIZE(ssm2518_rates_3072000), +}; + +static const unsigned int ssm2518_rates_12288000[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_12288000 = { + .list = ssm2518_rates_12288000, + .count = ARRAY_SIZE(ssm2518_rates_12288000), +}; + +static unsigned int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, + unsigned int rate) +{ + const unsigned int *sysclks = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(ssm2518_mcs_lut); i++) { + if (ssm2518_mcs_lut[i].rate == rate) { + sysclks = ssm2518_mcs_lut[i].sysclks; + break; + } + } + if (!sysclks) + return -EINVAL; + + for (i = 0; sysclks[i]; i++) { + if (sysclks[i] == ssm2518->sysclk) + return i; + } + return -EINVAL; +} + +static int ssm2518_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int ctrl1, ctrl1_mask; + int mcs; + int ret; + + mcs = ssm2518_lookup_mcs(ssm2518, rate); + if (mcs < 0) + return mcs; + + ctrl1_mask = SSM2518_SAI_CTRL1_FS_MASK; + + if (rate >= 8000 && rate <= 12000) + ctrl1 = SSM2518_SAI_CTRL1_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + ctrl1 = SSM2518_SAI_CTRL1_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + ctrl1 = SSM2518_SAI_CTRL1_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + ctrl1 = SSM2518_SAI_CTRL1_FS_64000_96000; + else + return -EINVAL; + + if (ssm2518->right_j) { + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_16BIT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + break; + default: + return -EINVAL; + } + ctrl1_mask |= SSM2518_SAI_CTRL1_FMT_MASK; + } + + /* Disable auto samplerate detection */ + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_CLOCK, + SSM2518_CLOCK_ASR, SSM2518_CLOCK_ASR); + if (ret < 0) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + ctrl1_mask, ctrl1); + if (ret < 0) + return ret; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_MCS_MASK, mcs << 1); +} + +static int ssm2518_mute(struct snd_soc_dai *dai, int mute) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (mute) + val = SSM2518_MUTE_CTRL_MUTE_MASTER; + else + val = 0; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_MUTE_CTRL, + SSM2518_MUTE_CTRL_MUTE_MASTER, val); +} + +static int ssm2518_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1 = 0, ctrl2 = 0; + bool invert_fclk; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_fclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_fclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = true; + break; + default: + return -EINVAL; + } + + ssm2518->right_j = false; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + ssm2518->right_j = true; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = false; + break; + default: + return -EINVAL; + } + + if (invert_fclk) + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_INVERT; + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, ctrl1); + if (ret) + return ret; + return regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, ctrl2); +} + +static int ssm2518_set_power(struct ssm2518 *ssm2518, bool enable) +{ + int ret = 0; + + if (!enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN, SSM2518_POWER1_SPWDN); + regcache_mark_dirty(ssm2518->regmap); + } + + if (gpio_is_valid(ssm2518->enable_gpio)) + gpio_set_value(ssm2518->enable_gpio, enable); + + regcache_cache_only(ssm2518->regmap, !enable); + + if (enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN | SSM2518_POWER1_RESET, 0x00); + regcache_sync(ssm2518->regmap); + } + + return ret; +} + +static int ssm2518_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = ssm2518_set_power(ssm2518, true); + break; + case SND_SOC_BIAS_OFF: + ret = ssm2518_set_power(ssm2518, false); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + return 0; +} + +static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1, ctrl2; + int left_slot, right_slot; + int ret; + + if (slots == 0) { + return regmap_update_bits(ssm2518->regmap, + SSM2518_REG_SAI_CTRL1, SSM2518_SAI_CTRL1_SAI_MASK, + SSM2518_SAI_CTRL1_SAI_I2S); + } + + if (tx_mask == 0 || tx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + /* We assume the left channel < right channel */ + left_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + switch (width) { + case 16: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_16; + break; + case 24: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_24; + break; + case 32: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_32; + break; + default: + return -EINVAL; + } + + switch (slots) { + case 1: + ctrl1 = SSM2518_SAI_CTRL1_SAI_MONO; + break; + case 2: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_2; + break; + case 4: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_4; + break; + case 8: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_8; + break; + case 16: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_16; + break; + default: + return -EINVAL; + } + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_CHAN_MAP, + (left_slot << SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET) | + (right_slot << SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET)); + if (ret) + return ret; + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + SSM2518_SAI_CTRL1_SAI_MASK, ctrl1); + if (ret) + return ret; + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, + SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK, ctrl2); +} + +static int ssm2518_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + + if (ssm2518->constraints) { + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, ssm2518->constraints); + } + + return 0; +} + +#define SSM2518_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ssm2518_dai_ops = { + .startup = ssm2518_startup, + .hw_params = ssm2518_hw_params, + .digital_mute = ssm2518_mute, + .set_fmt = ssm2518_set_dai_fmt, + .set_tdm_slot = ssm2518_set_tdm_slot, +}; + +static struct snd_soc_dai_driver ssm2518_dai = { + .name = "ssm2518-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SSM2518_FORMATS, + }, + .ops = &ssm2518_dai_ops, +}; + +static int ssm2518_probe(struct snd_soc_codec *codec) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = ssm2518->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER2, + SSM2518_POWER2_APWDN, 0x00); + if (ret) + return ret; + + return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int ssm2518_remove(struct snd_soc_codec *codec) +{ + ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (clk_id != SSM2518_SYSCLK) + return -EINVAL; + + switch (source) { + case SSM2518_SYSCLK_SRC_MCLK: + val = 0; + break; + case SSM2518_SYSCLK_SRC_BCLK: + /* In this case the bitclock is used as the system clock, and + * the bitclock signal needs to be connected to the MCLK pin and + * the BCLK pin is left unconnected */ + val = SSM2518_POWER1_NO_BCLK; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 2048000: + case 4096000: + case 8192000: + case 3200000: + case 6400000: + case 12800000: + ssm2518->constraints = &ssm2518_constraints_2048000; + break; + case 2822000: + case 5644800: + case 11289600: + case 16934400: + case 22579200: + case 33868800: + case 4410000: + case 8820000: + case 17640000: + ssm2518->constraints = &ssm2518_constraints_2822000; + break; + case 3072000: + case 6144000: + case 38864000: + case 4800000: + case 9600000: + case 19200000: + ssm2518->constraints = &ssm2518_constraints_3072000; + break; + case 12288000: + case 16384000: + case 24576000: + ssm2518->constraints = &ssm2518_constraints_12288000; + break; + default: + return -EINVAL; + } + + ssm2518->sysclk = freq; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_NO_BCLK, val); +} + +static struct snd_soc_codec_driver ssm2518_codec_driver = { + .probe = ssm2518_probe, + .remove = ssm2518_remove, + .set_bias_level = ssm2518_set_bias_level, + .set_sysclk = ssm2518_set_sysclk, + .idle_bias_off = true, + + .controls = ssm2518_snd_controls, + .num_controls = ARRAY_SIZE(ssm2518_snd_controls), + .dapm_widgets = ssm2518_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2518_dapm_widgets), + .dapm_routes = ssm2518_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2518_routes), +}; + +static bool ssm2518_register_volatile(struct device *dev, unsigned int reg) +{ + return false; +} + +static const struct regmap_config ssm2518_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = SSM2518_REG_DRC_9, + .volatile_reg = ssm2518_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ssm2518_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ssm2518_reg_defaults), +}; + +static int ssm2518_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm2518_platform_data *pdata = i2c->dev.platform_data; + struct ssm2518 *ssm2518; + int ret; + + ssm2518 = devm_kzalloc(&i2c->dev, sizeof(*ssm2518), GFP_KERNEL); + if (ssm2518 == NULL) + return -ENOMEM; + + if (pdata) { + ssm2518->enable_gpio = pdata->enable_gpio; + } else if (i2c->dev.of_node) { + ssm2518->enable_gpio = of_get_gpio(i2c->dev.of_node, 0); + if (ssm2518->enable_gpio == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { + ssm2518->enable_gpio = -1; + } + + if (gpio_is_valid(ssm2518->enable_gpio)) { + ret = devm_gpio_request_one(&i2c->dev, ssm2518->enable_gpio, + GPIOF_OUT_INIT_HIGH, "SSM2518 nSD"); + if (ret) + return ret; + } + + i2c_set_clientdata(i2c, ssm2518); + + ssm2518->regmap = devm_regmap_init_i2c(i2c, &ssm2518_regmap_config); + if (IS_ERR(ssm2518->regmap)) + return PTR_ERR(ssm2518->regmap); + + /* + * The reset bit is obviously volatile, but we need to be able to cache + * the other bits in the register, so we can't just mark the whole + * register as volatile. Since this is the only place where we'll ever + * touch the reset bit just bypass the cache for this operation. + */ + regcache_cache_bypass(ssm2518->regmap, true); + ret = regmap_write(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_RESET); + if (ret) + return ret; + regcache_cache_bypass(ssm2518->regmap, false); + + return snd_soc_register_codec(&i2c->dev, &ssm2518_codec_driver, + &ssm2518_dai, 1); +} + +static int ssm2518_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm2518_i2c_ids[] = { + { "ssm2518", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids); + +static struct i2c_driver ssm2518_driver = { + .driver = { + .name = "ssm2518", + .owner = THIS_MODULE, + }, + .probe = ssm2518_i2c_probe, + .remove = ssm2518_i2c_remove, + .id_table = ssm2518_i2c_ids, +}; +module_i2c_driver(ssm2518_driver); + +MODULE_DESCRIPTION("ASoC SSM2518 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2518.h b/sound/soc/codecs/ssm2518.h new file mode 100644 index 0000000..62511d8 --- /dev/null +++ b/sound/soc/codecs/ssm2518.h @@ -0,0 +1,20 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __SND_SOC_CODECS_SSM2518_H__ +#define __SND_SOC_CODECS_SSM2518_H__ + +#define SSM2518_SYSCLK 0 + +enum ssm2518_sysclk_src { + SSM2518_SYSCLK_SRC_MCLK = 0, + SSM2518_SYSCLK_SRC_BCLK = 1, +}; + +#endif -- 1.8.0

2 3

[alsa-devel] [PATCH] ALSA: ASoC: add codec driver for TI TAS5086
by Daniel Mack 22 May '13

22 May '13

This patch adds a driver for TI's TA5086 6-channel PWM processor. This chip has a very unusual register layout, specifically because the registers are of unequal size, and multi-byte registers require bulk writes to take effect. Regmap does not support these kind of mappings. Currently, the driver does not touch any of the registers >= 0x20, so it doesn't matter, because the register map is mapped to an 8-bit array. In case more features will be added in the future that require access to higher registers, the entire regmap H/W I/O routines have to be open-coded. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- .../devicetree/bindings/sound/ti,tas5086.txt | 32 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5086.c | 587 +++++++++++++++++++++ 4 files changed, 625 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas5086.txt create mode 100644 sound/soc/codecs/tas5086.c diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt new file mode 100644 index 0000000..8ea4f5b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -0,0 +1,32 @@ +Texas Instruments TAS5086 6-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,tas5086". + - reg: The i2c address. Should contain <0x1b>. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + + - ti,charge-period: This property should contain the time in microseconds + that closely matches the external single-ended + split-capacitor charge period. The hardware chip + waits for this period of time before starting the + PWM signals. This helps reduce pops and clicks. + + When not specified, the hardware default of 1300ms + is retained. + +Examples: + + i2c_bus { + tas5086@1b { + compatible = "ti,tas5086"; + reg = <0x1b>; + reset-gpio = <&gpio 23 0>; + ti,charge-period = <156000>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 45b7256..86b3524 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -63,6 +63,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS + select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C @@ -320,6 +321,9 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate +config SND_SOC_TAS5086 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a3b3c3..8077bc2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o +snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -177,6 +178,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o +obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c new file mode 100644 index 0000000..cd3c267 --- /dev/null +++ b/sound/soc/codecs/tas5086.c @@ -0,0 +1,587 @@ +/* + * TAS5086 ASoC codec driver + * + * Copyright (c) 2013 Daniel Mack <zonque(a)gmail.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * TODO: + * - implement DAPM and input muxing + * - implement modulation limit + * - implement non-default PWM start + * + * Note that this chip has a very unusual register layout, specifically + * because the registers are of unequal size, and multi-byte registers + * require bulk writes to take effect. Regmap does not support that kind + * of devices. + * + * Currently, the driver does not touch any of the registers >= 0x20, so + * it doesn't matter because the entire map can be accessed as 8-bit + * array. In case more features will be added in the future + * that require access to higher registers, the entire regmap H/W I/O + * routines have to be open-coded. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define TAS5086_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE) + +#define TAS5086_PCM_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +/* + * TAS5086 registers + */ +#define TAS5086_CLOCK_CONTROL 0x00 /* Clock control register */ +#define TAS5086_CLOCK_RATE(val) (val << 5) +#define TAS5086_CLOCK_RATE_MASK (0x7 << 5) +#define TAS5086_CLOCK_RATIO(val) (val << 2) +#define TAS5086_CLOCK_RATIO_MASK (0x7 << 2) +#define TAS5086_CLOCK_SCLK_RATIO_48 (1 << 1) +#define TAS5086_CLOCK_VALID (1 << 0) + +#define TAS5086_DEV_ID 0x01 /* Device ID register */ +#define TAS5086_ERROR_STATUS 0x02 /* Error status register */ +#define TAS5086_SYS_CONTROL_1 0x03 /* System control register 1 */ +#define TAS5086_SERIAL_DATA_IF 0x04 /* Serial data interface register */ +#define TAS5086_SYS_CONTROL_2 0x05 /* System control register 2 */ +#define TAS5086_SOFT_MUTE 0x06 /* Soft mute register */ +#define TAS5086_MASTER_VOL 0x07 /* Master volume */ +#define TAS5086_CHANNEL_VOL(X) (0x08 + (X)) /* Channel 1-6 volume */ +#define TAS5086_VOLUME_CONTROL 0x09 /* Volume control register */ +#define TAS5086_MOD_LIMIT 0x10 /* Modulation limit register */ +#define TAS5086_PWM_START 0x18 /* PWM start register */ +#define TAS5086_SURROUND 0x19 /* Surround register */ +#define TAS5086_SPLIT_CAP_CHARGE 0x1a /* Split cap charge period register */ +#define TAS5086_OSC_TRIM 0x1b /* Oscillator trim register */ +#define TAS5086_BKNDERR 0x1c + +#define TAS5086_DEEMPH_MASK 0x03 + +/* + * Default TAS5086 power-up configuration + */ +static const struct reg_default tas5086_reg_defaults[] = { + { 0x00, 0x6c }, + { 0x01, 0x03 }, + { 0x02, 0x00 }, + { 0x03, 0xa0 }, + { 0x04, 0x05 }, + { 0x05, 0x60 }, + { 0x06, 0x00 }, + { 0x07, 0xff }, + { 0x08, 0x30 }, + { 0x09, 0x30 }, + { 0x0a, 0x30 }, + { 0x0b, 0x30 }, + { 0x0c, 0x30 }, + { 0x0d, 0x30 }, + { 0x0e, 0xb1 }, + { 0x0f, 0x00 }, + { 0x10, 0x02 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x3f }, + { 0x19, 0x00 }, + { 0x1a, 0x18 }, + { 0x1b, 0x82 }, + { 0x1c, 0x05 }, +}; + +static bool tas5086_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17)); +} + +static bool tas5086_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == TAS5086_DEV_ID || + reg == TAS5086_ERROR_STATUS; +} + +static bool tas5086_writeable_reg(struct device *dev, unsigned int reg) +{ + return tas5086_accessible_reg(dev, reg) && (reg != TAS5086_DEV_ID); +} + +struct tas5086_private { + struct regmap *regmap; + unsigned int mclk, sclk; + unsigned int format; + bool deemph; + /* Current sample rate for de-emphasis control */ + int rate; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; +}; + +static int tas5086_deemph[] = { 0, 32000, 44100, 48000 }; + +static int tas5086_set_deemph(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int i, val = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++) + if (tas5086_deemph[i] == priv->rate) + val = i; + + return regmap_update_bits(priv->regmap, TAS5086_SYS_CONTROL_1, + TAS5086_DEEMPH_MASK, val); +} + +static int tas5086_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int tas5086_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return tas5086_set_deemph(codec); +} + + +static int tas5086_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case 0: /* MCLK */ + priv->mclk = freq; + break; + case 1: /* SCLK */ + priv->sclk = freq; + break; + } + + return 0; +} + +static int tas5086_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The TAS5086 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + /* we need to refer to the data format from hw_params() */ + priv->format = format; + + return 0; +} + +static const int tas5086_sample_rates[] = { + 32000, 38000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +static const int tas5086_ratios[] = { + 64, 128, 192, 256, 384, 512 +}; + +static int index_in_array(const int *array, int len, int needle) +{ + int i; + + for (i = 0; i < len; i++) + if (array[i] == needle) + return i; + + return -ENOENT; +} + +static int tas5086_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int ret; + + priv->rate = params_rate(params); + + /* Look up the sample rate and refer to the offset in the list */ + val = index_in_array(tas5086_sample_rates, + ARRAY_SIZE(tas5086_sample_rates), priv->rate); + + if (val < 0) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATE_MASK, + TAS5086_CLOCK_RATE(val)); + if (ret < 0) + return ret; + + /* MCLK / Fs ratio */ + val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios), + priv->mclk / priv->rate); + if (val < 0) { + dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATIO_MASK, + TAS5086_CLOCK_RATIO(val)); + if (ret < 0) + return ret; + + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_SCLK_RATIO_48, + (priv->sclk == 48 * priv->rate) ? + TAS5086_CLOCK_SCLK_RATIO_48 : 0); + if (ret < 0) + return ret; + + /* + * The chip has a very unituitive register mapping and muxes information + * about data format and sample depth into the same register, but not on + * a logical bit-boundary. Hence, we have to refer to the format passed + * in the set_dai_fmt() callback and set up everything from here. + * + * First, determine the 'base' value, using the format ... + */ + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x03; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x06; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* ... then add the offset for the sample bit depth. */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val += 0; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val += 1; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + val += 2; + break; + default: + dev_err(codec->dev, "Invalid bit width\n"); + return -EINVAL; + }; + + ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val); + if (ret < 0) + return ret; + + /* clock is considered valid now */ + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_VALID, TAS5086_CLOCK_VALID); + if (ret < 0) + return ret; + + return tas5086_set_deemph(codec); +} + +static int tas5086_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, + mute ? 0x3f : 0x00); +} + +/* TAS5086 controls */ +static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new tas5086_controls[] = { + SOC_SINGLE_TLV("Master Playback Volume", TAS5086_MASTER_VOL, + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + TAS5086_CHANNEL_VOL(0), TAS5086_CHANNEL_VOL(1), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + TAS5086_CHANNEL_VOL(2), TAS5086_CHANNEL_VOL(3), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + TAS5086_CHANNEL_VOL(4), TAS5086_CHANNEL_VOL(5), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + tas5086_get_deemph, tas5086_put_deemph), +}; + +static const struct snd_soc_dai_ops tas5086_dai_ops = { + .hw_params = tas5086_hw_params, + .set_sysclk = tas5086_set_dai_sysclk, + .set_fmt = tas5086_set_dai_fmt, + .digital_mute = tas5086_digital_mute, +}; + +static struct snd_soc_dai_driver tas5086_dai = { + .name = "tas5086-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = TAS5086_PCM_RATES, + .formats = TAS5086_PCM_FORMATS, + }, + .ops = &tas5086_dai_ops, +}; + +#ifdef CONFIG_PM +static int tas5086_soc_suspend(struct snd_soc_codec *codec) +{ + return 0; +} + +static int tas5086_soc_resume(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* Restore codec state */ + return regcache_sync(priv->regmap); +} +#else +#define tas5086_soc_suspend NULL +#define tas5086_soc_resume NULL +#endif /* CONFIG_PM */ + +#ifdef CONFIG_OF +static const struct of_device_id tas5086_dt_ids[] = { + { .compatible = "ti,tas5086", }, + { } +}; +MODULE_DEVICE_TABLE(of, tas5086_dt_ids); +#endif + +/* charge period values in microseconds */ +static const int tas5086_charge_period[] = { + 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, + 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, + 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, +}; + +static int tas5086_probe(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int charge_period = 1300000; /* hardware default is 1300 ms */ + int gpio_nreset = -EINVAL; + int i, ret; + + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { + struct device_node *of_node = codec->dev->of_node; + + gpio_nreset = of_get_named_gpio(of_node, "reset-gpio", 0); + of_property_read_u32(of_node, "ti,charge-period", &charge_period); + } + + if (gpio_nreset >= 0) + if (devm_gpio_request(codec->dev, gpio_nreset, "TAS5086 Reset")) + gpio_nreset = -EINVAL; + if (gpio_nreset >= 0) { + /* Reset codec - minimum assertion time is 400ns */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + + /* Codec needs ~15ms to wake up */ + msleep(15); + } +msleep(300); + priv->gpio_nreset = gpio_nreset; + + /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ + ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); + if (ret < 0) + return ret; + + if (i != 0x3) { + dev_err(codec->dev, + "Failed to identify TAS5086 codec (got %02x)\n", i); + return -ENODEV; + } + + /* lookup and set split-capacitor charge period */ + if (charge_period == 0) { + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); + } else { + i = index_in_array(tas5086_charge_period, + ARRAY_SIZE(tas5086_charge_period), + charge_period); + if (i >= 0) + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, + i + 0x08); + else + dev_warn(codec->dev, + "Invalid split-cap charge period of %d ns.\n", + charge_period); + } + + /* enable factory trim */ + ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); + if (ret < 0) + return ret; + + /* start all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + if (ret < 0) + return ret; + + /* set master volume to 0 dB */ + ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); + if (ret < 0) + return ret; + + return 0; +} + +static int tas5086_remove(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(priv->gpio_nreset)) + /* Set codec to the reset state */ + gpio_set_value(priv->gpio_nreset, 0); + + return 0; +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { + .probe = tas5086_probe, + .remove = tas5086_remove, + .suspend = tas5086_soc_suspend, + .resume = tas5086_soc_resume, + .controls = tas5086_controls, + .num_controls = ARRAY_SIZE(tas5086_controls), +}; + +static const struct i2c_device_id tas5086_i2c_id[] = { + { "tas5086", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id); + +static const struct regmap_config tas5086_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(tas5086_reg_defaults), + .reg_defaults = tas5086_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5086_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5086_volatile_reg, + .writeable_reg = tas5086_writeable_reg, + .readable_reg = tas5086_accessible_reg, +}; + +static int tas5086_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tas5086_private *priv; + int ret; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(i2c, &tas5086_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, priv); + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, + &tas5086_dai, 1); +} + +static int tas5086_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static struct i2c_driver tas5086_i2c_driver = { + .driver = { + .name = "tas5086", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas5086_dt_ids), + }, + .id_table = tas5086_i2c_id, + .probe = tas5086_i2c_probe, + .remove = tas5086_i2c_remove, +}; + +static int __init tas5086_modinit(void) +{ + return i2c_add_driver(&tas5086_i2c_driver); +} +module_init(tas5086_modinit); + +static void __exit tas5086_modexit(void) +{ + i2c_del_driver(&tas5086_i2c_driver); +} +module_exit(tas5086_modexit); + +MODULE_AUTHOR("Daniel Mack <zonque(a)gmail.com>"); +MODULE_DESCRIPTION("Texas Instruments TAS5086 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); -- 1.8.1.4

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Re: [alsa-devel] [PATCH 3/3] sound/soc/fsl: remove leftover release_mem_region
by Mark Brown 22 May '13

22 May '13

On Wed, May 22, 2013 at 06:04:00PM +0200, Wolfram Sang wrote: > On Wed, May 22, 2013 at 10:56:28AM -0500, Mark Brown wrote: > > This doesn't apply against current code, can you please regenerate > > against topic/fsl or mainline? > ? Works fine against v3.10-rc2 here. Or is mainline v3.9 for you? To be honest I didn't actually try mainline as I wasn't expecting any conflicts from the branch and it didn't seem urgent as a bug fix.

1 0

[alsa-devel] [PATCH v4] ASoC: dfbmcs320: make the driver common for other BT modules
by Barry Song 22 May '13

22 May '13

DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, we don't want everyone to have a seperate codec driver. anyway, the feature of Bluetooth SCO is same on all platforms, so this patch makes the DFBM-CS320 driver become a common BT SCO link driver. Cc: Graeme Gregory <graeme(a)openmoko.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- v4: change "select SND_SOC_DFBMCS320" to "select SND_SOC_BT_SCO" for OpenMOKO user as v3 missed this sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +++++++++++++++++++----------- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- 5 files changed, 29 insertions(+), 20 deletions(-) rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d8c4f3d..386292d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DFBMCS320 + select SND_SOC_BT_SCO select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -263,7 +263,7 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DFBMCS320 +config SND_SOC_BT_SCO tristate config SND_SOC_DMIC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 49ff127..82aa34e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,7 +27,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o -snd-soc-dfbmcs320-objs := dfbmcs320.o +snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o @@ -154,7 +154,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o -obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o +obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/bt-sco.c similarity index 53% rename from sound/soc/codecs/dfbmcs320.c rename to sound/soc/codecs/bt-sco.c index 4f4f7f4..a081d9f 100644 --- a/sound/soc/codecs/dfbmcs320.c +++ b/sound/soc/codecs/bt-sco.c @@ -1,5 +1,5 @@ /* - * Driver for the DFBM-CS320 bluetooth module + * Driver for generic Bluetooth SCO link * Copyright 2011 Lars-Peter Clausen <lars(a)metafoo.de> * * This program is free software; you can redistribute it and/or modify it @@ -15,8 +15,8 @@ #include <sound/soc.h> -static struct snd_soc_dai_driver dfbmcs320_dai = { - .name = "dfbmcs320-pcm", +static struct snd_soc_dai_driver bt_sco_dai = { + .name = "bt-sco-pcm", .playback = { .channels_min = 1, .channels_max = 1, @@ -31,32 +31,41 @@ static struct snd_soc_dai_driver dfbmcs320_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320; +static struct snd_soc_codec_driver soc_codec_dev_bt_sco; -static int dfbmcs320_probe(struct platform_device *pdev) +static int bt_sco_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320, - &dfbmcs320_dai, 1); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco, + &bt_sco_dai, 1); } -static int dfbmcs320_remove(struct platform_device *pdev) +static int bt_sco_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver dfmcs320_driver = { +static struct platform_device_id bt_sco_driver_ids[] = { + { + .name = "dfbmcs320", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); + +static struct platform_driver bt_sco_driver = { .driver = { - .name = "dfbmcs320", + .name = "bt-sco", .owner = THIS_MODULE, }, - .probe = dfbmcs320_probe, - .remove = dfbmcs320_remove, + .probe = bt_sco_probe, + .remove = bt_sco_remove, + .id_table = bt_sco_driver_ids, }; -module_platform_driver(dfmcs320_driver); +module_platform_driver(bt_sco_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); +MODULE_DESCRIPTION("ASoC generic bluethooth sco link driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 475fb0d..ae0ea87 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -39,7 +39,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 select SND_S3C24XX_I2S select SND_SOC_WM8753 - select SND_SOC_DFBMCS320 + select SND_SOC_SCO help Say Y here to enable audio support for the Openmoko Neo1973 Smartphones. diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index e591c38..807db41 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -373,7 +373,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Voice via BT */ .name = "Bluetooth", .stream_name = "Voice", - .cpu_dai_name = "dfbmcs320-pcm", + .cpu_dai_name = "bt-sco-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, -- 1.8.2.3

3 4

Re: [alsa-devel] [PATCH 3/3] sound/soc/fsl: remove leftover release_mem_region
by Mark Brown 22 May '13

22 May '13

On Wed, May 22, 2013 at 03:21:26PM +0200, Wolfram Sang wrote: > When converting this driver to devm_ioremap_resource, the removal of this now > unneeded function has been forgotten. This doesn't apply against current code, can you please regenerate against topic/fsl or mainline?

1 0

[alsa-devel] [PATCH] ALSA: hda - Enable runtime pm for Haswell
by Wang Xingchao 22 May '13

22 May '13

Haswell doesnot support runtime pm by default. This patch let haswell Display HD-A controller enter runtime suspend, and bring more power saving whith power-well. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bf27693..eb25888 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3755,6 +3755,12 @@ static int azx_probe(struct pci_dev *pci, if (pci_dev_run_wake(pci)) pm_runtime_put_noidle(&pci->dev); + else if (chip->driver_caps + & AZX_DCAPS_I915_POWERWELL) { + /* Haswell doesnot support runtime pm by default */ + pm_runtime_put_noidle(&pci->dev); + pm_runtime_allow(&pci->dev); + } dev++; complete_all(&chip->probe_wait); -- 1.7.9.5

3 4

[alsa-devel] [PATCH] ASoC: davinci: fix sample rotation
by Daniel Mack 21 May '13

21 May '13

McASP serial audio engine needs different rotation values on TX and RX channels. Commit dde109fb462 ("ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback") changed the calculation to fix the playback format, but broke the capture stream by doing it for both TXFMT and RXFMT. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Michal Bachraty <michal.bachraty(a)streamunlimited.com> Cc: stable(a)vger.kernel.org [3.9 only] --- sound/soc/davinci/davinci-mcasp.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 63633e9..8d805ec 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -628,7 +628,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int word_length) { u32 fmt; - u32 rotate = (word_length / 4) & 0x7; + u32 tx_rotate = (word_length / 4) & 0x7; + u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; /* @@ -651,9 +652,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXROT(rotate), TXROT(7)); + TXROT(tx_rotate), TXROT(7)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXROT(rotate), RXROT(7)); + RXROT(rx_rotate), RXROT(7)); mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); } -- 1.8.1.4

2 1

[alsa-devel] [PATCH 0/2] Set ab8500-codec dai slots from machine drivers
by Fabio Baltieri 21 May '13

21 May '13

Hi Mark, these two patches rework the slot selection code in ab8500-codec to use the slots masks provided by the machine drivers instead of the hardcoded ones as suggested. The patches replaces some of the previous macros with a parametrized version in the effort of making the actual code compact and readable, and should be applied in order. Also, the second patch drops the hardcoded defintions swapped in -rc2, so you may want to rebase the ux500 topic branch to avoid the conflict, or I can resend the patch based on the topic branch directly. Just let me know. Big thanks to Ola for helping me figuring out the weird ab8500 slot mapping logic. Fabio Fabio Baltieri (2): ASoC: ab8500-codec: Set tx dai slots from tx_mask ASoC: ab8500-codec: Set rx dai slots from rx_mask sound/soc/codecs/ab8500-codec.c | 60 ++++++++++++++++++++++++++--------------- sound/soc/codecs/ab8500-codec.h | 42 +++++++++++++++-------------- 2 files changed, 61 insertions(+), 41 deletions(-) -- 1.8.2

2 5

[alsa-devel] [PATCH 0/2 V4] Power-well API implementation for Haswell
by Wang Xingchao 21 May '13

21 May '13

Hi all, This is V4 and here're some changes notes: change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (2): i915/drm: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA drivers/gpu/drm/i915/i915_dma.c | 6 +++ drivers/gpu/drm/i915/i915_drv.h | 12 +++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 98 +++++++++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 ++++++++++++++ sound/pci/hda/Kconfig | 10 ++++ sound/pci/hda/Makefile | 3 ++ sound/pci/hda/hda_i915.c | 75 +++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 ++++++++++++++ sound/pci/hda/hda_intel.c | 41 ++++++++++++++-- 10 files changed, 310 insertions(+), 10 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 7

Re: [alsa-devel] [Alsa-devel] add support for syba 7.1 surround sound card
by Clemens Ladisch 21 May '13

21 May '13

good1.2guy(a)gmail.com wrote: > Summary: add support for syba 7.1 surround sound card > Keywords: syba via VT1723 Tremor > Kernel: 3.8.8-203.fc18.x86_64 What are these three lines for? > add support for syba 7.1 surround sound card > actually the card only has 6 dmas, so it is really only 5.1 The website shows a daughterboard. > added code to force_rate setting to not fail EBUSY when the dmas are running. > suggest using force_rate with pulseaudio to avoid loops with error logging. This sounds as if it has nothing to do with this particular SYBA card but affects all VT1723 cards. Please move it into a separate patch. And there should be no reason for a module parameter; if this is a bug in the driver and/or in PulseAudio, it should be just fixed. > Signed-off-by: psmith2004(a)yahoo.com Real names only. Please read Documentation/SubmittingPatches. > +static void syba_quirk_open(struct snd_ice1712 *ice) > +{ > + unsigned long flags; It's common to have an empty line here. > + spin_lock_irqsave(&ice->reg_lock, flags); > + outb(0xff, ICEREG1724(ice, AC97_CFG)); > + udelay(500); > + outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG)); > + spin_unlock_irqrestore(&ice->reg_lock, flags); > +} What is the reason for this quirk? Is it possible that this might be needed for other cards using a AC'97 codec? Regards, Clemens

1 0

[alsa-devel] what is the difference between SNDRV_PCM_IOCTL_WRITEN_FRAMES and SNDRV_PCM_IOCTL_WRITEI_FRAMES
by Zhang wei 21 May '13

21 May '13

I know the format likes below when use SNDRV_PCM_IOCTL_WRITEI_FRAMES <http://lxr.free-electrons.com/ident?i=SNDRV_PCM_IOCTL_WRITEN_FRAMES> L+R+L+R+L+R... L:a frame left channel data. R:a frame right channel data. Right? But what is the kind of format when use SNDRV_PCM_IOCTL_WRITEN_FRAMES <http://lxr.free-electrons.com/ident?i=SNDRV_PCM_IOCTL_WRITEN_FRAMES> ? Thank you:-) Br zw

2 4

[alsa-devel] [PATCH] ASoC: wm8994: Handle LRCLK inversion for WM8958 and WM1811A
by Mark Brown 20 May '13

20 May '13

On WM8958 and WM1811A separate control of the LRCLK inversion bit is available for the DAC and ADC LRCLKs which for compatibility reasons is done in a new register bit. Since writes to each scheme have no effect on parts using the other just always write to both for simplicity. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Tested-by: Samreen Nilofer <samreen.nilofer(a)intel.com> --- include/linux/mfd/wm8994/registers.h | 8 ++++++++ sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index 0535489..db8cef3 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -2668,6 +2668,10 @@ /* * R772 (0x304) - AIF1ADC LRCLK */ +#define WM8958_AIF1_LRCLK_INV 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_MASK 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_SHIFT 12 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_WIDTH 1 /* AIF1_LRCLK_INV */ #define WM8994_AIF1ADC_LRCLK_DIR 0x0800 /* AIF1ADC_LRCLK_DIR */ #define WM8994_AIF1ADC_LRCLK_DIR_MASK 0x0800 /* AIF1ADC_LRCLK_DIR */ #define WM8994_AIF1ADC_LRCLK_DIR_SHIFT 11 /* AIF1ADC_LRCLK_DIR */ @@ -2679,6 +2683,10 @@ /* * R773 (0x305) - AIF1DAC LRCLK */ +#define WM8958_AIF1_LRCLK_INV 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_MASK 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_SHIFT 12 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_WIDTH 1 /* AIF1_LRCLK_INV */ #define WM8994_AIF1DAC_LRCLK_DIR 0x0800 /* AIF1DAC_LRCLK_DIR */ #define WM8994_AIF1DAC_LRCLK_DIR_MASK 0x0800 /* AIF1DAC_LRCLK_DIR */ #define WM8994_AIF1DAC_LRCLK_DIR_SHIFT 11 /* AIF1DAC_LRCLK_DIR */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index cb67603..0bcbd12 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2574,17 +2574,24 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct wm8994 *control = wm8994->wm8994; int ms_reg; int aif1_reg; + int dac_reg; + int adc_reg; int ms = 0; int aif1 = 0; + int lrclk = 0; switch (dai->id) { case 1: ms_reg = WM8994_AIF1_MASTER_SLAVE; aif1_reg = WM8994_AIF1_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; case 2: ms_reg = WM8994_AIF2_MASTER_SLAVE; aif1_reg = WM8994_AIF2_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; default: return -EINVAL; @@ -2603,6 +2610,7 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; @@ -2641,12 +2649,14 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_IB_IF: aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; case SND_SOC_DAIFMT_IB_NF: aif1 |= WM8994_AIF1_BCLK_INV; break; case SND_SOC_DAIFMT_NB_IF: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; default: return -EINVAL; @@ -2677,6 +2687,10 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) aif1); snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR, ms); + snd_soc_update_bits(codec, dac_reg, + WM8958_AIF1_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, adc_reg, + WM8958_AIF1_LRCLK_INV, lrclk); return 0; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2] ASoC: soc-compress: Send correct stream event for capture start
by Charles Keepax 20 May '13

20 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3853f7e..06a8000 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, goto err; } - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_START); + if (cstream->direction == SND_COMPRESS_PLAYBACK) + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_START); /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; -- 1.7.2.5

2 1

Re: [alsa-devel] [PATCH] ASoC: max98090: request IRQF_ONESHOT interrupt
by Mark Brown 20 May '13

20 May '13

On Sun, May 19, 2013 at 10:58:07PM -0700, Andrew Bresticker wrote: > request_threaded_irq() rejects calls which both do not specify a handler > (indicating that the primary IRQ handler should be used) and do not set > IRQF_ONESHOT because the combination is unsafe with level-triggered > interrupts. It is safe in this case, though, since max98090 IRQs are Applied, thanks. If request_threaded_irq() is rejecting these that's not ideal - some threaded interrupts can only be generated from within threaded context and therefore can't have primary handlers. Unless it only rejects in the case where the parent IRQ is a hard IRQ of course.

1 0

[alsa-devel] [PATCH 0/2 V3] Power-well API implementation for Haswell
by Wang Xingchao 20 May '13

20 May '13

Hi all, This is V3 and some fixes after Takashi's review. change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (2): drm/915: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA drivers/gpu/drm/i915/intel_pm.c | 76 +++++++++++++++++++++++++++++++++++---- include/drm/i915_powerwell.h | 36 +++++++++++++++++++ sound/pci/hda/Kconfig | 10 ++++++ sound/pci/hda/Makefile | 3 ++ sound/pci/hda/hda_i915.c | 75 ++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 ++++++++++++++++++ sound/pci/hda/hda_intel.c | 41 +++++++++++++++++++-- 7 files changed, 262 insertions(+), 10 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 6

Re: [alsa-devel] [PATCH v2] ASoC: dfbmcs320: make the driver common for other BT modules
by Mark Brown 18 May '13

18 May '13

On Fri, May 17, 2013 at 10:19:37AM +0800, Barry Song wrote: > DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, > we don't want everyone to have a seperate codec driver. anyway, the > feature of Bluetooth handsfree is same on all platforms, so this patch > makes the DFBM-CS320 driver become a common BT handsfree driver. Sorry, should have thought of this before but why are we calling this "handsfree"? It's just a generic bluetooth link as far as the host system is concerned, "bt-sco" might be more appropriate? Otherwise this looks OK for ASoC though I'd expect to see a change to registration of the OpenMoko BT device (perhaps I missed it though).

3 4

[alsa-devel] [patch] ASoC: wm8994: missing break in wm8994_get_fll_config()
by Dan Carpenter 18 May '13

18 May '13

Smatch complains that: sound/soc/codecs/wm8994.c:2087 wm8994_get_fll_config() warn: missing break? reassigning 'fll->k' Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9f32dd8..303d755 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2080,6 +2080,7 @@ static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll, fll->k = K / 10; pr_debug("N=%x K=%x\n", fll->n, fll->k); + break; default: gcd_fll = gcd(freq_out, freq_in);

2 1

[alsa-devel] [PATCH v3] ASoC: dfbmcs320: make the driver common for other BT modules
by Barry Song 18 May '13

18 May '13

DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, we don't want everyone to have a seperate codec driver. anyway, the feature of Bluetooth SCO is same on all platforms, so this patch makes the DFBM-CS320 driver become a common BT SCO link driver. Cc: Graeme Gregory <graeme(a)openmoko.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- -v3: rename to sco according to Mark's comment sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +++++++++++++++++++----------- sound/soc/samsung/neo1973_wm8753.c | 2 +- 4 files changed, 28 insertions(+), 19 deletions(-) rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d8c4f3d..386292d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DFBMCS320 + select SND_SOC_BT_SCO select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -263,7 +263,7 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DFBMCS320 +config SND_SOC_BT_SCO tristate config SND_SOC_DMIC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 49ff127..82aa34e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,7 +27,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o -snd-soc-dfbmcs320-objs := dfbmcs320.o +snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o @@ -154,7 +154,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o -obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o +obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/bt-sco.c similarity index 53% rename from sound/soc/codecs/dfbmcs320.c rename to sound/soc/codecs/bt-sco.c index 4f4f7f4..a081d9f 100644 --- a/sound/soc/codecs/dfbmcs320.c +++ b/sound/soc/codecs/bt-sco.c @@ -1,5 +1,5 @@ /* - * Driver for the DFBM-CS320 bluetooth module + * Driver for generic Bluetooth SCO link * Copyright 2011 Lars-Peter Clausen <lars(a)metafoo.de> * * This program is free software; you can redistribute it and/or modify it @@ -15,8 +15,8 @@ #include <sound/soc.h> -static struct snd_soc_dai_driver dfbmcs320_dai = { - .name = "dfbmcs320-pcm", +static struct snd_soc_dai_driver bt_sco_dai = { + .name = "bt-sco-pcm", .playback = { .channels_min = 1, .channels_max = 1, @@ -31,32 +31,41 @@ static struct snd_soc_dai_driver dfbmcs320_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320; +static struct snd_soc_codec_driver soc_codec_dev_bt_sco; -static int dfbmcs320_probe(struct platform_device *pdev) +static int bt_sco_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320, - &dfbmcs320_dai, 1); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco, + &bt_sco_dai, 1); } -static int dfbmcs320_remove(struct platform_device *pdev) +static int bt_sco_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver dfmcs320_driver = { +static struct platform_device_id bt_sco_driver_ids[] = { + { + .name = "dfbmcs320", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); + +static struct platform_driver bt_sco_driver = { .driver = { - .name = "dfbmcs320", + .name = "bt-sco", .owner = THIS_MODULE, }, - .probe = dfbmcs320_probe, - .remove = dfbmcs320_remove, + .probe = bt_sco_probe, + .remove = bt_sco_remove, + .id_table = bt_sco_driver_ids, }; -module_platform_driver(dfmcs320_driver); +module_platform_driver(bt_sco_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); +MODULE_DESCRIPTION("ASoC generic bluethooth sco link driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index e591c38..807db41 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -373,7 +373,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Voice via BT */ .name = "Bluetooth", .stream_name = "Voice", - .cpu_dai_name = "dfbmcs320-pcm", + .cpu_dai_name = "bt-sco-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, -- 1.8.2.3

1 0

[alsa-devel] [PATCH 0/6] Second set of fixes for ux500 ASoC drivers
by Fabio Baltieri 18 May '13

18 May '13

Hi, These are some others fixes for various small issues I found while testing the ux500 ASoC driver on -next. Patch 1 adds some missing declarations for AD controls that were causing some weird behaviour in alsamixer, as the default state was outside the declared range. Patch 2 fixes a kernel crash when opening and closing the audio device without sending any data. Patch 3 drops pinctrl code altogether from the driver. The actual implementation is buggy as the pins are only registered to the playback interfaces, which gives a bunch of warnings during kernel startup and also kills the capture interface by setting the shared pins to hi-z mode even if that's still active. As putting those pins in high-z is not really needed and was removed from the internal STE driver anyway, I'm just dropping that code form here as well. In parallel, I'm sending a pinctrl patch to declare those pin as a hog. Patches 4 to 6 fixes some weirdness with time slot usage. After this series the driver seems to work fine for both capture and playback interface (tested on a snowball v11). Thanks, Fabio Fabio Baltieri (6): ASoC: ab8500-codec: Add missing ad_to_slot definitions ASoC: ux500: Do not clear state if already idle ASoC: ux500: Drop pinctrl sleep support ASoC: ux500: Update tx tdm slots configuration ASoC: ux500: Swap even/odd AD slot definitions ASoC: ux500: Use the first two AD slots for capture sound/soc/codecs/ab8500-codec.c | 39 +++++++++++++++------------ sound/soc/codecs/ab8500-codec.h | 36 ++++++++++++------------- sound/soc/ux500/mop500_ab8500.c | 4 +-- sound/soc/ux500/ux500_msp_i2s.c | 58 +++-------------------------------------- sound/soc/ux500/ux500_msp_i2s.h | 6 ----- 5 files changed, 46 insertions(+), 97 deletions(-) -- 1.8.2

4 54

[alsa-devel] [PATCH] ASoC: soc-compress: Send correct stream event for capture start
by Charles Keepax 17 May '13

17 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Sorry little bug I missed due to some integration bugs in our setup here. Again this applies onto Takashi's sound for-next branch. Thanks, Charles sound/soc/soc-compress.c | 9 +++++++-- 1 files changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3853f7e..a5107df 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -220,8 +220,13 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, goto err; } - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_START); + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + } else { + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_START); + } /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Handle LRCLK inversion for WM8958 and WM1811A
by Mark Brown 17 May '13

17 May '13

On WM8958 and WM1811A separate control of the LRCLK inversion bit is available for the DAC and ADC LRCLKs which for compatibility reasons is done in a new register bit. Since writes to each scheme have no effect on parts using the other just always write to both for simplicity. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ab7ebd9..74238bd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2573,17 +2573,24 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct wm8994 *control = wm8994->wm8994; int ms_reg; int aif1_reg; + int dac_reg; + int adc_reg; int ms = 0; int aif1 = 0; + int lrclk = 0; switch (dai->id) { case 1: ms_reg = WM8994_AIF1_MASTER_SLAVE; aif1_reg = WM8994_AIF1_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; case 2: ms_reg = WM8994_AIF2_MASTER_SLAVE; aif1_reg = WM8994_AIF2_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; default: return -EINVAL; @@ -2602,6 +2609,7 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; @@ -2640,12 +2648,14 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_IB_IF: aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; case SND_SOC_DAIFMT_IB_NF: aif1 |= WM8994_AIF1_BCLK_INV; break; case SND_SOC_DAIFMT_NB_IF: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; default: return -EINVAL; @@ -2676,6 +2686,10 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) aif1); snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR, ms); + snd_soc_update_bits(codec, dac_reg, + WM8958_AIF1_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, adc_reg, + WM8958_AIF1_LRCLK_INV, lrclk); return 0; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm5110: Add missing speaker initialisation
by Mark Brown 17 May '13

17 May '13

Add callback to initialise the speaker in the core following the recent changes to handling of integration with the thermal interrupts. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index e77410e..91a7015 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1252,6 +1252,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; + arizona_init_spk(codec); + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); priv->core.arizona->dapm = &codec->dapm; -- 1.7.10.4

1 0

[alsa-devel] module dependency between i915 and snd-hda-intel
by Wang, Xingchao 17 May '13

17 May '13

Hi Takashi, Seems kernel building will determine the module loading sequence, this would fix the dependency for i915 and snd-hda-intel. If we call the i915 module API directly in snd-hda-intel side, kernel building will change the module.dep accordingly like: " kernel/sound/pci/hda/snd-hda-intel.ko: kernel/drivers/gpu/drm/i915/i915.ko kernel/drivers/gpu/dr m/drm_kms_helper.ko kernel/drivers/gpu/drm/drm.ko kernel/drivers/i2c/algos/i2c-algo-bit.ko kerne l/drivers/acpi/video.ko kernel/sound/pci/hda/snd-hda-codec.ko kernel/sound/core/snd-hwdep.ko ker nel/sound/core/snd-pcm.ko kernel/sound/core/snd-timer.ko kernel/sound/core/snd.ko kernel/sound/s oundcore.ko kernel/sound/core/snd-page-alloc.ko " So snd-hda-intel will wait for i915 loading. Even I added i915 into blacklist.conf, it will force load i915.ko. So I think the only bad case is no i915 module built-in, that would cause haswell hda initialize fail, and the patch would output error message. thanks --xingchao

2 6

[alsa-devel] [PATCH 1/3] ASoC: arizona: Provide simple DAI ops for autoconfiguring interfaces
by Mark Brown 17 May '13

17 May '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 7 +++++++ sound/soc/codecs/arizona.h | 1 + 2 files changed, 8 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 90c3db1..4a42041 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1198,6 +1198,13 @@ const struct snd_soc_dai_ops arizona_dai_ops = { }; EXPORT_SYMBOL_GPL(arizona_dai_ops); +const struct snd_soc_dai_ops arizona_simple_dai_ops = { + .startup = arizona_startup, + .hw_params = arizona_hw_params_rate, + .set_sysclk = arizona_dai_set_sysclk, +}; +EXPORT_SYMBOL_GPL(arizona_simple_dai_ops); + int arizona_init_dai(struct arizona_priv *priv, int id) { struct arizona_dai_priv *dai_priv = &priv->dai[id]; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index af39f10..a1a5dad 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -213,6 +213,7 @@ extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); extern const struct snd_soc_dai_ops arizona_dai_ops; +extern const struct snd_soc_dai_ops arizona_simple_dai_ops; #define ARIZONA_FLL_NAME_LEN 20 -- 1.7.10.4

1 2

[alsa-devel] [PATCH] ASoC: simplify registration of snd-soc-dummy device
by Uwe Kleine-König 17 May '13

17 May '13

Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/soc-utils.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 4b3be6c..29b211e 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -159,15 +159,10 @@ int __init snd_soc_util_init(void) { int ret; - soc_dummy_dev = platform_device_alloc("snd-soc-dummy", -1); - if (!soc_dummy_dev) - return -ENOMEM; - - ret = platform_device_add(soc_dummy_dev); - if (ret != 0) { - platform_device_put(soc_dummy_dev); - return ret; - } + soc_dummy_dev = + platform_device_register_simple("snd-soc-dummy", -1, NULL, 0); + if (IS_ERR(soc_dummy_dev)) + return PTR_ERR(soc_dummy_dev); ret = platform_driver_register(&soc_dummy_driver); if (ret != 0) -- 1.8.2.1

2 1

Re: [alsa-devel] [PATCH v2] ASoC: max98090: add digital mic mux to record path
by Mark Brown 17 May '13

17 May '13

On Thu, May 16, 2013 at 12:03:54PM -0700, Andrew Bresticker wrote: > The max98090 driver currently treats the digital mic enable as a supply > on the record path, causing the digital mic enable to always be turned on > when attempting to record. This is incorrect, however, since the digital Applied, thanks.

1 0

[alsa-devel] [PATCH v2] ALSA: hda - add PCI IDs for Intel BayTrail
by Chew Chiau Ee 17 May '13

17 May '13

From: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> Add HD Audio Device PCI ID for the Intel BayTrail platform. Signed-off-by: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> Signed-off-by: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bcd40ee..fa902a4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3847,6 +3847,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Oaktrail */ { PCI_DEVICE(0x8086, 0x080a), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + /* BayTrail */ + { PCI_DEVICE(0x8086, 0x0f04), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | -- 1.7.4.4

2 1

Re: [alsa-devel] [PATCH] ALSA: usb-audio: proc: use found syncmaxsize to determine feedback format
by Takashi Iwai 17 May '13

17 May '13

At Thu, 16 May 2013 20:26:17 +0200, Torstein Hegge wrote: > > freqshift is only set for the data endpoint and syncmaxsize is only set > for the sync endpoint. This results in a syncmaxsize of zero used in the > proc output feedback format calculation, which gives a feedback format > incorrectly shown as 8.16 for UAC2 devices. > > As neither the data nor the sync endpoint gives all the relevant > content, output the two combined. > > Also remove the sync_endpoint "packet size" which is always zero > and the sync_endpoint "momentary freq" which is constant. > > Tested with UAC2 async and UAC1 adaptive, not tested with UAC1 async. > > Reported-by: B. Zhang <bb.zhang(a)free.fr> > Signed-off-by: Torstein Hegge <hegge(a)resisty.net> Thanks, applied. Takashi > --- > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: Daniel Mack <zonque(a)gmail.com> > Cc: Clemens Ladisch <clemens(a)ladisch.de> > Cc: Davor Herga <davor.herga(a)guest.arnes.si> > > sound/usb/proc.c | 22 +++++++++++----------- > 1 file changed, 11 insertions(+), 11 deletions(-) > > diff --git a/sound/usb/proc.c b/sound/usb/proc.c > index d218f76..462a5ed 100644 > --- a/sound/usb/proc.c > +++ b/sound/usb/proc.c > @@ -117,21 +117,22 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s > } > > static void proc_dump_ep_status(struct snd_usb_substream *subs, > - struct snd_usb_endpoint *ep, > + struct snd_usb_endpoint *data_ep, > + struct snd_usb_endpoint *sync_ep, > struct snd_info_buffer *buffer) > { > - if (!ep) > + if (!data_ep) > return; > - snd_iprintf(buffer, " Packet Size = %d\n", ep->curpacksize); > + snd_iprintf(buffer, " Packet Size = %d\n", data_ep->curpacksize); > snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n", > subs->speed == USB_SPEED_FULL > - ? get_full_speed_hz(ep->freqm) > - : get_high_speed_hz(ep->freqm), > - ep->freqm >> 16, ep->freqm & 0xffff); > - if (ep->freqshift != INT_MIN) { > - int res = 16 - ep->freqshift; > + ? get_full_speed_hz(data_ep->freqm) > + : get_high_speed_hz(data_ep->freqm), > + data_ep->freqm >> 16, data_ep->freqm & 0xffff); > + if (sync_ep && data_ep->freqshift != INT_MIN) { > + int res = 16 - data_ep->freqshift; > snd_iprintf(buffer, " Feedback Format = %d.%d\n", > - (ep->syncmaxsize > 3 ? 32 : 24) - res, res); > + (sync_ep->syncmaxsize > 3 ? 32 : 24) - res, res); > } > } > > @@ -141,8 +142,7 @@ static void proc_dump_substream_status(struct snd_usb_substream *subs, struct sn > snd_iprintf(buffer, " Status: Running\n"); > snd_iprintf(buffer, " Interface = %d\n", subs->interface); > snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx); > - proc_dump_ep_status(subs, subs->data_endpoint, buffer); > - proc_dump_ep_status(subs, subs->sync_endpoint, buffer); > + proc_dump_ep_status(subs, subs->data_endpoint, subs->sync_endpoint, buffer); > } else { > snd_iprintf(buffer, " Status: Stop\n"); > } > -- > 1.8.3.rc1.377.g7010c6b >

1 0

Re: [alsa-devel] [PATCH] ASoC: dfbmcs320: make the driver common for other BT modules
by Lars-Peter Clausen 16 May '13

16 May '13

On 05/16/2013 08:53 AM, Barry Song wrote: > DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, > we don't want everyone to have a seperate codec driver. anyway, the > feature of Bluetooth handsfree is same on all platforms, so this patch > makes the DFBM-CS320 driver become a common BT handsfree driver. [...] > -static struct platform_driver dfmcs320_driver = { > +static struct platform_driver bt_handsfree_driver = { > .driver = { > - .name = "dfbmcs320", > + .name = "bt-handsfree", > .owner = THIS_MODULE, In my opinion it's a good idea to keep a platform_id list of all the supported devices. > }, > - .probe = dfbmcs320_probe, > - .remove = dfbmcs320_remove, > + .probe = bt_handsfree_probe, > + .remove = bt_handsfree_remove, > }; > > -module_platform_driver(dfmcs320_driver); > +module_platform_driver(bt_handsfree_driver);

3 4

[alsa-devel] [PATCH] ASoC: wm5102: Correct OSR control name for EPOUT
by Charles Keepax 16 May '13

16 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index cb03cc4..803d1b8 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -814,7 +814,7 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, SOC_VALUE_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]), SOC_VALUE_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]), -SOC_VALUE_ENUM("HPOUT3 OSR", wm5102_hpout_osr[2]), +SOC_VALUE_ENUM("EPOUT OSR", wm5102_hpout_osr[2]), SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), -- 1.7.2.5

2 1

[alsa-devel] Debian alsa-plugins patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-plugins that were pending upstreaming Thanks, Jordi

3 8

[alsa-devel] Only stereo output possible with ATI Tahiti XT HDMI Audio [Radeon HD 7970 Series] [1002:aaa0]
by Olivier Langlois 16 May '13

16 May '13

I have an HDMI chain with a Radeon card on 1 side and a 1080P monitor at the other end with a Pioner SC-09TX A/V Receiver in the middle. Video is transmitted fine. 2 channels pcm plays fine on the receiver. I am totally unable to output sound with 8 channels. lano1106@whippet2 /proc/asound/card0 $ speaker-test -c8 speaker-test 1.0.27 Playback device is default Stream parameters are 48000Hz, S16_LE, 8 channels Using 16 octaves of pink noise Channels count (8) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument My neophyte instinct tells me that the problem is interoperability issue between the Radeon Audio controller and the SC-09TX receiver. That would be great to be able to tweak the hda driver to workaround the problem but if I was able to manually program the hda codec module to accept 8 channel LPCM input, that would be good enough for me. My understanding is that in the advent of HDMI handshake failure, the sound source fallbacks to the bare minimum requirement that the HDMI standard requires. I have started to read Intel HDA manual. What I would like to know is: Is it known if my particular HDA device is able to output 48KHz 8 channels LPCM? would it configurable manually? Which HDA verbs should get familiar with if I want to tweak the code to configure the controller manually? Thank you, Olivier lano1106@whippet2 ~ $ lspci -nn | grep Audio 01:00.1 Audio device [0403]: Advanced Micro Devices [AMD] nee ATI Tahiti XT HDMI Audio [Radeon HD 7970 Series] [1002:aaa0] Linux: 3.8.12 alsalib:1.0.27 lano1106@whippet2 /proc/asound/card0 $ dmesg | grep ALSA [ 6.351490] ALSA sound/pci/hda/hda_intel.c:2848 0000:01:00.1: Handle VGA-switcheroo audio client [ 6.351491] ALSA sound/pci/hda/hda_intel.c:3040 0000:01:00.1: Using LPIB position fix [ 6.351493] ALSA sound/pci/hda/hda_intel.c:3157 0000:01:00.1: Force to non-snoop mode [ 6.351619] ALSA sound/pci/hda/hda_intel.c:3282 0000:01:00.1: chipset global capabilities = 0x0 [ 6.354036] ALSA sound/pci/hda/hda_intel.c:1131 0000:01:00.1: codec_mask = 0x1 [ 6.354061] ALSA sound/pci/hda/hda_intel.c:1600 0000:01:00.1: codec #0 probed OK [ 6.354062] ALSA sound/pci/hda/hda_intel.c:1716 0000:01:00.1: Enable sync_write for stable communication [ 6.411369] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=3 Presence_Detect=0 ELD_Valid=0 [ 6.411397] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=5 Presence_Detect=0 ELD_Valid=0 [ 6.411424] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=7 Presence_Detect=0 ELD_Valid=0 [ 6.411453] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=9 Presence_Detect=0 ELD_Valid=0 [ 6.411482] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=11 Presence_Detect=0 ELD_Valid=0 [ 6.411512] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=13 Presence_Detect=0 ELD_Valid=0 [ 29.487577] ALSA sound/pci/hda/patch_hdmi.c:951 HDMI hot plug event: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=0 [ 29.487593] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=1 [ 29.487605] ALSA sound/pci/hda/hda_eld.c:337 HDMI: ELD buf size is 0, force 128 [ 29.487617] ALSA sound/pci/hda/hda_eld.c:356 HDMI: invalid ELD data byte 0 lano1106@whippet2 /proc/asound/card0 $ cat eld#0.0 monitor_present 1 eld_valid 0 lano1106@whippet2 /proc/asound/card0 :( $ cat codec#0 Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100300 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=1, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Connection: 1 0x02 Node 0x04 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=7 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Control: name="ELD", index=0, device=7 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=02, enabled=1 Connection: 1 0x04 Node 0x06 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=8 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Control: name="ELD", index=0, device=8 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=03, enabled=1 Connection: 1 0x06 Node 0x08 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=9 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=3, device=0 Control: name="IEC958 Playback Pro Mask", index=3, device=0 Control: name="IEC958 Playback Default", index=3, device=0 Control: name="IEC958 Playback Switch", index=3, device=0 Control: name="ELD", index=0, device=9 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=1 Connection: 1 0x08 Node 0x0a [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=10 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=4, device=0 Control: name="IEC958 Playback Pro Mask", index=4, device=0 Control: name="IEC958 Playback Default", index=4, device=0 Control: name="IEC958 Playback Switch", index=4, device=0 Control: name="ELD", index=0, device=10 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=05, enabled=1 Connection: 1 0x0a Node 0x0c [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=11 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=5, device=0 Control: name="IEC958 Playback Pro Mask", index=5, device=0 Control: name="IEC958 Playback Default", index=5, device=0 Control: name="IEC958 Playback Switch", index=5, device=0 Control: name="ELD", index=0, device=11 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=06, enabled=1 Connection: 1 0x0c whippet2 /sys/kernel/debug/tracing # cat trace_pipe kworker/u:5-130 [002] ...1 79390.094652: hda_send_cmd: [0:0] val=3f0900 kworker/u:5-130 [002] ...1 79390.094665: hda_get_response: [0:0] val=ffffffff kworker/u:5-130 [002] ...1 79390.094667: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [002] ...1 79390.094677: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.094679: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [002] ...1 79390.094689: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.394001: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [002] ...1 79390.394045: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.394050: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [002] ...1 79390.394071: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.694227: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [002] ...1 79390.694281: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.694286: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [002] ...1 79390.694307: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.617406: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28163.617420: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.617425: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28163.617436: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.917628: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28163.917642: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.917647: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28163.917657: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.217846: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28164.217860: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.217865: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28164.217875: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.518101: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28164.518115: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.518119: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28164.518130: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.818310: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28164.818323: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.818328: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28164.818339: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28165.118549: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28165.118563: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28165.118568: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28165.118579: hda_get_response: [0:0] val=0

1 2

[alsa-devel] 7.1 surround channels missing on CS46xx
by Ondrej Zary 16 May '13

16 May '13

Hello, I got a Hercules Fortissimo 7.1 sound card (CS4624 + two CS4294) but snd-cs46xx driver supports only 6 channels (front, rear, center/lfe). The surround output (7&8) is not supported - as seen in dsp_spos_scb_lib.c: case DSP_PCM_S71_CHANNEL: /* TODO */ snd_BUG(); break; Tried to add a code for this, mostly copying the center/lfe code but I miss some data: - which slots (ACOSV_SLVx) should be used (maybe 10 and 12?) - what are the channel_disp and fifo_addr values for cs46xx_dsp_create_codec_out_scb() function - what values should S71_MIXER_SCB_ADDR and S71_CODEC_SCB_ADDR be (maybe 0x1b0 and 0x1c0?) -- Ondrej Zary

1 0

[alsa-devel] Debian alsa-tools patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-tools that were pending upstreaming. Please review patch 8, as this one is new and the general state of configure scripts in alsa-tools is less than optimal. I've left out the patch to correct envy24control's configuration directory, as the additional check for the old dir is not implemented. If anyone wants to pick that up, please contact me or fetch the (trivial) patch from Debian sources. Thanks, Jordi

3 15

[alsa-devel] Debian alsa-utils patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-utils that were pending upstreaming. Thanks, Jordi

2 5

[alsa-devel] Debian alsa-lib patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-lib that were pending upstreaming. Thanks, Jordi

2 4

[alsa-devel] Debian alsa-lib patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-oss that were pending upstreaming. Thanks, Jordi [PATCH 1/3] Fix path to libaoss.so. [PATCH 2/3] Show a warning if the alsa wrapper failed. [PATCH 3/3] Add AM_MAINTAINER_MODE([enable]) to configure.in.

2 6

[alsa-devel] [PATCH - pcm_plug] Add details about what pcm_plug does in doc
by olivier＠trillion01.com 16 May '13

16 May '13

From: Olivier Langlois <olivier(a)trillion01.com> Current doc only mention conversion to channels, rate and format. It alsa takes care of access mode mismatch. This is what I was looking for and I had to look the source code to figure out. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index fa84eaa..8a639af 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -1155,7 +1155,8 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, \section pcm_plugins_plug Automatic conversion plugin -This plugin converts channels, rate and format on request. +This plugin converts channels, rate and format on request. It will also +implicitly adapt to the slave plugin access mode if different than our own. \code pcm.name { -- 1.8.2.3

1 0

Re: [alsa-devel] [PATCH] ALSA: hda - add PCI IDs for Intel BayTrail
by Takashi Iwai 16 May '13

16 May '13

At Thu, 16 May 2013 13:27:18 +0800, Chew Chiau Ee wrote: > > From: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> > > Add HD Audio Device PCI ID for the Intel BayTrail platform. It's rather the successor of Oaktrail & co, right? Then put after Oaktrail entry, please. thanks, Takashi > Signed-off-by: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> > Signed-off-by: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> > --- > sound/pci/hda/hda_intel.c | 3 +++ > 1 files changed, 3 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index bcd40ee..18640d9 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -3811,6 +3811,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* CPT */ > { PCI_DEVICE(0x8086, 0x1c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > + /* BayTrail */ > + { PCI_DEVICE(0x8086, 0x0f04), > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > -- > 1.7.4.4 >

1 0

[alsa-devel] [PATCH 0/2 V2] Power-well API implementation for Haswell
by Wang Xingchao 16 May '13

16 May '13

Hi all, This is V2 and some fixes after first round review: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (2): drm/915: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA drivers/gpu/drm/i915/intel_pm.c | 76 +++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 +++++++++++++++ sound/pci/hda/Kconfig | 8 ++++ sound/pci/hda/Makefile | 3 ++ sound/pci/hda/hda_i915.c | 92 +++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 +++++++++++++++ sound/pci/hda/hda_intel.c | 33 ++++++++++++-- 7 files changed, 273 insertions(+), 10 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 7

[alsa-devel] [PATCH] ALSA: hda - Add headset mic support for another Dell machine
by David Henningsson 15 May '13

15 May '13

BugLink: https://bugs.launchpad.net/bugs/1180351 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6bf47f7..59d2e91 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3482,6 +3482,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.7.9.5

2 1

[alsa-devel] Error after wakeup from suspend?! (HDA Intel PCH)
by Tom 15 May '13

15 May '13

Hi, I woke up my ArchLinux system and pulseaudio is not working. Also I got this message from the commandline: https://gist.github.com/anonymous/5576712 Pulseaudio is running like it would work fine. I can see the volume bars shaking in Pulse Audio Volume Manager. Except, I can't hear any sound. It's like the audio plug is not connected. Hopefully this helps you a bit.

2 1

[alsa-devel] [Fwd: [PATCH] ALSA: snd-aoa: Add a layout entry for PowerBook6, 5]
by Benjamin Herrenschmidt 15 May '13

15 May '13

Acked-by: Benjamin Herrenschmidt <benh(a)kernel.crashing.org>

2 1

[alsa-devel] [PATCH 0/5] ASoC: fsl: imx-pcm driver cleanup
by Shawn Guo 15 May '13

15 May '13

The series removes the use of imx-pcm-audio/imx-fiq-pcm-audio platform device, and deletes imx-pcm driver by moving the functions needed by imx-pcm-fiq driver into imx-pcm-fiq.c. This can be done because imx-pcm-dma has moved to generic dmaengine pcm driver and does not use any helper from imx-pcm driver now. Shawn Guo (5): ASoC: fsl: remove use of imx-pcm-audio from fsl_ssi ASoC: fsl: remove use of imx-pcm-audio from imx-ssi ASoC: fsl: create function imx_pcm_fiq_exit() ASoC: fsl: remove use of imx-fiq-pcm-audio from imx-ssi ASoC: fsl: remove imx-pcm driver sound/soc/fsl/Kconfig | 5 -- sound/soc/fsl/Makefile | 11 +-- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 13 ++-- sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-pcm-dma.c | 2 + sound/soc/fsl/imx-pcm-fiq.c | 92 +++++++++++++++++++++++++ sound/soc/fsl/imx-pcm.c | 145 --------------------------------------- sound/soc/fsl/imx-pcm.h | 10 +-- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/imx-ssi.c | 44 +++--------- sound/soc/fsl/imx-ssi.h | 3 - sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/phycore-ac97.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- 15 files changed, 122 insertions(+), 215 deletions(-) delete mode 100644 sound/soc/fsl/imx-pcm.c -- 1.7.9.5

2 13

[alsa-devel] ASoC updates for v3.10
by Mark Brown 15 May '13

15 May '13

The following changes since commit f722406faae2d073cc1d01063d1123c35425939e: Linux 3.10-rc1 (2013-05-11 17:14:08 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-rc1 for you to fetch changes up to 407a0e18ecfbd7f58f49c307a657b63e6f32544d: Merge remote-tracking branch 'asoc/fix/wm0010' into asoc-linus (2013-05-14 16:53:21 +0400) ---------------------------------------------------------------- ASoC: Fixes for v3.10 A few small driver-specific fixes, none of them especially crippling. ---------------------------------------------------------------- Axel Lin (1): ASoC: da7213: Fix setting dmic_samplephase and dmic_clk_rate Fabio Baltieri (1): ASoC: ux500: Swap even/odd AD slot definitions Mark Brown (3): Merge remote-tracking branch 'asoc/fix/da7213' into asoc-linus Merge remote-tracking branch 'asoc/fix/ux500' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm0010' into asoc-linus Wei Yongjun (1): ASoC: wm0010: fix error return code in wm0010_boot() sound/soc/codecs/ab8500-codec.h | 36 ++++++++++++++++++------------------ sound/soc/codecs/da7213.c | 8 ++++---- sound/soc/codecs/wm0010.c | 1 + 3 files changed, 23 insertions(+), 22 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: hda - Check the activity of the NID to be powered down
by Takashi Iwai 15 May '13

15 May '13

When an inactive path is powered down with spec->power_down_unused flag, we should check the activity of each widget in the path whether it's still referred from any active path. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac079f9..ae85bbd2 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -606,6 +606,10 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, return false; } +/* check whether the NID is referred by any active paths */ +#define is_active_nid_for_any(codec, nid) \ + is_active_nid(codec, nid, HDA_OUTPUT, 0) + /* get the default amp value for the target state */ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps, bool enable) @@ -759,7 +763,8 @@ static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) for (i = 0; i < path->depth; i++) { hda_nid_t nid = path->path[i]; - if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3)) { + if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3) && + !is_active_nid_for_any(codec, nid)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); @@ -4157,7 +4162,7 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, return power_state; if (get_wcaps_type(get_wcaps(codec, nid)) >= AC_WID_POWER) return power_state; - if (is_active_nid(codec, nid, HDA_OUTPUT, 0)) + if (is_active_nid_for_any(codec, nid)) return power_state; return AC_PWRST_D3; } -- 1.8.2.1

1 0

[alsa-devel] Segmentation fault during snd_pcm_prepare()
by Shane Loretz 15 May '13

15 May '13

I'm interested in writing an application using alsa. I've chosen to explore the library by writing a program to echo sound data from a microphone back to my speakers. I'm only interested in mono sound. When running this program, the first call to snd_pcm_readi() fails, and a segmentation fault occurs inside of the snd_pcm_prepare() function according to a backtrace from gdb. I suspect I'm not allocating enough space for the pointer capdata, but I don't know how to determine how much space I need. Here is the code // Test out the alsa library // #include <alsa/asoundlib.h> #include <cstdio> #include <unistd.h> bool getMonoDevice(snd_pcm_t *pcm_handle, snd_pcm_stream_t stream, unsigned int rate, int periods, snd_pcm_uframes_t periodsize); int main() { snd_pcm_t *pcm_handle_read; snd_pcm_t *pcm_handle_write; snd_pcm_stream_t stream = SND_PCM_STREAM_CAPTURE; unsigned int rate = 44100; int periods = 2; snd_pcm_uframes_t periodsize = 8192; if (!getMonoDevice(pcm_handle_write, SND_PCM_STREAM_PLAYBACK,rate,periods,periodsize)) { fprintf(stderr, "Error couldn't open speakers\n"); return -1; } if (!getMonoDevice(pcm_handle_read, SND_PCM_STREAM_CAPTURE,rate,periods,periodsize)) { fprintf(stderr, "Error couldn't open microphone\n"); return -1; } //read frames from device //and echo them out unsigned char *capdata; int pcmreturn; int num_frames; int error; capdata = (unsigned char *)malloc(periodsize); num_frames = periodsize * 2; while (true) { while ((pcmreturn = snd_pcm_readi(pcm_handle_read, capdata, num_frames)) < 0) { usleep(10); error = snd_pcm_prepare(pcm_handle_read); if (error < 0) { fprintf(stderr, "Unrecoverable mic error %s\n",snd_strerror(error)); return error; } } while ((pcmreturn = snd_pcm_writei(pcm_handle_write, capdata, num_frames)) < 0) { usleep(10); error = snd_pcm_prepare(pcm_handle_write); if (error < 0) { fprintf(stderr, "Unrecoverable speaker error %s\n",snd_strerror(error)); return error; } } } //stop pcm device and drop pending frames snd_pcm_drop(pcm_handle_read); snd_pcm_drain(pcm_handle_read); snd_pcm_drop(pcm_handle_write); snd_pcm_drain(pcm_handle_write); } bool getMonoDevice(snd_pcm_t *pcm_handle, snd_pcm_stream_t stream, unsigned int rate, int periods, snd_pcm_uframes_t periodsize) { snd_pcm_hw_params_t *hwparams; char *pcm_name; pcm_name = strdup("plughw:0,0"); snd_pcm_hw_params_alloca(&hwparams); if (snd_pcm_open(&pcm_handle, pcm_name, stream, 0) < 0) { fprintf(stderr, "Error opening PCM device %s\n", pcm_name); return false; } if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0 ) { fprintf(stderr, "Can not configure this PCM device.\n"); return false; } unsigned int exact_rate; int dir; if (snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) { fprintf(stderr, "Error setting access.\n"); return false; } if (snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE) < 0) { fprintf(stderr, "Error setting format.\n"); return false; } exact_rate = rate; if (snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0) < 0) { fprintf(stderr, "Error setting rate.\n"); return false; } if (rate != exact_rate) { fprintf(stderr, "The rate %d is not supported by your hardware.\n Using %d Hz instead.\n", rate, exact_rate); } //Configure mono sound if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 1) < 0) { fprintf(stderr, "Error setting channels.\n"); return false; } if (snd_pcm_hw_params_set_periods(pcm_handle, hwparams, periods, 0) < 0) { fprintf(stderr, "Error setting periods.\n"); return false; } if (snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (periodsize * periods) >> 2) < 0 ) { fprintf(stderr, "Error setting buffersize.\n"); return false; } if (snd_pcm_hw_params(pcm_handle, hwparams) < 0) { fprintf(stderr, "Error setting HW params.\n"); return false; } //device is finally configured return true; }

2 1

[alsa-devel] [PATCH] ASoC: remove saarb and tavorevb3 machine drivers
by Paul Bolle 15 May '13

15 May '13

Support for PXA95x was removed in v3.8. This means that the Kconfig symbols MACH_SAARB and MACH_TAVOREVB3 are no longer available. This leaves the SoC Audio support for Marvell Saarb and Marvell Tavor EVB3 unbuildable. Remove these drivers too. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- 0) Basically untested. 1) To be specific: it was commit 49ea7fc094 ("ARM: pxa: remove pxa95x support") that removed MACH_SAARB and MACH_TAVOREVB3. sound/soc/pxa/Kconfig | 20 ----- sound/soc/pxa/Makefile | 4 - sound/soc/pxa/saarb.c | 190 ---------------------------------------------- sound/soc/pxa/tavorevb3.c | 189 --------------------------------------------- 4 files changed, 403 deletions(-) delete mode 100644 sound/soc/pxa/saarb.c delete mode 100644 sound/soc/pxa/tavorevb3.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 4d2e46f..b358094 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -130,26 +130,6 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. -config SND_SOC_SAARB - tristate "SoC Audio support for Marvell Saarb" - depends on SND_PXA2XX_SOC && MACH_SAARB - select MFD_88PM860X - select SND_PXA_SOC_SSP - select SND_SOC_88PM860X - help - Say Y if you want to add support for SoC audio on the - Marvell Saarb reference platform. - -config SND_SOC_TAVOREVB3 - tristate "SoC Audio support for Marvell Tavor EVB3" - depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 - select MFD_88PM860X - select SND_PXA_SOC_SSP - select SND_SOC_88PM860X - help - Say Y if you want to add support for SoC audio on the - Marvell Saarb reference platform. - config SND_PXA910_SOC tristate "SoC Audio for Marvell PXA910 chip" depends on ARCH_MMP && SND diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index d8a265d..2cff67b 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,8 +23,6 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o -snd-soc-saarb-objs := saarb.o -snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-hx4700-objs := hx4700.o snd-soc-magician-objs := magician.o @@ -48,8 +46,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o -obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o -obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c deleted file mode 100644 index c34146b..0000000 --- a/sound/soc/pxa/saarb.c +++ /dev/null @@ -1,190 +0,0 @@ -/* - * saarb.c -- SoC audio for saarb - * - * Copyright (C) 2010 Marvell International Ltd. - * Haojian Zhuang <haojian.zhuang(a)marvell.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <linux/clk.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> - -#include "../codecs/88pm860x-codec.h" -#include "pxa-ssp.h" - -static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd); - -static struct platform_device *saarb_snd_device; - -static struct snd_soc_jack hs_jack, mic_jack; - -static struct snd_soc_jack_pin hs_jack_pins[] = { - { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, -}; - -static struct snd_soc_jack_pin mic_jack_pins[] = { - { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, -}; - -/* saarb machine dapm widgets */ -static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Stereophone", NULL), - SND_SOC_DAPM_LINE("Lineout Out 1", NULL), - SND_SOC_DAPM_LINE("Lineout Out 2", NULL), - SND_SOC_DAPM_SPK("Ext Speaker", NULL), - SND_SOC_DAPM_MIC("Ext Mic 1", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Ext Mic 3", NULL), -}; - -/* saarb machine audio map */ -static const struct snd_soc_dapm_route saarb_audio_map[] = { - {"Headset Stereophone", NULL, "HS1"}, - {"Headset Stereophone", NULL, "HS2"}, - - {"Ext Speaker", NULL, "LSP"}, - {"Ext Speaker", NULL, "LSN"}, - - {"Lineout Out 1", NULL, "LINEOUT1"}, - {"Lineout Out 2", NULL, "LINEOUT2"}, - - {"MIC1P", NULL, "Mic1 Bias"}, - {"MIC1N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Ext Mic 1"}, - - {"MIC2P", NULL, "Mic1 Bias"}, - {"MIC2N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Headset Mic 2"}, - - {"MIC3P", NULL, "Mic3 Bias"}, - {"MIC3N", NULL, "Mic3 Bias"}, - {"Mic3 Bias", NULL, "Ext Mic 3"}, -}; - -static int saarb_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int width = snd_pcm_format_physical_width(params_format(params)); - int ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, - PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); - - return ret; -} - -static struct snd_soc_ops saarb_i2s_ops = { - .hw_params = saarb_i2s_hw_params, -}; - -static struct snd_soc_dai_link saarb_dai[] = { - { - .name = "88PM860x I2S", - .stream_name = "I2S Audio", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "88pm860x-i2s", - .platform_name = "pxa-pcm-audio", - .codec_name = "88pm860x-codec", - .init = saarb_pm860x_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &saarb_i2s_ops, - }, -}; - -static struct snd_soc_card snd_soc_card_saarb = { - .name = "Saarb", - .owner = THIS_MODULE, - .dai_link = saarb_dai, - .num_links = ARRAY_SIZE(saarb_dai), - - .dapm_widgets = saarb_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets), - .dapm_routes = saarb_audio_map, - .num_dapm_routes = ARRAY_SIZE(saarb_audio_map), -}; - -static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); - snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - - /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); - - /* headphone, microphone detection & headset short detection */ - pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, - SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); - pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); - return 0; -} - -static int __init saarb_init(void) -{ - int ret; - - if (!machine_is_saarb()) - return -ENODEV; - saarb_snd_device = platform_device_alloc("soc-audio", -1); - if (!saarb_snd_device) - return -ENOMEM; - - platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb); - - ret = platform_device_add(saarb_snd_device); - if (ret) - platform_device_put(saarb_snd_device); - - return ret; -} - -static void __exit saarb_exit(void) -{ - platform_device_unregister(saarb_snd_device); -} - -module_init(saarb_init); -module_exit(saarb_exit); - -MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); -MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c deleted file mode 100644 index 8b5ab8f..0000000 --- a/sound/soc/pxa/tavorevb3.c +++ /dev/null @@ -1,189 +0,0 @@ -/* - * tavorevb3.c -- SoC audio for Tavor EVB3 - * - * Copyright (C) 2010 Marvell International Ltd. - * Haojian Zhuang <haojian.zhuang(a)marvell.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <linux/clk.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> - -#include "../codecs/88pm860x-codec.h" -#include "pxa-ssp.h" - -static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd); - -static struct platform_device *evb3_snd_device; - -static struct snd_soc_jack hs_jack, mic_jack; - -static struct snd_soc_jack_pin hs_jack_pins[] = { - { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, -}; - -static struct snd_soc_jack_pin mic_jack_pins[] = { - { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, -}; - -/* tavorevb3 machine dapm widgets */ -static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_LINE("Lineout Out 1", NULL), - SND_SOC_DAPM_LINE("Lineout Out 2", NULL), - SND_SOC_DAPM_SPK("Ext Speaker", NULL), - SND_SOC_DAPM_MIC("Ext Mic 1", NULL), - SND_SOC_DAPM_MIC("Headset Mic 2", NULL), - SND_SOC_DAPM_MIC("Ext Mic 3", NULL), -}; - -/* tavorevb3 machine audio map */ -static const struct snd_soc_dapm_route evb3_audio_map[] = { - {"Headset Stereophone", NULL, "HS1"}, - {"Headset Stereophone", NULL, "HS2"}, - - {"Ext Speaker", NULL, "LSP"}, - {"Ext Speaker", NULL, "LSN"}, - - {"Lineout Out 1", NULL, "LINEOUT1"}, - {"Lineout Out 2", NULL, "LINEOUT2"}, - - {"MIC1P", NULL, "Mic1 Bias"}, - {"MIC1N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Ext Mic 1"}, - - {"MIC2P", NULL, "Mic1 Bias"}, - {"MIC2N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Headset Mic 2"}, - - {"MIC3P", NULL, "Mic3 Bias"}, - {"MIC3N", NULL, "Mic3 Bias"}, - {"Mic3 Bias", NULL, "Ext Mic 3"}, -}; - -static int evb3_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int width = snd_pcm_format_physical_width(params_format(params)); - int ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, - PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); - return ret; -} - -static struct snd_soc_ops evb3_i2s_ops = { - .hw_params = evb3_i2s_hw_params, -}; - -static struct snd_soc_dai_link evb3_dai[] = { - { - .name = "88PM860x I2S", - .stream_name = "I2S Audio", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "88pm860x-i2s", - .platform_name = "pxa-pcm-audio", - .codec_name = "88pm860x-codec", - .init = evb3_pm860x_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &evb3_i2s_ops, - }, -}; - -static struct snd_soc_card snd_soc_card_evb3 = { - .name = "Tavor EVB3", - .owner = THIS_MODULE, - .dai_link = evb3_dai, - .num_links = ARRAY_SIZE(evb3_dai), - - .dapm_widgets = evb3_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(evb3_dapm_widgets), - .dapm_routes = evb3_audio_map, - .num_dapm_routes = ARRAY_SIZE(evb3_audio_map), -}; - -static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); - snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - - /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); - - /* headphone, microphone detection & headset short detection */ - pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, - SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); - pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); - return 0; -} - -static int __init tavorevb3_init(void) -{ - int ret; - - if (!machine_is_tavorevb3()) - return -ENODEV; - evb3_snd_device = platform_device_alloc("soc-audio", -1); - if (!evb3_snd_device) - return -ENOMEM; - - platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3); - - ret = platform_device_add(evb3_snd_device); - if (ret) - platform_device_put(evb3_snd_device); - - return ret; -} - -static void __exit tavorevb3_exit(void) -{ - platform_device_unregister(evb3_snd_device); -} - -module_init(tavorevb3_init); -module_exit(tavorevb3_exit); - -MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); -MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3"); -MODULE_LICENSE("GPL"); -- 1.7.11.7

4 3

[alsa-devel] [PATCH 01/11] ASoC: ep93xx-i2s: Staticize non exported struct
by Lars-Peter Clausen 15 May '13

15 May '13

The ep93xx_i2s_dma_data struct is not used outside of ep93xx-i2s.c, so make it static. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Ryan Mallon <ryan(a)bluewatersys.com> --- sound/soc/cirrus/ep93xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index ef731e6..17ad70b 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -63,7 +63,7 @@ struct ep93xx_i2s_info { void __iomem *regs; }; -struct ep93xx_dma_data ep93xx_i2s_dma_data[] = { +static struct ep93xx_dma_data ep93xx_i2s_dma_data[] = { [SNDRV_PCM_STREAM_PLAYBACK] = { .name = "i2s-pcm-out", .port = EP93XX_DMA_I2S1, -- 1.8.0

3 21

[alsa-devel] Debian patches for ALSA
by Jordi Mallach 15 May '13

15 May '13

Hi! I've been wanting to put some order in the collection of patches we've been building up in Debian due to me neglecting forwarding anything for far too long. Let's try to correct this. I have a number of patches, most of which should be trivial and no-brainers, all updated to 1.0.27. I'm going to outline all the individual patches below. The patch tracker has the patches just as I would send them upstream, but if you prefer git formatted patches, I can go ahead and do that too, if it makes your life easier. Thanks! ALSA-LIB: http://patch-tracker.debian.org/package/alsa-lib/1.0.27-3 =================================================================== - http://patch-tracker.debian.org/patch/series/view/alsa-lib/1.0.27-3/add-mai… add-maintainer-mode.patch is something I'd like to see in all ALSA configure scripts: have maintainer mode enabled, but add the possibility to disable it via --disable-maintainer-mode. See http://blogs.gnome.org/desrt/2011/09/08/am_maintainer_mode-is-not-cool/ for reference. - http://patch-tracker.debian.org/patch/series/view/alsa-lib/1.0.27-3/multiar… The patch header is self explanatory: it fixes the plugin lookup path for Debian/Ubuntu's multiarch paths. - http://patch-tracker.debian.org/patch/series/view/alsa-lib/1.0.27-3/spellin… Little to explain here. ;) ALSA-UTILS: http://patch-tracker.debian.org/package/alsa-utils/1.0.27-2 ======================================================================= - http://patch-tracker.debian.org/patch/series/view/alsa-utils/1.0.27-2/spell… Again, trivial. - http://patch-tracker.debian.org/patch/series/view/alsa-utils/1.0.27-2/fix_m… Even more trivial. - http://patch-tracker.debian.org/patch/series/view/alsa-utils/1.0.27-2/utils… Fixes a build failure when using -Werror=format-security, which is default in Debian and Uubntu's toolchain. - http://patch-tracker.debian.org/patch/series/view/alsa-utils/1.0.27-2/alsa-… This one I want to discuss. We think the After condition is wrong, as it's depending on an optional service (alsa-state might not happen if the config file is not present). Instead, it should be checking for sysinit, to ensure local-fs. - http://patch-tracker.debian.org/patch/series/view/alsa-utils/1.0.27-2/manpa… Extensive formatting fixes for all alsa-utils manpages, plus an English improvement (allows to → allows one to). ALSA-PLUGINS: http://patch-tracker.debian.org/package/alsa-plugins/1.0.27-2 =========================================================================== - http://patch-tracker.debian.org/patch/series/view/alsa-plugins/1.0.27-2/arc… Fix linking to avoid undefined symbols in arcam-av. - http://patch-tracker.debian.org/patch/series/view/alsa-plugins/1.0.27-2/use… The libav changes in 1.0.27 were insufficient: a52 uses av_free(), which is part of libavutil. The patch needs to be completed to look for the correct header files, though, but it's enough to get the package compiling in our setup. ALSA-TOOLS: http://patch-tracker.debian.org/package/alsa-tools/1.0.27-2 - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/init_… Fix trivial bashisms. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/manpa… Spelling fixes. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/ld10k… More -Werror=format-security issues. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/hda-v… Extend check for sys/io.h for other platforms in Debian that don't support it. Alternatively, the check could be reversed to chef for the architectures currently know to have the header, as stated in the comment. This fixes the build in a number of Debian architectures. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/envy2… This one is to be discussed. envy24control currently creates a visible directory in $HOME, which is quite unacceptable. This patch just moves the directory to a dotdir, but an upgrade path might need to be coded. I would just move the thing away. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/as10k… For convenience to our build systems, add an alias for clean. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/usx2y… While not ready for submission, this one probably reveals a hardcoded path that should be made configurable, if not all distros store firmware in this dir. Not sure about Fedora, at least. /etc/hotplug looks plain old, anyway. ALSA-OSS: http://patch-tracker.debian.org/package/alsa-oss/1.0.25-1 =================================================================== - http://patch-tracker.debian.org/patch/series/view/alsa-oss/1.0.25-1/aoss_wr… Make it obvious aoss is not doing anything useful if ALSA is not present. - http://patch-tracker.debian.org/patch/series/view/alsa-oss/1.0.25-1/aoss_li… Fix path to libaoss on the multiarch case. Thanks, Jordi -- Jordi Mallach Pérez -- Debian developer http://www.debian.org/ jordi(a)sindominio.net jordi(a)debian.org http://www.sindominio.net/ GnuPG public key information available at http://oskuro.net/ -- Jordi Mallach Pérez -- Debian developer http://www.debian.org/ jordi(a)sindominio.net jordi(a)debian.org http://www.sindominio.net/ GnuPG public key information available at http://oskuro.net/

3 3

[alsa-devel] [PATCH] conf/ucm: DAISY-I2S: Set default filter mode for HiFi.
by Dylan Reid 14 May '13

14 May '13

Default to music mode filter for the HiFi use case on the Samsung ARM Chromebook. This mode is better at 44.1k and 48k audio than the "Voice" setting. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- src/conf/ucm/DAISY-I2S/HiFi.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/ucm/DAISY-I2S/HiFi.conf b/src/conf/ucm/DAISY-I2S/HiFi.conf index 66b0afa..d8c06ad 100644 --- a/src/conf/ucm/DAISY-I2S/HiFi.conf +++ b/src/conf/ucm/DAISY-I2S/HiFi.conf @@ -11,6 +11,7 @@ SectionVerb { cset "name='Left ADC Mixer MIC2 Switch' on" cset "name='Right ADC Mixer MIC2 Switch' on" cset "name='EQ1 Mode' Default" + cset "name='DAI1 Filter Mode' Music" ] DisableSequence [ ] -- 1.8.1.3.605.g02339dd

2 1

Re: [alsa-devel] [PATCH 1/5] DAISY-I2S: add Samsung ARM Chromebook UCM config from ChromeOS
by Dylan Reid 14 May '13

14 May '13

On Fri, May 10, 2013 at 5:51 AM, Marcin Juszkiewicz <marcin(a)juszkiewicz.com.pl> wrote: > Signed-off-by: Marcin Juszkiewicz <marcin(a)juszkiewicz.com.pl> > --- > DAISY-I2S/DAISY-I2S.conf | 6 +++++ > DAISY-I2S/HiFi.conf | 61 ++++++++++++++++++++++++++++++++++++++++++++++++ > 2 files changed, 67 insertions(+) > create mode 100644 DAISY-I2S/DAISY-I2S.conf > create mode 100644 DAISY-I2S/HiFi.conf > > diff --git a/DAISY-I2S/DAISY-I2S.conf b/DAISY-I2S/DAISY-I2S.conf > new file mode 100644 > index 0000000..0f4b5fc > --- /dev/null > +++ b/DAISY-I2S/DAISY-I2S.conf > @@ -0,0 +1,6 @@ > +Comment "Daisy internal card" > + > +SectionUseCase."HiFi" { > + File "HiFi.conf" > + Comment "Default" > +} > diff --git a/DAISY-I2S/HiFi.conf b/DAISY-I2S/HiFi.conf > new file mode 100644 > index 0000000..66b0afa > --- /dev/null > +++ b/DAISY-I2S/HiFi.conf > @@ -0,0 +1,61 @@ > +SectionVerb { > + EnableSequence [ > + cdev "hw:DAISYI2S" > + cset "name='Left Speaker Mixer Left DAC1 Switch' on" > + cset "name='Right Speaker Mixer Right DAC1 Switch' on" > + cset "name='Left Headphone Mixer Left DAC1 Switch' on" > + cset "name='Right Headphone Mixer Right DAC1 Switch' on" > + cset "name='DMIC1 Left Capture Switch' on" > + cset "name='DMIC1 Right Capture Switch' on" > + cset "name='MIC2 External Mic Switch' on" > + cset "name='Left ADC Mixer MIC2 Switch' on" > + cset "name='Right ADC Mixer MIC2 Switch' on" > + cset "name='EQ1 Mode' Default" The current version has one additional line here: cset "name='DAI1 Filter Mode' Music" Thanks for finding a potential home for this. > + ] > + DisableSequence [ > + ] > +} > + > +SectionDevice."Headphone".0 { > + Value { > + JackName "DAISY-I2S Headphone Jack" > + } > + > + EnableSequence [ > + cdev "hw:DAISYI2S" > + cset "name='EQ1 Switch' off" > + cset "name='Left Headphone Mixer Left DAC1 Switch' on" > + cset "name='Right Headphone Mixer Right DAC1 Switch' on" > + ] > + DisableSequence [ > + cdev "hw:DAISYI2S" > + cset "name='EQ1 Mode' Default" > + cset "name='EQ1 Switch' on" > + cset "name='Left Speaker Mixer Left DAC1 Switch' on" > + cset "name='Right Speaker Mixer Right DAC1 Switch' on" > + ] > +} > + > +SectionDevice."Mic".0 { > + Value { > + JackName "DAISY-I2S Mic Jack" > + } > + > + EnableSequence [ > + cdev "hw:DAISYI2S" > + > + cset "name='DMIC1 Left Capture Switch' off" > + cset "name='DMIC1 Right Capture Switch' off" > + cset "name='Left ADC Mixer MIC2 Switch' on" > + cset "name='Right ADC Mixer MIC2 Switch' on" > + ] > + > + DisableSequence [ > + cdev "hw:DAISYI2S" > + > + cset "name='Left ADC Mixer MIC2 Switch' off" > + cset "name='Right ADC Mixer MIC2 Switch' off" > + cset "name='DMIC1 Left Capture Switch' on" > + cset "name='DMIC1 Right Capture Switch' on" > + ] > +} > -- > 1.8.1.2 > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 1

Re: [alsa-devel] UCM profiles taken from Ubuntu
by Liam Girdwood 14 May '13

14 May '13

On Fri, 2013-05-10 at 15:56 +0200, Jordi Mallach wrote: > El dv 10 de 05 de 2013 a les 15:17 +0200, en/na Takashi Iwai va > escriure: > > At Fri, 10 May 2013 14:51:05 +0200, > > Marcin Juszkiewicz wrote: > > > > > > When I bought Samsung ARM Chromebook few months ago I had no idea > > > about UCM profiles and burnt speakers (left is dead, right is > > > resting). Then I learnt, took ones from ChromeOS and added them into > > > Ubuntu so other users will be a bit more safe. > > > > > > Later I had a discussion with some Debian, Ubuntu, Fedora developers > > > about how to solve such problems and how to keep UCM profiles shared > > > between distributions. > > > > > > Liam Girdwood pointed me to not used git tree at ALSA Project server > > > and finally I reminded myself about it. > > > > > > The idea is to collect UCM profiles, keep them in one place and create > > > alsa-ucm package in each distribution. > > Thanks Marcin! This part of the Debian←→Ubuntu delta was something I > wanted to bring up with upstream after the wheezy release. > I'll echo that thought, many thanks Marcin ! > > Would it make sense to put into alsa-lib.git? Since UCM itself was > > already integrated into alsa-lib and there are already tons of > > card-based config files in alsa-lib/src/conf/*, these also fit to a > > similar directory there, I think. > > > > They could be in a separate repo, but UCM config is coupled with UCM > > code, so we may keep in the single repo. > > It makes sense to me, fwiw. > Agreed, the initial intention of the separate repo was to do frequent releases that would be likely out of sync alsa-lib. But I think combining is the best way forward now. Takashi, please feel free to remove the other repo (I think it's mostly empty anyway). Liam

3 2

[alsa-devel] [PATCH v2 8/8] sound/oxygen_io: take msecs_to_jiffies_timeout into use
by Imre Deak 14 May '13

14 May '13

Use msecs_to_jiffies_timeout instead of open-coding the same. Signed-off-by: Imre Deak <imre.deak(a)intel.com> --- sound/pci/oxygen/oxygen_io.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 521eae4..51bfcd0 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -108,7 +108,7 @@ static int oxygen_ac97_wait(struct oxygen *chip, unsigned int mask) wait_event_timeout(chip->ac97_waitqueue, ({ status |= oxygen_read8(chip, OXYGEN_AC97_INTERRUPT_STATUS); status & mask; }), - msecs_to_jiffies(1) + 1); + msecs_to_jiffies_timeout(1)); /* * Check even after a timeout because this function should not require * the AC'97 interrupt to be enabled. -- 1.7.10.4

2 1

[alsa-devel] [patch] ASoC: wm_adsp: memory leak in wm_adsp_create_control()
by Dan Carpenter 14 May '13

14 May '13

There are two return paths which don't kfree(name). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 1378306..d715c8e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -836,7 +836,8 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, region_name = "ZM"; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_name; } snprintf(name, PAGE_SIZE, "DSP%d %s %x", @@ -847,7 +848,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, if (!strcmp(ctl->name, name)) { if (!ctl->enabled) ctl->enabled = 1; - return 0; + goto found; } } @@ -887,6 +888,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, INIT_WORK(&ctl_work->work, wm_adsp_ctl_work); schedule_work(&ctl_work->work); +found: kfree(name); return 0;

3 2

[alsa-devel] [PATCH RFT] ASoC: da7213: Fix setting dmic_samplephase and dmic_clk_rate
by Axel Lin 14 May '13

14 May '13

When set dmic_samplephase and dmic_clk_rate bits for dmic_cfg, current code checks pdata->dmic_data_sel which is wrong. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- This looks like a copy-paste bug. sound/soc/codecs/da7213.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 41230ad..4a6f1da 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1488,17 +1488,17 @@ static int da7213_probe(struct snd_soc_codec *codec) DA7213_DMIC_DATA_SEL_SHIFT); break; } - switch (pdata->dmic_data_sel) { + switch (pdata->dmic_samplephase) { case DA7213_DMIC_SAMPLE_ON_CLKEDGE: case DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE: - dmic_cfg |= (pdata->dmic_data_sel << + dmic_cfg |= (pdata->dmic_samplephase << DA7213_DMIC_SAMPLEPHASE_SHIFT); break; } - switch (pdata->dmic_data_sel) { + switch (pdata->dmic_clk_rate) { case DA7213_DMIC_CLK_3_0MHZ: case DA7213_DMIC_CLK_1_5MHZ: - dmic_cfg |= (pdata->dmic_data_sel << + dmic_cfg |= (pdata->dmic_clk_rate << DA7213_DMIC_CLK_RATE_SHIFT); break; } -- 1.8.1.2

3 2

[alsa-devel] [PATCH 1/3] ASoC: Move snd_soc_set_runtime_hwparams() to soc-pcm.c
by Lars-Peter Clausen 14 May '13

14 May '13

snd_soc_set_runtime_hwparams() is the only PCM related function that lives in soc-core.c. All other PCM related functions live in soc-pcm.c, so move snd_soc_set_runtime_hwparams() over as well for a bit more consistency. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 23 ----------------------- sound/soc/soc-pcm.c | 23 +++++++++++++++++++++++ 2 files changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 308895a..9d95ef5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2220,29 +2220,6 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, EXPORT_SYMBOL_GPL(snd_soc_test_bits); /** - * snd_soc_set_runtime_hwparams - set the runtime hardware parameters - * @substream: the pcm substream - * @hw: the hardware parameters - * - * Sets the substream runtime hardware parameters. - */ -int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, - const struct snd_pcm_hardware *hw) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.info = hw->info; - runtime->hw.formats = hw->formats; - runtime->hw.period_bytes_min = hw->period_bytes_min; - runtime->hw.period_bytes_max = hw->period_bytes_max; - runtime->hw.periods_min = hw->periods_min; - runtime->hw.periods_max = hw->periods_max; - runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; - runtime->hw.fifo_size = hw->fifo_size; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); - -/** * snd_soc_cnew - create new control * @_template: control template * @data: control private data diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 73bb8ee..058b403 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -33,6 +33,29 @@ #define DPCM_MAX_BE_USERS 8 +/** + * snd_soc_set_runtime_hwparams - set the runtime hardware parameters + * @substream: the pcm substream + * @hw: the hardware parameters + * + * Sets the substream runtime hardware parameters. + */ +int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, + const struct snd_pcm_hardware *hw) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + runtime->hw.info = hw->info; + runtime->hw.formats = hw->formats; + runtime->hw.period_bytes_min = hw->period_bytes_min; + runtime->hw.period_bytes_max = hw->period_bytes_max; + runtime->hw.periods_min = hw->periods_min; + runtime->hw.periods_max = hw->periods_max; + runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; + runtime->hw.fifo_size = hw->fifo_size; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); + /* DPCM stream event, send event to FE and all active BEs. */ static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) -- 1.8.0

3 4

[alsa-devel] [PATCH] ASoC: tegra: always use clk_get() in utility code
by Stephen Warren 14 May '13

14 May '13

From: Stephen Warren <swarren(a)nvidia.com> Now that all of the Tegra device trees have been updated to represent the required audio clocks, remove the compatibility code from the Tegra ASoC utility code, and always use clk_get() rather than clk_get_sys(). Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_asoc_utils.c | 23 +++++------------------ 1 file changed, 5 insertions(+), 18 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 24fb001b..d173880 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -173,7 +173,6 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, struct device *dev) { int ret; - bool new_clocks = false; data->dev = dev; @@ -181,40 +180,28 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else if (of_machine_is_compatible("nvidia,tegra30")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; - else if (of_machine_is_compatible("nvidia,tegra114")) { + else if (of_machine_is_compatible("nvidia,tegra114")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA114; - new_clocks = true; - } else { + else { dev_err(data->dev, "SoC unknown to Tegra ASoC utils\n"); return -EINVAL; } - if (new_clocks) - data->clk_pll_a = clk_get(dev, "pll_a"); - else - data->clk_pll_a = clk_get_sys(NULL, "pll_a"); + data->clk_pll_a = clk_get(dev, "pll_a"); if (IS_ERR(data->clk_pll_a)) { dev_err(data->dev, "Can't retrieve clk pll_a\n"); ret = PTR_ERR(data->clk_pll_a); goto err; } - if (new_clocks) - data->clk_pll_a_out0 = clk_get(dev, "pll_a_out0"); - else - data->clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); + data->clk_pll_a_out0 = clk_get(dev, "pll_a_out0"); if (IS_ERR(data->clk_pll_a_out0)) { dev_err(data->dev, "Can't retrieve clk pll_a_out0\n"); ret = PTR_ERR(data->clk_pll_a_out0); goto err_put_pll_a; } - if (new_clocks) - data->clk_cdev1 = clk_get(dev, "mclk"); - else if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) - data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); - else - data->clk_cdev1 = clk_get_sys("extern1", NULL); + data->clk_cdev1 = clk_get(dev, "mclk"); if (IS_ERR(data->clk_cdev1)) { dev_err(data->dev, "Can't retrieve clk cdev1\n"); ret = PTR_ERR(data->clk_cdev1); -- 1.7.10.4

2 1

[alsa-devel] [PATCH 1/8] ASoC: generic-dmaengine-pcm: Add support for half-duplex
by Lars-Peter Clausen 13 May '13

13 May '13

Some platforms which are half-duplex share the same DMA channel between the playback and capture stream. Add support for this to the generic dmaengine PCM driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/dmaengine_pcm.h | 5 ++++ sound/soc/soc-generic-dmaengine-pcm.c | 43 ++++++++++++++++++++++++----------- 2 files changed, 35 insertions(+), 13 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index b1d1150..f11c35c 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -91,6 +91,11 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * bytes that are still left to transfer. */ #define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(2) +/* + * The PCM is half duplex and the DMA channel is shared between capture and + * playback. + */ +#define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ae0c37e..5fd5ed4 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -29,7 +29,7 @@ struct dmaengine_pcm { struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; const struct snd_dmaengine_pcm_config *config; struct snd_soc_platform platform; - bool compat; + unsigned int flags; }; static struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p) @@ -128,6 +128,9 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) + return pcm->chan[0]; + if (pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); @@ -148,7 +151,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - if (!pcm->chan[i] && pcm->compat) { + if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); } @@ -215,6 +218,25 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_CAPTURE] = "rx", }; +static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, + struct device_node *of_node) +{ + unsigned int i; + + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !of_node) + return; + + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { + pcm->chan[0] = of_dma_request_slave_channel(of_node, "tx_rx"); + pcm->chan[1] = pcm->chan[0]; + } else { + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { + pcm->chan[i] = of_dma_request_slave_channel(of_node, + dmaengine_pcm_dma_channel_names[i]); + } + } +} + /** * snd_dmaengine_pcm_register - Register a dmaengine based PCM device * @dev: The parent device for the PCM device @@ -225,23 +247,15 @@ int snd_dmaengine_pcm_register(struct device *dev, const struct snd_dmaengine_pcm_config *config, unsigned int flags) { struct dmaengine_pcm *pcm; - unsigned int i; pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); if (!pcm) return -ENOMEM; pcm->config = config; + pcm->flags = flags; - if (flags & SND_DMAENGINE_PCM_FLAG_COMPAT) - pcm->compat = true; - - if (!(flags & SND_DMAENGINE_PCM_FLAG_NO_DT) && dev->of_node) { - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = of_dma_request_slave_channel(dev->of_node, - dmaengine_pcm_dma_channel_names[i]); - } - } + dmaengine_pcm_request_chan_of(pcm, dev->of_node); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) return snd_soc_add_platform(dev, &pcm->platform, @@ -272,8 +286,11 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_platform_to_pcm(platform); for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - if (pcm->chan[i]) + if (pcm->chan[i]) { dma_release_channel(pcm->chan[i]); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; + } } snd_soc_remove_platform(platform); -- 1.8.0

3 19

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Simplify the handling of VDDD
by Fabio Estevam 13 May '13

13 May '13

sgtl5000 has two power supplies that are mandatory: VDDA and VDDIO. VDDD is an optional supply and can be left unconnected and in this case an internal regulator is used. Current code always use the internal regulator even if an external VDDD supply is provided and the sglt5000 revision is equal or greater than 0x11. It is better to simplify the code to handle VDDD and always turn on the internal regulator since all the production sgtl5000 have revision equal or greater 0x11. While at it, turn on the regulators prior to reading the SGTL5000_CHIP_ID register. Tested on a mx51evk board that has VDDA and VDDIO provided by the PMIC. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 394 ++++++++----------------------------------- 1 file changed, 70 insertions(+), 324 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1c3b20f..2e56d00 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -65,45 +65,12 @@ static const struct reg_default sgtl5000_reg_defaults[] = { enum sgtl5000_regulator_supplies { VDDA, VDDIO, - VDDD, SGTL5000_SUPPLY_NUM }; -/* vddd is optional supply */ static const char *supply_names[SGTL5000_SUPPLY_NUM] = { "VDDA", "VDDIO", - "VDDD" -}; - -#define LDO_CONSUMER_NAME "VDDD_LDO" -#define LDO_VOLTAGE 1200000 - -static struct regulator_consumer_supply ldo_consumer[] = { - REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL), -}; - -static struct regulator_init_data ldo_init_data = { - .constraints = { - .min_uV = 1200000, - .max_uV = 1200000, - .valid_modes_mask = REGULATOR_MODE_NORMAL, - .valid_ops_mask = REGULATOR_CHANGE_STATUS, - }, - .num_consumer_supplies = 1, - .consumer_supplies = &ldo_consumer[0], -}; - -/* - * sgtl5000 internal ldo regulator, - * enabled when VDDD not provided - */ -struct ldo_regulator { - struct regulator_desc desc; - struct regulator_dev *dev; - int voltage; - void *codec_data; - bool enabled; }; /* sgtl5000 private structure in codec */ @@ -112,7 +79,7 @@ struct sgtl5000_priv { int master; /* i2s master or not */ int fmt; /* i2s data format */ struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; - struct ldo_regulator *ldo; + struct regulator *vddd; struct regmap *regmap; }; @@ -732,157 +699,6 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -#ifdef CONFIG_REGULATOR -static int ldo_regulator_is_enabled(struct regulator_dev *dev) -{ - struct ldo_regulator *ldo = rdev_get_drvdata(dev); - - return ldo->enabled; -} - -static int ldo_regulator_enable(struct regulator_dev *dev) -{ - struct ldo_regulator *ldo = rdev_get_drvdata(dev); - struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; - int reg; - - if (ldo_regulator_is_enabled(dev)) - return 0; - - /* set regulator value firstly */ - reg = (1600 - ldo->voltage / 1000) / 50; - reg = clamp(reg, 0x0, 0xf); - - /* amend the voltage value, unit: uV */ - ldo->voltage = (1600 - reg * 50) * 1000; - - /* set voltage to register */ - snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - SGTL5000_LINREG_VDDD_MASK, reg); - - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINEREG_D_POWERUP, - SGTL5000_LINEREG_D_POWERUP); - - /* when internal ldo enabled, simple digital power can be disabled */ - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINREG_SIMPLE_POWERUP, - 0); - - ldo->enabled = 1; - return 0; -} - -static int ldo_regulator_disable(struct regulator_dev *dev) -{ - struct ldo_regulator *ldo = rdev_get_drvdata(dev); - struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; - - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINEREG_D_POWERUP, - 0); - - /* clear voltage info */ - snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - SGTL5000_LINREG_VDDD_MASK, 0); - - ldo->enabled = 0; - - return 0; -} - -static int ldo_regulator_get_voltage(struct regulator_dev *dev) -{ - struct ldo_regulator *ldo = rdev_get_drvdata(dev); - - return ldo->voltage; -} - -static struct regulator_ops ldo_regulator_ops = { - .is_enabled = ldo_regulator_is_enabled, - .enable = ldo_regulator_enable, - .disable = ldo_regulator_disable, - .get_voltage = ldo_regulator_get_voltage, -}; - -static int ldo_regulator_register(struct snd_soc_codec *codec, - struct regulator_init_data *init_data, - int voltage) -{ - struct ldo_regulator *ldo; - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - struct regulator_config config = { }; - - ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); - - if (!ldo) { - dev_err(codec->dev, "failed to allocate ldo_regulator\n"); - return -ENOMEM; - } - - ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL); - if (!ldo->desc.name) { - kfree(ldo); - dev_err(codec->dev, "failed to allocate decs name memory\n"); - return -ENOMEM; - } - - ldo->desc.type = REGULATOR_VOLTAGE; - ldo->desc.owner = THIS_MODULE; - ldo->desc.ops = &ldo_regulator_ops; - ldo->desc.n_voltages = 1; - - ldo->codec_data = codec; - ldo->voltage = voltage; - - config.dev = codec->dev; - config.driver_data = ldo; - config.init_data = init_data; - - ldo->dev = regulator_register(&ldo->desc, &config); - if (IS_ERR(ldo->dev)) { - int ret = PTR_ERR(ldo->dev); - - dev_err(codec->dev, "failed to register regulator\n"); - kfree(ldo->desc.name); - kfree(ldo); - - return ret; - } - sgtl5000->ldo = ldo; - - return 0; -} - -static int ldo_regulator_remove(struct snd_soc_codec *codec) -{ - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - struct ldo_regulator *ldo = sgtl5000->ldo; - - if (!ldo) - return 0; - - regulator_unregister(ldo->dev); - kfree(ldo->desc.name); - kfree(ldo); - - return 0; -} -#else -static int ldo_regulator_register(struct snd_soc_codec *codec, - struct regulator_init_data *init_data, - int voltage) -{ - dev_err(codec->dev, "this setup needs regulator support in the kernel\n"); - return -EINVAL; -} - -static int ldo_regulator_remove(struct snd_soc_codec *codec) -{ - return 0; -} -#endif - /* * set dac bias * common state changes: @@ -910,6 +726,11 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, sgtl5000->supplies); if (ret) return ret; + if (sgtl5000->vddd) { + ret = regulator_enable(sgtl5000->vddd); + if (ret) + return ret; + } udelay(10); } @@ -917,6 +738,8 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + if (sgtl5000->vddd) + regulator_disable(sgtl5000->vddd); break; } @@ -1136,7 +959,10 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) vdda = regulator_get_voltage(sgtl5000->supplies[VDDA].consumer); vddio = regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer); - vddd = regulator_get_voltage(sgtl5000->supplies[VDDD].consumer); + if (sgtl5000->vddd) + vddd = regulator_get_voltage(sgtl5000->vddd); + else + vddd = 0; vdda = vdda / 1000; vddio = vddio / 1000; @@ -1198,15 +1024,14 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) * simple digital supply should be clear to get * proper VDDD voltage. */ - if (ana_pwr & SGTL5000_LINEREG_D_POWERUP) - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINEREG_D_POWERUP, + SGTL5000_LINEREG_D_POWERUP); + + /* when internal ldo enabled, simple digital power can be disabled */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_LINREG_SIMPLE_POWERUP, 0); - else - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINREG_SIMPLE_POWERUP | - SGTL5000_STARTUP_POWERUP, - 0); /* * set ADC/DAC VAG to vdda / 2, @@ -1245,110 +1070,6 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) return 0; } -static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) -{ - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - int ret; - - /* set internal ldo to 1.2v */ - ret = ldo_regulator_register(codec, &ldo_init_data, LDO_VOLTAGE); - if (ret) { - dev_err(codec->dev, - "Failed to register vddd internal supplies: %d\n", ret); - return ret; - } - - sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - if (ret) { - ldo_regulator_remove(codec); - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - - dev_info(codec->dev, "Using internal LDO instead of VDDD\n"); - return 0; -} - -static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) -{ - int reg; - int ret; - int rev; - int i; - int external_vddd = 0; - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - - for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) - sgtl5000->supplies[i].supply = supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (!ret) - external_vddd = 1; - else { - ret = sgtl5000_replace_vddd_with_ldo(codec); - if (ret) - return ret; - } - - ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (ret) - goto err_regulator_free; - - /* wait for all power rails bring up */ - udelay(10); - - /* - * workaround for revision 0x11 and later, - * roll back to use internal LDO - */ - - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); - if (ret) - goto err_regulator_disable; - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - - if (external_vddd && rev >= 0x11) { - /* disable all regulator first */ - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - /* free VDDD regulator */ - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - ret = sgtl5000_replace_vddd_with_ldo(codec); - if (ret) - return ret; - - ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (ret) - goto err_regulator_free; - - /* wait for all power rails bring up */ - udelay(10); - } - - return 0; - -err_regulator_disable: - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); -err_regulator_free: - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (external_vddd) - ldo_regulator_remove(codec); - return ret; - -} - static int sgtl5000_probe(struct snd_soc_codec *codec) { int ret; @@ -1362,14 +1083,10 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) return ret; } - ret = sgtl5000_enable_regulators(codec); - if (ret) - return ret; - /* power up sgtl5000 */ ret = sgtl5000_set_power_regs(codec); if (ret) - goto err; + return ret; /* enable small pop, introduce 400ms delay in turning off */ snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, @@ -1412,32 +1129,16 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) /* leading to standby state */ ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret) - goto err; + return ret; return 0; -err: - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - ldo_regulator_remove(codec); - - return ret; } static int sgtl5000_remove(struct snd_soc_codec *codec) { - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - ldo_regulator_remove(codec); - return 0; } @@ -1497,30 +1198,63 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct sgtl5000_priv *sgtl5000; - int ret, reg, rev; + int ret, reg, rev, i; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); if (!sgtl5000) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) + sgtl5000->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(&client->dev, + ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + if (ret) { + dev_err(&client->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) { + dev_err(&client->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + sgtl5000->vddd = devm_regulator_get(&client->dev, "VDDD"); + if (!IS_ERR(sgtl5000->vddd)) { + ret = regulator_enable(sgtl5000->vddd); + if (ret) { + dev_err(&client->dev, + "Failed to enable VDDD: %d\n", ret); + goto err_bulk; + } + } else { + sgtl5000->vddd = NULL; + } + + /* wait for all power rails bring up */ + udelay(10); + sgtl5000->regmap = devm_regmap_init_i2c(client, &sgtl5000_regmap); if (IS_ERR(sgtl5000->regmap)) { ret = PTR_ERR(sgtl5000->regmap); dev_err(&client->dev, "Failed to allocate regmap: %d\n", ret); - return ret; + goto err_vddd; } /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); if (ret) - return ret; + goto err_vddd; if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != SGTL5000_PARTID_PART_ID) { dev_err(&client->dev, "Device with ID register %x is not a sgtl5000\n", reg); - return -ENODEV; + ret = -ENODEV; + goto err_vddd; } rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; @@ -1531,10 +1265,22 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, /* Ensure sgtl5000 will start with sane register values */ ret = sgtl5000_fill_defaults(sgtl5000); if (ret) - return ret; + goto err_vddd; ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) + goto err_vddd; + + return 0; + +err_vddd: + if (sgtl5000->vddd) + regulator_disable(sgtl5000->vddd); +err_bulk: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + return ret; } -- 1.7.9.5

2 4

Re: [alsa-devel] [PATCH 11/11] sound/oxygen_io: take msecs_to_jiffies_min into use
by Takashi Iwai 13 May '13

13 May '13

At Fri, 10 May 2013 15:13:29 +0300, Imre Deak wrote: > > Use msecs_to_jiffies_min instead of open-coding the same. > > Signed-off-by: Imre Deak <imre.deak(a)intel.com> > --- > sound/pci/oxygen/oxygen_io.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c > index 521eae4..132ecbe 100644 > --- a/sound/pci/oxygen/oxygen_io.c > +++ b/sound/pci/oxygen/oxygen_io.c > @@ -108,7 +108,7 @@ static int oxygen_ac97_wait(struct oxygen *chip, unsigned int mask) > wait_event_timeout(chip->ac97_waitqueue, > ({ status |= oxygen_read8(chip, OXYGEN_AC97_INTERRUPT_STATUS); > status & mask; }), > - msecs_to_jiffies(1) + 1); > + msecs_to_jiffies_min(1)); This would change the behavior, I guess. (Though, I'm not sure whether the original code was intentional.) And, isn't msecs_to_jiffies_min(1) identical with msecs_to_jiffies(1)? thanks, Takashi

3 3

[alsa-devel] [PATCH] sound: Fix make allmodconfig on MIPS correctly
by Paul Bolle 13 May '13

13 May '13

Commit d4702b189c ("sound: Fix make allmodconfig on MIPS") added a (negative) dependency on ISA_DMA_SUPPORT_BROKEN. Since that Kconfig symbol doesn't exist, this dependency will always evaluate to true. Apparently GENERIC_ISA_DMA_SUPPORT_BROKEN was meant to be used here. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Tested only with "git grep". sound/oss/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 51c4ba9..1a96402 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -250,7 +250,7 @@ config MSND_FIFOSIZE menuconfig SOUND_OSS tristate "OSS sound modules" depends on ISA_DMA_API && VIRT_TO_BUS - depends on !ISA_DMA_SUPPORT_BROKEN + depends on !GENERIC_ISA_DMA_SUPPORT_BROKEN help OSS is the Open Sound System suite of sound card drivers. They make sound programming easier since they provide a common API. Say Y or -- 1.7.11.7

2 1

[alsa-devel] [PATCH -next] ASoC: mid-x86: add missing iounmap() to probe error and remove
by Wei Yongjun 13 May '13

13 May '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add the missing iounmap() before return from snd_mfld_mc_probe() in the error handling case and to snd_mfld_mc_remove(). Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/mid-x86/mfld_machine.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 78d5825..8206e84 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -397,7 +397,7 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); if (ret_val) { pr_err("cannot register IRQ\n"); - goto unalloc; + goto unmap; } /* register the soc card */ snd_soc_card_mfld.dev = &pdev->dev; @@ -412,6 +412,8 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) freeirq: free_irq(irq, mc_drv_ctx); +unmap: + iounmap(mc_drv_ctx->int_base); unalloc: kfree(mc_drv_ctx); return ret_val; @@ -424,6 +426,7 @@ static int snd_mfld_mc_remove(struct platform_device *pdev) pr_debug("snd_mfld_mc_remove called\n"); free_irq(platform_get_irq(pdev, 0), mc_drv_ctx); snd_soc_unregister_card(&snd_soc_card_mfld); + iounmap(mc_drv_ctx->int_base); kfree(mc_drv_ctx); return 0; }

2 1

[alsa-devel] [PATCH 0/5] Power-well API implementation for Haswell
by Wang Xingchao 13 May '13

13 May '13

This patchset intended to Add power-well api support for Haswell. For Intel Haswell , “power well” in GPU has impact for both Display HD-A controller and codecs. Gfx driver has power well feature enaled but donot think much about audio side. The issue is if gfx driver disabled power well, audio side would lose power and crash. David helped a initial patch to add external module patch_hdmi_i915, which built in only DRM_I915 and HDMI_CODEC enabled, This patchset was based on David’s work, thanks David! As codec depends on controller’s power, so we removed the pm callback temporary. So only HDA controller driver would such need to request/release power, the codec will always be safe. Gfx driver would shutdown power well in below cases, audio driver would be damaged as losing power totally without any notification. (There would be more cases not covered, welcome more comments here. i can verify the patchset under more cases) - Only eDP port active - HDMI monitor hot plug - System suspend Also there’re some calling from i915 to disable power well from time to time, this patchset would reject those operations if audio using power well, but it will record i915 driver’s request and shut down power if audio release it. Basically, here's the latest working status on my side: I tested the power well feature on Haswell ULT board, there's only one eDP port used. Without this patch and we enabled power well feature, gfx driver will shutdown power well, audio driver will crash. Applied this patch, display audio driver will request power well on before initialize the card. In gfx side, it will enable power well. Also power_save feature in audio side should be enabled, I set "5" second to let codec enter suspend when free for 5s long. Audio controller driver detects all codecs are suspended it will release power well and suspend too. Gfx driver will receive the request and shut down power well. If user trigger some audio operations(cat codec# info), audio controller driver will request power well again... If gfx side decided to disable power well now, while audio is in use, power well would be kept on. (i will send out a documentation which has some logs and analysis there.) Generally audio drvier will request power well on need and release it immediately after suspend. Gfx should make decision at his side. The new introduced module "hda-i915" has dependency on i915, only built in when i915 enabled. So it's safe for call. For non-Haswell platform, power well is no need atm. Even the module is built in wrongly, hda driver will not call the power well api. Also gfx will reject power well request at its side, so it's safe too. This patch could make sure it would not cause damage when snd-hda-intel and i915 module loaded in any order. If i915 module loaded at first, it's safe to get the symbol. If snd-hda-intel loaded first, it will try to load i915 manually and get notification when the power-well api are exported. Wang Xingchao (5): drm/915: Add private api for power well usage ALSA: hda - Add external module hda-i915 for power well ALSA: hda - Power well request/release for hda controller ALSA: hda - Fix module dependency with gfx i915 ALSA/i915: Check power well API existense before calling drivers/gpu/drm/i915/i915_dma.c | 3 + drivers/gpu/drm/i915/intel_drv.h | 2 + drivers/gpu/drm/i915/intel_pm.c | 139 ++++++++++++++++++++++++++++++++++++-- include/drm/i915_powerwell.h | 40 +++++++++++ sound/pci/hda/Kconfig | 13 ++++ sound/pci/hda/Makefile | 6 ++ sound/pci/hda/hda_i915.c | 92 +++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 15 ++++ sound/pci/hda/hda_intel.c | 22 ++++++ 9 files changed, 325 insertions(+), 7 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

5 21

[alsa-devel] [PATCH v2] ASoC: mxs: move to use generic DMA helper
by Shawn Guo 13 May '13

13 May '13

With mxs-dma converted to generic DMA bindings, let's move mxs-pcm to use it by removing flages SND_DMAENGINE_PCM_FLAG_NO_DT and SND_DMAENGINE_PCM_FLAG_COMPAT. As the result, those mxs custom dma params code can be removed now. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v1: * rebase against Mark's topic/mxs branch .../devicetree/bindings/sound/mxs-saif.txt | 17 ++++++++---- sound/soc/mxs/mxs-pcm.c | 18 ------------ sound/soc/mxs/mxs-pcm.h | 7 ----- sound/soc/mxs/mxs-saif.c | 29 +------------------- sound/soc/mxs/mxs-saif.h | 1 - 5 files changed, 12 insertions(+), 60 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt index c37ba61..7ba07a1 100644 --- a/Documentation/devicetree/bindings/sound/mxs-saif.txt +++ b/Documentation/devicetree/bindings/sound/mxs-saif.txt @@ -3,8 +3,11 @@ Required properties: - compatible: Should be "fsl,<chip>-saif" - reg: Should contain registers location and length -- interrupts: Should contain ERROR and DMA interrupts -- fsl,saif-dma-channel: APBX DMA channel for the SAIF +- interrupts: Should contain ERROR interrupt number +- dmas: DMA specifier, consisting of a phandle to DMA controller node + and SAIF DMA channel ID. + Refer to dma.txt and fsl-mxs-dma.txt for details. +- dma-names: Must be "rx-tx". Optional properties: - fsl,saif-master: phandle to the master SAIF. It's only required for @@ -23,14 +26,16 @@ aliases { saif0: saif@80042000 { compatible = "fsl,imx28-saif"; reg = <0x80042000 2000>; - interrupts = <59 80>; - fsl,saif-dma-channel = <4>; + interrupts = <59>; + dmas = <&dma_apbx 4>; + dma-names = "rx-tx"; }; saif1: saif@80046000 { compatible = "fsl,imx28-saif"; reg = <0x80046000 2000>; - interrupts = <58 81>; - fsl,saif-dma-channel = <5>; + interrupts = <58>; + dmas = <&dma_apbx 5>; + dma-names = "rx-tx"; fsl,saif-master = <&saif0>; }; diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index b41fffc..b16abbb 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -49,24 +49,8 @@ static const struct snd_pcm_hardware snd_mxs_hardware = { .fifo_size = 32, }; -static bool filter(struct dma_chan *chan, void *param) -{ - struct mxs_pcm_dma_params *dma_params = param; - - if (!mxs_dma_is_apbx(chan)) - return false; - - if (chan->chan_id != dma_params->chan_num) - return false; - - chan->private = &dma_params->dma_data; - - return true; -} - static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { .pcm_hardware = &snd_mxs_hardware, - .compat_filter_fn = filter, .prealloc_buffer_size = 64 * 1024, }; @@ -74,8 +58,6 @@ int mxs_pcm_platform_register(struct device *dev) { return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_COMPAT | SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 3aa918f..bc685b6 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -19,13 +19,6 @@ #ifndef _MXS_PCM_H #define _MXS_PCM_H -#include <linux/fsl/mxs-dma.h> - -struct mxs_pcm_dma_params { - struct mxs_dma_data dma_data; - int chan_num; -}; - int mxs_pcm_platform_register(struct device *dev); void mxs_pcm_platform_unregister(struct device *dev); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 71a972f..49d8700 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -26,7 +26,6 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/time.h> -#include <linux/fsl/mxs-dma.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -604,8 +603,6 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai) struct mxs_saif *saif = dev_get_drvdata(dai->dev); snd_soc_dai_set_drvdata(dai, saif); - dai->playback_dma_data = &saif->dma_param; - dai->capture_dma_data = &saif->dma_param; return 0; } @@ -664,7 +661,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct resource *iores, *dmares; + struct resource *iores; struct mxs_saif *saif; int ret = 0; struct device_node *master; @@ -719,22 +716,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmares) { - /* - * TODO: This is a temporary solution and should be changed - * to use generic DMA binding later when the helplers get in. - */ - ret = of_property_read_u32(np, "fsl,saif-dma-channel", - &saif->dma_param.chan_num); - if (ret) { - dev_err(&pdev->dev, "failed to get dma channel\n"); - return ret; - } - } else { - saif->dma_param.chan_num = dmares->start; - } - saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { ret = saif->irq; @@ -751,14 +732,6 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->dma_param.dma_data.chan_irq = platform_get_irq(pdev, 1); - if (saif->dma_param.dma_data.chan_irq < 0) { - ret = saif->dma_param.dma_data.chan_irq; - dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", - ret); - return ret; - } - platform_set_drvdata(pdev, saif); ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 3cb342e..53eaa4b 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -117,7 +117,6 @@ struct mxs_saif { unsigned int mclk_in_use; void __iomem *base; int irq; - struct mxs_pcm_dma_params dma_param; unsigned int id; unsigned int master_id; unsigned int cur_rate; -- 1.7.9.5

2 1

[alsa-devel] ASOC bug? maybe no need to set bias_level to PREPARE.
by Zhang, Ning A 13 May '13

13 May '13

When I try to fix wm1811 standby issue for our customer, I find a strange behavior in ASOC core. here is my test code: snd_soc_dapm_disable_pin(&dapm,"MICBIAS2"); snd_soc_dapm_disable_pin(&dapm,"CLK_SYS"); snd_soc_dapm_sync(&dapm); and add debug info in dapm_pre_sequence_async() and we will find these debug info printed: d->bias_level = 1 d->target_bias_level = 0 in snd_soc_dapm_set_bias_level from the debug log, when STANDBY -> OFF transition, it set bias_level to PREPARE. that's wrong logic. And more there is comment in source code: /* Prepare for a STADDBY->ON or ON->STANDBY transition */ But, the code isn't the same as the comment. ---- >From 58d8994bdc543909e9904b8df0787afc36684283 Mon Sep 17 00:00:00 2001 From: ning.a.zhang <ning.a.zhang(a)intel.com> Date: Fri, 10 May 2013 09:44:06 +0700 Subject: [PATCH] Asoc dapm behavior fix in soc_dapm.c, there is a illogic behavior, when bias_level from STANDBY -> OFF transition. source does not follow its comment. this patch is based on the comment Signed-off-by: ning.a.zhang <ning.a.zhang(a)intel.com> --- sound/soc/soc-dapm.c | 14 ++++++++++++-- 1 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d6d9ba2..834ab4c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1486,8 +1486,9 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) "ASoC: Failed to turn on bias: %d\n", ret); } - /* Prepare for a STADDBY->ON or ON->STANDBY transition */ - if (d->bias_level != d->target_bias_level) { + /* If we're on and we're not supposed to be go into PREPARE */ + if (d->bias_level == SND_SOC_BIAS_ON && + d->target_bias_level != SND_SOC_BIAS_ON) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); if (ret != 0) dev_err(d->dev, @@ -1525,6 +1526,15 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) pm_runtime_put(d->dev); } + if (d->bias_level == SND_SOC_BIAS_STANDBY && + (d->target_bias_level == SND_SOC_BIAS_PREPARE || + d->target_bias_level == SND_SOC_BIAS_ON)) { + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); + if (ret != 0) + dev_err(d->dev, "ASoC: Failed to apply prepare bias: %d\n", + ret); + } + /* If we just powered up then move to active bias */ if (d->bias_level == SND_SOC_BIAS_PREPARE && d->target_bias_level == SND_SOC_BIAS_ON) { -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Support EFS mode for FLL
by Mark Brown 13 May '13

13 May '13

Later WM8994 devices support an enhanced accuracy FLL divisor mode called EFS which allows more precise selection of fractional source to output ratios. Support this on relevant devices. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 49 +++++++++++++++++++++++++++++++++------------ 1 file changed, 36 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1eb152c..9f32dd8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/gcd.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> @@ -2005,15 +2006,16 @@ struct fll_div { u16 outdiv; u16 n; u16 k; + u16 lambda; u16 clk_ref_div; u16 fll_fratio; }; -static int wm8994_get_fll_config(struct fll_div *fll, +static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll, int freq_in, int freq_out) { u64 Kpart; - unsigned int K, Ndiv, Nmod; + unsigned int K, Ndiv, Nmod, gcd_fll; pr_debug("FLL input=%dHz, output=%dHz\n", freq_in, freq_out); @@ -2062,20 +2064,30 @@ static int wm8994_get_fll_config(struct fll_div *fll, Nmod = freq_out % freq_in; pr_debug("Nmod=%d\n", Nmod); - /* Calculate fractional part - scale up so we can round. */ - Kpart = FIXED_FLL_SIZE * (long long)Nmod; + switch (control->type) { + case WM8994: + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; - do_div(Kpart, freq_in); + do_div(Kpart, freq_in); - K = Kpart & 0xFFFFFFFF; + K = Kpart & 0xFFFFFFFF; - if ((K % 10) >= 5) - K += 5; + if ((K % 10) >= 5) + K += 5; - /* Move down to proper range now rounding is done */ - fll->k = K / 10; + /* Move down to proper range now rounding is done */ + fll->k = K / 10; - pr_debug("N=%x K=%x\n", fll->n, fll->k); + pr_debug("N=%x K=%x\n", fll->n, fll->k); + + default: + gcd_fll = gcd(freq_out, freq_in); + + fll->k = (freq_out - (freq_in * fll->n)) / gcd_fll; + fll->lambda = freq_in / gcd_fll; + + } return 0; } @@ -2139,9 +2151,9 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, * analysis bugs spewing warnings. */ if (freq_out) - ret = wm8994_get_fll_config(&fll, freq_in, freq_out); + ret = wm8994_get_fll_config(control, &fll, freq_in, freq_out); else - ret = wm8994_get_fll_config(&fll, wm8994->fll[id].in, + ret = wm8994_get_fll_config(control, &fll, wm8994->fll[id].in, wm8994->fll[id].out); if (ret < 0) return ret; @@ -2186,6 +2198,17 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, WM8994_FLL1_N_MASK, fll.n << WM8994_FLL1_N_SHIFT); + if (fll.lambda) { + snd_soc_update_bits(codec, WM8958_FLL1_EFS_1 + reg_offset, + WM8958_FLL1_LAMBDA_MASK, + fll.lambda); + snd_soc_update_bits(codec, WM8958_FLL1_EFS_2 + reg_offset, + WM8958_FLL1_EFS_ENA, WM8958_FLL1_EFS_ENA); + } else { + snd_soc_update_bits(codec, WM8958_FLL1_EFS_2 + reg_offset, + WM8958_FLL1_EFS_ENA, 0); + } + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, WM8994_FLL1_FRC_NCO | WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: mxs: move to use generic DMA helper
by Shawn Guo 13 May '13

13 May '13

With mxs-dma converted to generic DMA bindings, let's move mxs-pcm to use it by removing flages SND_DMAENGINE_PCM_FLAG_NO_DT and SND_DMAENGINE_PCM_FLAG_COMPAT. As the result, those mxs custom dma params code can be removed now. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- .../devicetree/bindings/sound/mxs-saif.txt | 17 ++++++++---- sound/soc/mxs/mxs-pcm.c | 18 ------------ sound/soc/mxs/mxs-pcm.h | 7 ----- sound/soc/mxs/mxs-saif.c | 29 +------------------- sound/soc/mxs/mxs-saif.h | 1 - 5 files changed, 12 insertions(+), 60 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt index c37ba61..7ba07a1 100644 --- a/Documentation/devicetree/bindings/sound/mxs-saif.txt +++ b/Documentation/devicetree/bindings/sound/mxs-saif.txt @@ -3,8 +3,11 @@ Required properties: - compatible: Should be "fsl,<chip>-saif" - reg: Should contain registers location and length -- interrupts: Should contain ERROR and DMA interrupts -- fsl,saif-dma-channel: APBX DMA channel for the SAIF +- interrupts: Should contain ERROR interrupt number +- dmas: DMA specifier, consisting of a phandle to DMA controller node + and SAIF DMA channel ID. + Refer to dma.txt and fsl-mxs-dma.txt for details. +- dma-names: Must be "rx-tx". Optional properties: - fsl,saif-master: phandle to the master SAIF. It's only required for @@ -23,14 +26,16 @@ aliases { saif0: saif@80042000 { compatible = "fsl,imx28-saif"; reg = <0x80042000 2000>; - interrupts = <59 80>; - fsl,saif-dma-channel = <4>; + interrupts = <59>; + dmas = <&dma_apbx 4>; + dma-names = "rx-tx"; }; saif1: saif@80046000 { compatible = "fsl,imx28-saif"; reg = <0x80046000 2000>; - interrupts = <58 81>; - fsl,saif-dma-channel = <5>; + interrupts = <58>; + dmas = <&dma_apbx 5>; + dma-names = "rx-tx"; fsl,saif-master = <&saif0>; }; diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index b41fffc..b16abbb 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -49,24 +49,8 @@ static const struct snd_pcm_hardware snd_mxs_hardware = { .fifo_size = 32, }; -static bool filter(struct dma_chan *chan, void *param) -{ - struct mxs_pcm_dma_params *dma_params = param; - - if (!mxs_dma_is_apbx(chan)) - return false; - - if (chan->chan_id != dma_params->chan_num) - return false; - - chan->private = &dma_params->dma_data; - - return true; -} - static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { .pcm_hardware = &snd_mxs_hardware, - .compat_filter_fn = filter, .prealloc_buffer_size = 64 * 1024, }; @@ -74,8 +58,6 @@ int mxs_pcm_platform_register(struct device *dev) { return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_COMPAT | SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 3aa918f..bc685b6 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -19,13 +19,6 @@ #ifndef _MXS_PCM_H #define _MXS_PCM_H -#include <linux/fsl/mxs-dma.h> - -struct mxs_pcm_dma_params { - struct mxs_dma_data dma_data; - int chan_num; -}; - int mxs_pcm_platform_register(struct device *dev); void mxs_pcm_platform_unregister(struct device *dev); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index d31dc52..d8c9a19 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -26,7 +26,6 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/time.h> -#include <linux/fsl/mxs-dma.h> #include <linux/pinctrl/consumer.h> #include <sound/core.h> #include <sound/pcm.h> @@ -605,8 +604,6 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai) struct mxs_saif *saif = dev_get_drvdata(dai->dev); snd_soc_dai_set_drvdata(dai, saif); - dai->playback_dma_data = &saif->dma_param; - dai->capture_dma_data = &saif->dma_param; return 0; } @@ -665,7 +662,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct resource *iores, *dmares; + struct resource *iores; struct mxs_saif *saif; struct pinctrl *pinctrl; int ret = 0; @@ -727,22 +724,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmares) { - /* - * TODO: This is a temporary solution and should be changed - * to use generic DMA binding later when the helplers get in. - */ - ret = of_property_read_u32(np, "fsl,saif-dma-channel", - &saif->dma_param.chan_num); - if (ret) { - dev_err(&pdev->dev, "failed to get dma channel\n"); - return ret; - } - } else { - saif->dma_param.chan_num = dmares->start; - } - saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { ret = saif->irq; @@ -759,14 +740,6 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->dma_param.dma_data.chan_irq = platform_get_irq(pdev, 1); - if (saif->dma_param.dma_data.chan_irq < 0) { - ret = saif->dma_param.dma_data.chan_irq; - dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", - ret); - return ret; - } - platform_set_drvdata(pdev, saif); ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 3cb342e..53eaa4b 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -117,7 +117,6 @@ struct mxs_saif { unsigned int mclk_in_use; void __iomem *base; int irq; - struct mxs_pcm_dma_params dma_param; unsigned int id; unsigned int master_id; unsigned int cur_rate; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: jz4740-i2s: Use clk_prepare_enable/clk_disable_unprepare
by Lars-Peter Clausen 12 May '13

12 May '13

In preparation to switching the jz4740 clk driver to the common clk framework update the clk enable/disable calls to clk_prepare_enable/clk_disable_unprepare. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/jz4740-i2s.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 9a12644..12c21bd 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -118,7 +118,7 @@ static int jz4740_i2s_startup(struct snd_pcm_substream *substream, ctrl |= JZ_AIC_CTRL_FLUSH; jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); - clk_enable(i2s->clk_i2s); + clk_prepare_enable(i2s->clk_i2s); conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); conf |= JZ_AIC_CONF_ENABLE; @@ -140,7 +140,7 @@ static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream, conf &= ~JZ_AIC_CONF_ENABLE; jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); - clk_disable(i2s->clk_i2s); + clk_disable_unprepare(i2s->clk_i2s); } static int jz4740_i2s_trigger(struct snd_pcm_substream *substream, int cmd, @@ -314,10 +314,10 @@ static int jz4740_i2s_suspend(struct snd_soc_dai *dai) conf &= ~JZ_AIC_CONF_ENABLE; jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); - clk_disable(i2s->clk_i2s); + clk_disable_unprepare(i2s->clk_i2s); } - clk_disable(i2s->clk_aic); + clk_disable_unprepare(i2s->clk_aic); return 0; } @@ -327,10 +327,10 @@ static int jz4740_i2s_resume(struct snd_soc_dai *dai) struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); uint32_t conf; - clk_enable(i2s->clk_aic); + clk_prepare_enable(i2s->clk_aic); if (dai->active) { - clk_enable(i2s->clk_i2s); + clk_prepare_enable(i2s->clk_i2s); conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); conf |= JZ_AIC_CONF_ENABLE; @@ -368,7 +368,7 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); uint32_t conf; - clk_enable(i2s->clk_aic); + clk_prepare_enable(i2s->clk_aic); jz4740_i2c_init_pcm_config(i2s); @@ -388,7 +388,7 @@ static int jz4740_i2s_dai_remove(struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); - clk_disable(i2s->clk_aic); + clk_disable_unprepare(i2s->clk_aic); return 0; } -- 1.8.2.1

2 1

[alsa-devel] UCM profiles taken from Ubuntu
by Marcin Juszkiewicz 11 May '13

11 May '13

When I bought Samsung ARM Chromebook few months ago I had no idea about UCM profiles and burnt speakers (left is dead, right is resting). Then I learnt, took ones from ChromeOS and added them into Ubuntu so other users will be a bit more safe. Later I had a discussion with some Debian, Ubuntu, Fedora developers about how to solve such problems and how to keep UCM profiles shared between distributions. Liam Girdwood pointed me to not used git tree at ALSA Project server and finally I reminded myself about it. The idea is to collect UCM profiles, keep them in one place and create alsa-ucm package in each distribution.

6 11

[alsa-devel] [PATCH v5 1/2] ASoC: sgtl5000: Read SGTL5000_CHIP_ID in i2c_probe()
by Fabio Estevam 10 May '13

10 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> The usual place for reading chip ID is inside i2c_probe, so move it there and also convert it to regmap. sgtl5000_enable_regulators() needs to read the chip revision, so keep the revision check there. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v4: - Just introduced in this version. sound/soc/codecs/sgtl5000.c | 39 ++++++++++++++++++++++++--------------- 1 file changed, 24 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b443..af0c8aa 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1275,7 +1275,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { - u16 reg; + int reg; int ret; int rev; int i; @@ -1303,23 +1303,17 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) /* wait for all power rails bring up */ udelay(10); - /* read chip information */ - reg = snd_soc_read(codec, SGTL5000_CHIP_ID); - if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != - SGTL5000_PARTID_PART_ID) { - dev_err(codec->dev, - "Device with ID register %x is not a sgtl5000\n", reg); - ret = -ENODEV; - goto err_regulator_disable; - } - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - dev_info(codec->dev, "sgtl5000 revision 0x%x\n", rev); - /* * workaround for revision 0x11 and later, * roll back to use internal LDO */ + + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); + if (ret) + goto err_regulator_disable; + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + if (external_vddd && rev >= 0x11) { /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), @@ -1478,7 +1472,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct sgtl5000_priv *sgtl5000; - int ret; + int ret, reg, rev; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1492,6 +1486,21 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } + /* read chip information */ + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); + if (ret) + return ret; + + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(&client->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + return -ENODEV; + } + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); + i2c_set_clientdata(client, sgtl5000); ret = snd_soc_register_codec(&client->dev, -- 1.7.9.5

4 16

[alsa-devel] [PATCH] snd-es1968: Add radio support for MediaForte M56VAP
by Ondrej Zary 10 May '13

10 May '13

Add support for TEA5757 tuner on MediaForte M56VAP sound+modem+radio card. The GPIO connection type is automatically detected (like snd-fm801 driver). Also add a safety subsystem vendor check to skip radio detection if vendor differs from ESS (so we don't touch GPIOs on laptops). Tested with SF64-PCE2 and M56VAP cards. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/es1968.c | 73 +++++++++++++++++++++++++++++++++++---------------- 1 files changed, 50 insertions(+), 23 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a1f32b5..db08641 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -564,6 +564,7 @@ struct es1968 { #ifdef CONFIG_SND_ES1968_RADIO struct v4l2_device v4l2_dev; struct snd_tea575x tea; + unsigned int tea575x_tuner; #endif }; @@ -2557,37 +2558,47 @@ static int snd_es1968_input_register(struct es1968 *chip) bits 1=unmask write to given bit */ #define IO_DIR 8 /* direction register offset from GPIO_DATA bits 0/1=read/write direction */ -/* mask bits for GPIO lines */ -#define STR_DATA 0x0040 /* GPIO6 */ -#define STR_CLK 0x0080 /* GPIO7 */ -#define STR_WREN 0x0100 /* GPIO8 */ -#define STR_MOST 0x0200 /* GPIO9 */ + +/* GPIO to TEA575x maps */ +struct snd_es1968_tea575x_gpio { + u8 data, clk, wren, most; + char *name; +}; + +static struct snd_es1968_tea575x_gpio snd_es1968_tea575x_gpios[] = { + { .data = 6, .clk = 7, .wren = 8, .most = 9, .name = "SF64-PCE2" }, + { .data = 7, .clk = 8, .wren = 6, .most = 10, .name = "M56VAP" }, +}; + +#define get_tea575x_gpio(chip) \ + (&snd_es1968_tea575x_gpios[(chip)->tea575x_tuner]) + static void snd_es1968_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); u16 val = 0; - val |= (pins & TEA575X_DATA) ? STR_DATA : 0; - val |= (pins & TEA575X_CLK) ? STR_CLK : 0; - val |= (pins & TEA575X_WREN) ? STR_WREN : 0; + val |= (pins & TEA575X_DATA) ? (1 << gpio.data) : 0; + val |= (pins & TEA575X_CLK) ? (1 << gpio.clk) : 0; + val |= (pins & TEA575X_WREN) ? (1 << gpio.wren) : 0; - outw(val, io); + outw(val, chip->io_port + GPIO_DATA); } static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; - u16 val = inw(io); - u8 ret; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); + u16 val = inw(chip->io_port + GPIO_DATA); + u8 ret = 0; - ret = 0; - if (val & STR_DATA) + if (val & (1 << gpio.data)) ret |= TEA575X_DATA; - if (val & STR_MOST) + if (val & (1 << gpio.most)) ret |= TEA575X_MOST; + return ret; } @@ -2596,13 +2607,18 @@ static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool outpu struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; u16 odir = inw(io + IO_DIR); + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); if (output) { - outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); - outw(odir | STR_DATA | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren)), + io + IO_MASK); + outw(odir | (1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren), + io + IO_DIR); } else { - outw(~(STR_CLK | STR_WREN | STR_DATA | STR_MOST), io + IO_MASK); - outw((odir & ~(STR_DATA | STR_MOST)) | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.clk) | (1 << gpio.wren) | (1 << gpio.data) | (1 << gpio.most)), + io + IO_MASK); + outw((odir & ~((1 << gpio.data) | (1 << gpio.most))) + | (1 << gpio.clk) | (1 << gpio.wren), io + IO_DIR); } } @@ -2772,6 +2788,9 @@ static int snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_ES1968_RADIO + /* don't play with GPIOs on laptops */ + if (chip->pci->subsystem_vendor != 0x125d) + goto no_radio; err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); if (err < 0) { snd_es1968_free(chip); @@ -2781,10 +2800,18 @@ static int snd_es1968_create(struct snd_card *card, chip->tea.private_data = chip; chip->tea.radio_nr = radio_nr; chip->tea.ops = &snd_es1968_tea_ops; - strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); - if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) - printk(KERN_INFO "es1968: detected TEA575x radio\n"); + for (i = 0; i < ARRAY_SIZE(snd_es1968_tea575x_gpios); i++) { + chip->tea575x_tuner = i; + if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) { + snd_printk(KERN_INFO "es1968: detected TEA575x radio type %s\n", + get_tea575x_gpio(chip)->name); + strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, + sizeof(chip->tea.card)); + break; + } + } +no_radio: #endif *chip_ret = chip; -- Ondrej Zary

1 0

Re: [alsa-devel] [PATCH v5 2/2] ASoC: sgtl5000: Fix driver probe after reset
by Fabio Estevam 10 May '13

10 May '13

Hi Matt, On Fri, May 10, 2013 at 2:14 PM, Matt Sealey <matt(a)genesi-usa.com> wrote: > Hi Fabio! > > Is there any way to make that failure a little less of a "failure" or > even a warning, since the message below; > >> sgtl5000 0-000a: Using internal LDO instead of VDDD > > Pretty much covers that VDDD was not supplied. I would consider that > you would want to say "VDDD wasn't supplied" in the sense that it is > optional and not give in pdata or dt, and then remark that the LDO was > actually configured.. I don't like it when my kernels boot up and spit > out high numbered errno's for the sake of a completely optional entry > in the design vs. missing and still optional entry in the device tree. Yes, I plan to clean the sgtl5000 log messages. I agree that it is currently too noisy. Will do it after Mark applies this series. Regards, Fabio Estevam

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Re: [alsa-devel] [RFC 42/42] sound/soc/kirkwood: don't check resource with devm_ioremap_resource
by Mark Brown 10 May '13

10 May '13

On Fri, May 10, 2013 at 10:17:27AM +0200, Wolfram Sang wrote: > devm_ioremap_resource does sanity checks on the given resource. No need to > duplicate this in the driver. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

1 0

Re: [alsa-devel] [RFC 41/42] sound/soc/fsl: don't check resource with devm_ioremap_resource
by Mark Brown 10 May '13

10 May '13

On Fri, May 10, 2013 at 10:17:26AM +0200, Wolfram Sang wrote: > devm_ioremap_resource does sanity checks on the given resource. No need to > duplicate this in the driver. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] [PATCH v3] ASoC: sgtl5000: Fix driver probe after reset
by Fabio Estevam 10 May '13

10 May '13

sgtl5000 codec does not have a reset line, nor a reset command in software. After a 'reboot' command in Linux or after pressing the system's reset button the sgtl5000 driver fails to probe: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) As the sgtl5000 codec did not go through a real reset, we cannot rely on the sgtl5000_reg_defaults table, since these are only valid after a clean power-on reset. Fix this issue by explicitly reading all the sgtl5000 registers and filling sgtl5000_reg_defaults with such values. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v2: - Do not use reg_defaults_raw as it is not the correct purpose - Manually build sgtl5000_reg_default - Improve commitlog Changes since v1: - Remove sgtl5000_reg_defaults array - Do not use num_reg_defaults_raw sound/soc/codecs/sgtl5000.c | 58 ++++++++++++++++++++++++------------------- 1 file changed, 33 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b443..311dfb5 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -35,31 +35,7 @@ #define SGTL5000_MAX_REG_OFFSET 0x013A /* default value of sgtl5000 registers */ -static const struct reg_default sgtl5000_reg_defaults[] = { - { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, - { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, - { SGTL5000_CHIP_SSS_CTRL, 0x0008 }, - { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, - { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, - { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, - { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, - { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, - { SGTL5000_CHIP_ANA_POWER, 0x7060 }, - { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, - { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, - { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, - { SGTL5000_DAP_SURROUND, 0x0040 }, - { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND2, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, - { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, - { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, - { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, - { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -}; +static struct reg_default sgtl5000_reg_defaults[SGTL5000_MAX_REG_OFFSET + 1]; /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { @@ -1355,6 +1331,33 @@ err_regulator_free: } +/* + * Read all the sgtl5000 registers to fill the cache, as + * we cannot rely on a pre-defined table containing the + * power-on reset values of the registers as done in most + * of the other codec drivers. + * + * We follow this approach here because sgtl5000 does not have + * a reset line, nor a reset command in software, and this way + * we can guarantee that we always have sane register values + * stored in the reg_defaults table after a reset + */ +static int sgtl5000_fill_cache(struct snd_soc_codec *codec) +{ + int i, reg, ret; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + for (i = 0; i < SGTL5000_MAX_REG_OFFSET; i += 2) { + ret = regmap_read(sgtl5000->regmap, i, &reg); + if (ret < 0) + return ret; + sgtl5000_reg_defaults[i].reg = i; + sgtl5000_reg_defaults[i].def = reg; + } + + return 0; +} + static int sgtl5000_probe(struct snd_soc_codec *codec) { int ret; @@ -1377,6 +1380,11 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) if (ret) goto err; + /* Build the reg_defaults manually by reading the registers */ + ret = sgtl5000_fill_cache(codec); + if (ret) + goto err; + /* enable small pop, introduce 400ms delay in turning off */ snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, SGTL5000_SMALL_POP, -- 1.7.9.5

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Re: [alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix driver probe after reset
by Mark Brown 10 May '13

10 May '13

On Thu, May 09, 2013 at 12:08:38PM -0500, Matt Sealey wrote: > match a reset chip powered on for the first time. What we should > actually be doing is dumping these into the chip as the first thing we > do at probe, and then they DO become the reasonable defaults. Yes, this is eactly what I keep saying :/ > The fix is keep using cache/regmaps or whatever, but it definitely > would be an absolute requirement to reset the codec manually by > putting in all the right register values before using the cache, which You don't need to disable the cache at any point.

1 0

[alsa-devel] TLV question - ranges not in volume order
by David Henningsson 10 May '13

10 May '13

Hi, I recently came across a strange ASoC volume control. In this case, the values are not in volume order, i e, the control is declared something like below. I believe the numbers correctly match the hardware. However alsamixer seems not to handle this "reordering" correctly. My question is if this is something that should be fixed in alsamixer (or possibly alsa-lib?), or does one need to write some kind of mapping table in the kernel, just to make something come in volume order to userspace? Any TLV documentation on what TLVs are valid and invalid is also welcome :-) static const unsigned int tlv[] = { TLV_DB_RANGE_HEAD(8), 0, 0, TLV_DB_SCALE_ITEM(-100, 0, 1), 1, 1, TLV_DB_SCALE_ITEM(400, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(200, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(1000, 0, 0), 4, 4, TLV_DB_SCALE_ITEM(0, 0, 0), 5, 5, TLV_DB_SCALE_ITEM(800, 0, 0), 6, 6, TLV_DB_SCALE_ITEM(600, 0, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

4 6

Re: [alsa-devel] [RFC PATCH 1/3] drm/915: Add private api for power well usage
by Daniel Vetter 10 May '13

10 May '13

Hi all, Three things: 1. Any reason public mailing lists (intel-gfx, alsa-devel) aren't cc'ed? Afaics no secret stuff is being discussed here ... Please resend the patches with mailings lists cc'ed. 2. On a quick read of the hda driver stuff I don't think the power well enable/disable places are at the right spot: We force the power well on whenever the hda codec is used, but we must only do that when we want to output audio to external screens. And since that state transition can only really happen due to a modeset change on the gfx side I don't think we need any autosuspend delays either. The use-case I'm thinking of is media playback with just the panel enabled: In that case we can switch off the power well on the display side, and I don't think the power well is required for audio play back on the integrated speakerrs/headphone-jack either. 3. The moduel depencies seem to still not work out dynamically, at least I think we still have a hard chain between hda-intel and i915.ko (just with one thing in between now). Cheers, Daniel On Thu, May 9, 2013 at 9:23 AM, Wang, Xingchao <xingchao.wang(a)intel.com> wrote: > Hi All, > > This is a newer version patchset for power well feature implementation. > > I will send out a document later to describe the design, especially for David and Takashi as they could not see Bspec > so it maybe a bit confuse. Meanwhile Liam will send out a meeting request for discussion, welcome Daniel, Paulo, Jesse to join us. > > Basically, here's the latest working status on my side: > > I tested the power well feature on Haswell ULT board, there's only one eDP port used. > Without this patch and we enabled power well feature, gfx driver will shutdown power well, audio driver will crash. > Applied this patch, display audio driver will request power well on before initialize the card. In gfx side, it will enable power well. > > Also power_save feature in audio side should be enabled, I set "5" second to let codec enter suspend when free for 5s long. > Audio controller driver detects all codecs are suspended it will release power well and suspend too. Gfx driver will receive the request and shut down power well. > > If user trigger some audio operations(cat codec# info), audio controller driver will request power well again... > > If gfx side decided to disable power well now, while audio is in use, power well would be kept on. > > Generally audio drvier will request power well on need and release it immediately after suspend. Gfx should make decision at his side. > > The new introduced module "hda-i915" has dependency on i915, only built in when i915 enabled. So it's safe for call. > This module was based on David's original patch, which added pm_suspend/resume callback for hdmi codec. As the codecs and controller are both depending on powerwell, > and codec *must* already on power so I moved the power well request/release to controller side and changed the module name to "hda-i915". > David, would you like me to add you as author for the second patch? :) > > For non-Haswell platform, power well is no need atm. If the module is built in and gfx will reject power well request at its side, so it's safe too. > > thanks > --xingchao > > >> -----Original Message----- >> From: Wang Xingchao [mailto:xingchao.wang@linux.intel.com] >> Sent: Thursday, May 09, 2013 3:01 PM >> To: david.henningsson(a)canonical.com; Girdwood, Liam R; tiwai(a)suse.de; >> daniel(a)ffwll.ch >> Cc: Barnes, Jesse; Li, Jocelyn; Lin, Mengdong; Wang, Xingchao; Zanoni, Paulo R; >> Wang Xingchao >> Subject: [RFC PATCH 1/3] drm/915: Add private api for power well usage >> >> Haswell Display audio depends on power well in graphic side, it should request >> power well before use it and release power well after use. >> I915 will not shutdown power well if it detects audio is using. >> This patch protects display audio crash for Intel Haswell mobile >> C3 stepping board. >> >> Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> >> --- >> drivers/gpu/drm/i915/intel_pm.c | 127 >> ++++++++++++++++++++++++++++++++++++--- >> include/drm/audio_drm.h | 39 ++++++++++++ >> 2 files changed, 159 insertions(+), 7 deletions(-) create mode 100644 >> include/drm/audio_drm.h >> >> diff --git a/drivers/gpu/drm/i915/intel_pm.c b/drivers/gpu/drm/i915/intel_pm.c >> index 2557926..cba753e 100644 >> --- a/drivers/gpu/drm/i915/intel_pm.c >> +++ b/drivers/gpu/drm/i915/intel_pm.c >> @@ -4298,18 +4298,12 @@ bool intel_using_power_well(struct drm_device >> *dev) >> return true; >> } >> >> -void intel_set_power_well(struct drm_device *dev, bool enable) >> +void set_power_well(struct drm_device *dev, bool enable) >> { >> struct drm_i915_private *dev_priv = dev->dev_private; >> bool is_enabled, enable_requested; >> uint32_t tmp; >> >> - if (!HAS_POWER_WELL(dev)) >> - return; >> - >> - if (!i915_disable_power_well && !enable) >> - return; >> - >> tmp = I915_READ(HSW_PWR_WELL_DRIVER); >> is_enabled = tmp & HSW_PWR_WELL_STATE; >> enable_requested = tmp & HSW_PWR_WELL_ENABLE; @@ -4332,6 >> +4326,125 @@ void intel_set_power_well(struct drm_device *dev, bool >> enable) >> } >> } >> >> +/* Global drm_device for display audio drvier usage */ static struct >> +drm_device *power_well_device; >> +/* Lock protecting power well setting */ >> +DEFINE_SPINLOCK(powerwell_lock); static bool i915_power_well_using; >> + >> +struct power_well { >> + int user_cnt; >> + struct list_head power_well_list; >> +}; >> + >> +static struct power_well hsw_power = { >> + .user_cnt = 0, >> + .power_well_list = LIST_HEAD_INIT(hsw_power.power_well_list), >> +}; >> + >> +struct power_well_user { >> + char name[16]; >> + struct list_head list; >> +}; >> + >> +void i915_request_power_well(const char *name) { >> + struct power_well_user *user; >> + >> + DRM_DEBUG_KMS("request power well for %s\n", name); >> + spin_lock_irq(&powerwell_lock); >> + if (!power_well_device) >> + goto out; >> + >> + if (!IS_HASWELL(power_well_device)) >> + goto out; >> + >> + list_for_each_entry(user, &hsw_power.power_well_list, list) { >> + if (!strcmp(user->name, name)) { >> + goto out; >> + } >> + } >> + >> + user = kzalloc(sizeof(*user), GFP_KERNEL); >> + if (user == NULL) { >> + goto out; >> + } >> + strcpy(user->name, name); >> + >> + list_add(&user->list, &hsw_power.power_well_list); >> + hsw_power.user_cnt++; >> + >> + if (hsw_power.user_cnt == 1) { >> + set_power_well(power_well_device, true); >> + DRM_DEBUG_KMS("%s set power well on\n", user->name); >> + } >> +out: >> + spin_unlock_irq(&powerwell_lock); >> + return; >> +} >> +EXPORT_SYMBOL_GPL(i915_request_power_well); >> + >> +void i915_release_power_well(const char *name) { >> + struct power_well_user *user; >> + >> + DRM_DEBUG_KMS("release power well from %s\n", name); >> + spin_lock_irq(&powerwell_lock); >> + if (!power_well_device) >> + goto out; >> + >> + if (!IS_HASWELL(power_well_device)) >> + goto out; >> + >> + list_for_each_entry(user, &hsw_power.power_well_list, list) { >> + if (!strcmp(user->name, name)) { >> + list_del(&user->list); >> + hsw_power.user_cnt--; >> + DRM_DEBUG_KMS("Releaseing power well:%s, user_cnt:%d, >> i915 using:%d\n", >> + user->name, hsw_power.user_cnt, >> i915_power_well_using); >> + if (!hsw_power.user_cnt && !i915_power_well_using) >> + set_power_well(power_well_device, false); >> + goto out; >> + } >> + } >> + >> + DRM_DEBUG_KMS("power well %s doesnot exist\n", name); >> +out: >> + spin_unlock_irq(&powerwell_lock); >> + return; >> +} >> +EXPORT_SYMBOL_GPL(i915_release_power_well); >> + >> +/* TODO: Call this when i915 module unload */ void >> +remove_power_well(void) { >> + spin_lock_irq(&powerwell_lock); >> + i915_power_well_using = false; >> + power_well_device = NULL; >> + spin_unlock_irq(&powerwell_lock); >> +} >> + >> +void intel_set_power_well(struct drm_device *dev, bool enable) { >> + if (!HAS_POWER_WELL(dev)) >> + return; >> + >> + spin_lock_irq(&powerwell_lock); >> + power_well_device = dev; >> + i915_power_well_using = enable; >> + if (!enable && hsw_power.user_cnt) { >> + DRM_DEBUG_KMS("Display audio power well busy using now\n"); >> + goto out; >> + } >> + >> + if (!i915_disable_power_well && !enable) >> + goto out; >> + set_power_well(dev, enable); >> +out: >> + spin_unlock_irq(&powerwell_lock); >> + return; >> +} >> + >> /* >> * Starting with Haswell, we have a "Power Down Well" that can be turned off >> * when not needed anymore. We have 4 registers that can request the >> power well diff --git a/include/drm/audio_drm.h b/include/drm/audio_drm.h >> new file mode 100644 index 0000000..215295f >> --- /dev/null >> +++ b/include/drm/audio_drm.h >> @@ -0,0 +1,39 @@ >> +/************************************************************** >> ******** >> +**** >> + * >> + * Copyright 2013 Intel Inc. >> + * All Rights Reserved. >> + * >> + * Permission is hereby granted, free of charge, to any person >> +obtaining a >> + * copy of this software and associated documentation files (the >> + * "Software"), to deal in the Software without restriction, including >> + * without limitation the rights to use, copy, modify, merge, publish, >> + * distribute, sub license, and/or sell copies of the Software, and to >> + * permit persons to whom the Software is furnished to do so, subject >> +to >> + * the following conditions: >> + * >> + * The above copyright notice and this permission notice (including the >> + * next paragraph) shall be included in all copies or substantial >> +portions >> + * of the Software. >> + * >> + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, >> +EXPRESS OR >> + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF >> +MERCHANTABILITY, >> + * FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT. IN NO >> EVENT >> +SHALL >> + * THE COPYRIGHT HOLDERS, AUTHORS AND/OR ITS SUPPLIERS BE LIABLE >> FOR >> +ANY CLAIM, >> + * DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, >> TORT >> +OR >> + * OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE >> SOFTWARE >> +OR THE >> + * USE OR OTHER DEALINGS IN THE SOFTWARE. >> + * >> + * >> + >> +*************************************************************** >> ******** >> +***/ >> +/* >> + * Authors: >> + * Wang Xingchao <xingchao.wang(a)intel.com> */ >> + >> +#ifndef _AUDIO_DRM_H_ >> +#define _AUDIO_DRM_H_ >> + >> +extern void i915_request_power_well(const char *name); extern void >> +i915_release_power_well(const char *name); >> + >> +#endif /* _AUDIO_DRM_H_ */ >> -- >> 1.7.9.5 > -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

4 5

[alsa-devel] [PATCH v4] ASoC: sgtl5000: Fix driver probe after reset
by Fabio Estevam 10 May '13

10 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> After a 'reboot' command in Linux or after pressing the system's reset button the sgtl5000 driver fails to probe: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) sgtl5000 codec does not have a reset line, nor a reset command in software, so after a system reset the codec does not contain the default register values from sgtl5000_reg_defaults[] anymore, as these are only valid after a power-on-reset cycle. Fix this issue by explicitly reading all the reset register values from sgtl5000_reg_defaults[] and writing them back into sgtl5000 to ensure a sane state. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v3: - Read sgtl5000_reg_defaults and write these values into sgtl5000 Changes since v2: - Do not use reg_defaults_raw as it is not the correct purpose - Manually build sgtl5000_reg_default - Improve commitlog Changes since v1: - Remove sgtl5000_reg_defaults array - Do not use num_reg_defaults_raw sound/soc/codecs/sgtl5000.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b443..be23fce 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1474,6 +1474,31 @@ static const struct regmap_config sgtl5000_regmap = { .num_reg_defaults = ARRAY_SIZE(sgtl5000_reg_defaults), }; +/* + * Write all the default values from sgtl5000_reg_defaults[] array into the + * sgtl5000 registers, to make sure we always start with the sane registers + * values as stated in the datasheet. + * + * Since sgtl5000 does not have a reset line, nor a reset command in software, + * we follow this approach to guarantee we always start from the default values + * and avoid problems like, not being able to probe after an audio playback + * followed by a system reset or a 'reboot' command in Linux + */ +static int sgtl5000_fill_defaults(struct sgtl5000_priv *sgtl5000) +{ + int i, ret, val, index; + + for (i = 0; i < ARRAY_SIZE(sgtl5000_reg_defaults); i++) { + val = sgtl5000_reg_defaults[i].def; + index = sgtl5000_reg_defaults[i].reg; + ret = regmap_write(sgtl5000->regmap, index, val); + if (ret) + return ret; + } + + return 0; +} + static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -1494,6 +1519,11 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, sgtl5000); + /* Ensure sgtl5000 will start with sane register values */ + ret = sgtl5000_fill_defaults(sgtl5000); + if (ret) + return ret; + ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); return ret; -- 1.7.9.5

2 1

Re: [alsa-devel] BUG: USB audio discontinuities with 'UHCI: implement new semantics for URB_ISO_ASAP'
by Alan Stern 09 May '13

09 May '13

On Wed, 8 May 2013, Joe Rayhawk wrote: > For what it's worth, the following -stable patch > > "ALSA: USB: adjust for changed 3.8 USB API" > c75c5ab575af7db707689cdbb5a5c458e9a034bb > > fixes the discontinuous playback on period sizes between 139 and 192, > but the discontinuous playback on period sizes between 48 and 138 has > been replaced by rather worse kernel blocking and kernel errors: > > jrayhawk@richardiv:~$ sudo tail -F -n 0 /var/log/kern.log & > [1] 4490 > jrayhawk@richardiv:~$ time perl -e 'print pack "H*", "00FF" x 2048' | aplay --period-size=192 -r 48000 -f S16_LE -c2 -D hw:0,0 > Playing raw data 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo > > real 0m0.053s > user 0m0.008s > sys 0m0.000s > jrayhawk@richardiv:~$ time perl -e 'print pack "H*", "00FF" x 2048' | aplay --period-size=48 -r 48000 -f S16_LE -c2 -D hw:0,0 > Playing raw data 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo > May 8 17:01:56 richardiv kernel: [ 1680.923474] cannot submit urb (err = -18) > May 8 17:01:56 richardiv kernel: [ 1680.924472] cannot submit urb (err = -18) > > real 0m10.023s > user 0m0.008s > sys 0m0.004s I don't know if you want to investigate this in any detail. If you do, please post a usbmon trace of a failed playback. Alan Stern -- 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

1 0

[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix driver probe after reset
by Fabio Estevam 09 May '13

09 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 codec does not have a reset line, nor a reset command in software. After a 'reboot' command in Linux or after pressing the system's reset button the sgtl5000 driver fails to probe: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) As the sgtl5000 codec did not go through a real reset, we cannot rely on the sgtl5000_reg_defaults array, since these are only valid after a clean power-on reset. To fix this issue, use 'num_reg_defaults_raw' which tells the regcache driver to actually read the codec registers and then fill the cache with these values. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Remove sgtl5000_reg_defaults array - Do not use num_reg_defaults_raw sound/soc/codecs/sgtl5000.c | 30 +----------------------------- 1 file changed, 1 insertion(+), 29 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b443..fa342f7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -34,33 +34,6 @@ #define SGTL5000_DAP_REG_OFFSET 0x0100 #define SGTL5000_MAX_REG_OFFSET 0x013A -/* default value of sgtl5000 registers */ -static const struct reg_default sgtl5000_reg_defaults[] = { - { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, - { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, - { SGTL5000_CHIP_SSS_CTRL, 0x0008 }, - { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, - { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, - { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, - { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, - { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, - { SGTL5000_CHIP_ANA_POWER, 0x7060 }, - { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, - { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, - { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, - { SGTL5000_DAP_SURROUND, 0x0040 }, - { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND2, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, - { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, - { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, - { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, - { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, - { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -}; - /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { VDDA, @@ -1470,8 +1443,7 @@ static const struct regmap_config sgtl5000_regmap = { .readable_reg = sgtl5000_readable, .cache_type = REGCACHE_RBTREE, - .reg_defaults = sgtl5000_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(sgtl5000_reg_defaults), + .num_reg_defaults_raw = SGTL5000_MAX_REG_OFFSET, }; static int sgtl5000_i2c_probe(struct i2c_client *client, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix driver probe after reset
by Fabio Estevam 09 May '13

09 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 codec does not have a reset line, nor a reset command in software. After a 'reboot' command in Linux or after pressing the system's reset button the sgtl5000 driver fails to probe: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) As the sgtl5000 codec did not go through a real reset, we cannot rely on the reg_defaults array, since these are only valid after a clean power-on reset. To fix this issue, use 'reg_defaults_raw' which tells the regcache driver to actually read the codec registers and then fill the cache. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b443..c59fae8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1470,8 +1470,8 @@ static const struct regmap_config sgtl5000_regmap = { .readable_reg = sgtl5000_readable, .cache_type = REGCACHE_RBTREE, - .reg_defaults = sgtl5000_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(sgtl5000_reg_defaults), + .reg_defaults_raw = sgtl5000_reg_defaults, + .num_reg_defaults_raw = ARRAY_SIZE(sgtl5000_reg_defaults), }; static int sgtl5000_i2c_probe(struct i2c_client *client, -- 1.7.9.5

3 2

Re: [alsa-devel] [RFC] ASoC: sgtl5000: Remove cache support
by Mark Brown 08 May '13

08 May '13

On Tue, May 07, 2013 at 03:06:39PM -0500, Matt Sealey wrote: > On Mon, May 6, 2013 at 5:06 PM, Mark Brown <broonie(a)kernel.org> wrote: > Just for reference the supplies on the Efika MX are VDDD 1.2V, VDDDA > 1.65V and VDDIO 2.775V respectively. For the sake of a couple 0.1uF > capacitors you're not really saving much money in the lowest cost > configuration, and you're doubling to tripling power consumption. If > 15mW makes a difference to you (especially if you're supplying the > above from rails connected to other devices and you're concerned about > current limiting etc.) it makes sense to go for the lower power one. Well. some applications just don't care about power consumption. > It is hard to imagine an i.MX board without 1.2V, 1.6V and 3.3V from a > PMIC to supply the codec so letting VDDD float and using the internal > LDO seems like an exercise in selling the codec to people who aren't > using other Freescale parts.. Even if there's physically a supply at the appropriate voltage people can have concerns about running noisy digital stuff near their analogue, the on-chip regulators aren't purely about component cost. Not sure that this is a consideration here but... > I think most of the weird problems is the LDO configuration itself - > Linux assumes that it's not set up, when if VDDD isn't there, it is > already. Weird sequencing of supplies by boards (either not bringing > any supplies down or not bringing them down correctly in a reasonable > order) on warm reboot and enabling it *regardless* at revision 0x11 or > above which is just turning (on our boards) a 1.2V supply into a > better filtered 1.2V supply and making the chip run a little hotter. Is it actually having a postive impact on performance? > I am curious about the version check more than anything. Either the > internal LDO is required to filter poor supplies on some boards (which > ones?) or revisions equal or above 0x11 had some DIFFERENT digital > supply requirement - I seem to remember the reference manual stating > 1.2V was a suitable input supply, but the minimum is 1.1V in the > latest RM. Maybe if you run it at 1.2V there's some problem. It's not > documented why that decision would be made.. no errata.. That sounds like tolerances doesn't it? Spec of 1.2V. > >> So maybe we don't disable the cache, or it gets ported to regmap, but > >> the order should be enable regulators in a non-cached way then fill > >> the cache AFTERWARDS - the code at ToT mainline doesn't fix the cache > > I'm sorry, this makes no sense to me. If we know the physical defaults > > for the device and we can get the device into that state then we should > > have no need to read from the cache. If we don't know what they are > > then they shouldn't be listed. > In the case VDDD isn't supplied when the other two supplies are > brought up, then the LDO is enabled, so if the register controlling > the LDO configuration is in the cache.. that might explain a lot... I still don't understand this at all. Is the power on value for the register variable? > > I think the point that you lost me was your assumption that the > > regulator enable needs to be done without the cache. What makes you > > believe that this is the case? > Because that's what Freescale did in their code as per their comments > - if only because they needed to do this kind of setup before the > cache/defaults were populated. I assume they populate the cache from > the chip at probe because the defaults change per revision and the > presence of VDDD or not may change some of the registers in the cache. We just have no idea why they're doing what they're doing; this is all guesswork. I've not seen anything that indicates that anyone who's sent any code has any understanding of what they're changing. > It may also be on boards like i.MX6 Sabrelite that the "defaults" are > not the "defaults" over a warm reboot, since the chip never powers off > (just "optionally" loses clock depending on whether CLKO is > reconfigured or not at some point) and probably retains everything you > ever set. This is why most drivers do a hardware reset of the device on gaining control of it, in order to ensure that the hardware is in a known state. I see that the driver doesn't do that except through the regulators. I see this driver doesn't do that which isn't terribly clever, though it's not clear that the chip has suitable support. > We want to do two possible things... > * At probe, dump some known good defaults into the chip and use those as a cache > * At probe, read out all the registers assuming they're already known > good defaults into a cache Clearly the first option is more sane since otherwise you've no idea if the chip is configured sensibly. > The intent being in the first case to make for damn sure the codec is > in a well known state, and in the second case JUST to reduce i2c > writes further down the line. Freescale seem to like the second one, This isn't the sole function of the register cache, it's also used to implement most of suspend and resume and to allow the device to be powered off when idle but still present controls. > Mainline is doing the first set, although it's definitely possible > those defaults are not the true defaults if they were derived from a > board like the Sabrelite where a warm reboot might retain all the > state from previous boots.. the precedence of that list of defaults is > basically unknown.. and I recall Freescale updating them in the BSP a > lot. I'd *really* hope that the defaults come from the chip datasheet. If they don't that's extremely disappointing. > So the problem here might be some quirky board designs, a random > implementation of an erratum workaround with no documentation, a bunch My impression based on what I'm seeing here, including things like your comments about lots of register default changes, is that people working on the driver haven't had much understanding of what they're doing and have got confused as a result. Sounds like a good first step is getting the register defaults sane...

3 5

[alsa-devel] [PATCH v2] ASoC: wm_adsp: Expose coefficient blocks as ALSA binary controls
by Dimitris Papastamos 08 May '13

08 May '13

Add initial support for runtime tuning for the ADSP cores. This is achieved by exposing the coefficient configuration blocks as ALSA binary controls. The current code assumes that no controls on the DSP are volatile. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 458 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 3 + 2 files changed, 454 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9af1bdd..a9b9ebb 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -21,6 +21,7 @@ #include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> +#include <linux/workqueue.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -206,6 +207,36 @@ static struct { { .file = "rx-anc" }, }; +struct wm_coeff_ctl_ops { + int (*xget)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xput)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xinfo)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +}; + +struct wm_coeff { + struct device *dev; + struct list_head ctl_list; + struct regmap *regmap; +}; + +struct wm_coeff_ctl { + const char *name; + struct snd_card *card; + struct wm_adsp_alg_region region; + struct wm_coeff_ctl_ops ops; + struct wm_adsp *adsp; + void *private; + unsigned int enabled:1; + struct list_head list; + void *cache; + size_t len; + unsigned int dirty:1; + struct snd_kcontrol *kcontrol; +}; + static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -290,6 +321,181 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, } } +static int wm_coeff_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = ctl->len; + return 0; +} + +static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, + const void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmemdup(buf, ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_write(wm_coeff->regmap, reg, scratch, + ctl->len); + if (ret) { + adsp_err(adsp, "Failed to write %zu bytes to %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + kfree(scratch); + + return 0; +} + +static int wm_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(ctl->cache, p, ctl->len); + + if (!ctl->enabled) { + ctl->dirty = 1; + return 0; + } + + return wm_coeff_write_control(kcontrol, p, ctl->len); +} + +static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, + void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmalloc(ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len); + if (ret) { + adsp_err(adsp, "Failed to read %zu bytes from %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + memcpy(buf, scratch, ctl->len); + kfree(scratch); + + return 0; +} + +static int wm_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(p, ctl->cache, ctl->len); + return 0; +} + +static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl, + const struct snd_kcontrol_new *kctl) +{ + int ret; + struct snd_kcontrol *kcontrol; + + kcontrol = snd_ctl_new1(kctl, wm_coeff); + ret = snd_ctl_add(ctl->card, kcontrol); + if (ret < 0) { + dev_err(wm_coeff->dev, "Failed to add %s: %d\n", + kctl->name, ret); + return ret; + } + ctl->kcontrol = kcontrol; + return 0; +} + +struct wmfw_ctl_work { + struct wm_coeff *wm_coeff; + struct wm_coeff_ctl *ctl; + struct work_struct work; +}; + +static int wmfw_add_ctl(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl) +{ + struct snd_kcontrol_new *kcontrol; + int ret; + + if (!wm_coeff || !ctl || !ctl->name || !ctl->card) + return -EINVAL; + + kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); + if (!kcontrol) + return -ENOMEM; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + + kcontrol->name = ctl->name; + kcontrol->info = wm_coeff_info; + kcontrol->get = wm_coeff_get; + kcontrol->put = wm_coeff_put; + kcontrol->private_value = (unsigned long)ctl; + + ret = wm_coeff_add_kcontrol(wm_coeff, + ctl, kcontrol); + if (ret < 0) + goto err_kcontrol; + + kfree(kcontrol); + + list_add(&ctl->list, &wm_coeff->ctl_list); + return 0; + +err_kcontrol: + kfree(kcontrol); + return ret; +} + static int wm_adsp_load(struct wm_adsp *dsp) { LIST_HEAD(buf_list); @@ -503,7 +709,156 @@ out: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp) +static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled || ctl->dirty) + continue; + ret = wm_coeff_read_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + } + + return 0; +} + +static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled) + continue; + if (ctl->dirty) { + ret = wm_coeff_write_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + ctl->dirty = 0; + } + } + + return 0; +} + +static void wm_adsp_ctl_work(struct work_struct *work) +{ + struct wmfw_ctl_work *ctl_work = container_of(work, + struct wmfw_ctl_work, + work); + + wmfw_add_ctl(ctl_work->wm_coeff, ctl_work->ctl); + kfree(ctl_work); +} + +static int wm_adsp_create_control(struct snd_soc_codec *codec, + const struct wm_adsp_alg_region *region) + +{ + struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + struct wm_coeff_ctl *ctl; + struct wmfw_ctl_work *ctl_work; + char *name; + char *region_name; + int ret; + + name = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!name) + return -ENOMEM; + + switch (region->type) { + case WMFW_ADSP1_PM: + region_name = "PM"; + break; + case WMFW_ADSP1_DM: + region_name = "DM"; + break; + case WMFW_ADSP2_XM: + region_name = "XM"; + break; + case WMFW_ADSP2_YM: + region_name = "YM"; + break; + case WMFW_ADSP1_ZM: + region_name = "ZM"; + break; + default: + return -EINVAL; + } + + snprintf(name, PAGE_SIZE, "DSP%d %s %x", + dsp->num, region_name, region->alg); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + if (!strcmp(ctl->name, name)) { + if (!ctl->enabled) + ctl->enabled = 1; + return 0; + } + } + + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) { + ret = -ENOMEM; + goto err_name; + } + ctl->region = *region; + ctl->name = kmemdup(name, strlen(name) + 1, GFP_KERNEL); + if (!ctl->name) { + ret = -ENOMEM; + goto err_ctl; + } + ctl->enabled = 1; + ctl->dirty = 0; + ctl->ops.xget = wm_coeff_get; + ctl->ops.xput = wm_coeff_put; + ctl->card = codec->card->snd_card; + ctl->adsp = dsp; + + ctl->len = region->len; + ctl->cache = kzalloc(ctl->len, GFP_KERNEL); + if (!ctl->cache) { + ret = -ENOMEM; + goto err_ctl_name; + } + + ctl_work = kzalloc(sizeof(*ctl_work), GFP_KERNEL); + if (!ctl_work) { + ret = -ENOMEM; + goto err_ctl_cache; + } + + ctl_work->wm_coeff = dsp->wm_coeff; + ctl_work->ctl = ctl; + INIT_WORK(&ctl_work->work, wm_adsp_ctl_work); + schedule_work(&ctl_work->work); + + kfree(name); + + return 0; + +err_ctl_cache: + kfree(ctl->cache); +err_ctl_name: + kfree(ctl->name); +err_ctl: + kfree(ctl); +err_name: + kfree(name); + return ret; +} + +static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -644,7 +999,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].dm); + region->len -= be32_to_cpu(adsp1_alg[i].dm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region DM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -652,7 +1016,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp1_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } break; case WMFW_ADSP2: @@ -672,7 +1045,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].xm); + region->len -= be32_to_cpu(adsp2_alg[i].xm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region XM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -680,7 +1062,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].ym); + region->len -= be32_to_cpu(adsp2_alg[i].ym); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region YM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -688,7 +1079,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp2_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } break; } } @@ -884,6 +1284,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_coeff_ctl *ctl; int ret; int val; @@ -921,7 +1322,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -929,6 +1330,16 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + /* Start the core running */ regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_CORE_ENA | ADSP1_START, @@ -945,6 +1356,11 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, 0); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } break; default: @@ -997,6 +1413,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; unsigned int val; int ret; @@ -1062,7 +1479,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1070,6 +1487,16 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, ADSP2_CORE_ENA | ADSP2_START, @@ -1107,6 +1534,11 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret); } + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } + while (!list_empty(&dsp->alg_regions)) { alg_region = list_first_entry(&dsp->alg_regions, struct wm_adsp_alg_region, @@ -1145,36 +1577,48 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) INIT_LIST_HEAD(&adsp->alg_regions); + adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff), + GFP_KERNEL); + if (!adsp->wm_coeff) + return -ENOMEM; + adsp->wm_coeff->regmap = adsp->regmap; + adsp->wm_coeff->dev = adsp->dev; + INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list); + if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } } return 0; + +out_coeff: + kfree(adsp->wm_coeff); + return ret; } EXPORT_SYMBOL_GPL(wm_adsp2_init); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index cb8871a..e6e705a 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -30,6 +30,7 @@ struct wm_adsp_alg_region { unsigned int alg; int type; unsigned int base; + size_t len; }; struct wm_adsp { @@ -53,6 +54,8 @@ struct wm_adsp { bool running; struct regulator *dvfs; + + struct wm_coeff *wm_coeff; }; #define WM_ADSP1(wname, num) \ -- 1.8.2.2

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Mark CHIP_ANA_POWER as volatile
by Fabio Estevam 08 May '13

08 May '13

Register CHIP_ANA_POWER (address: 0x0030) contains all of the power down controls for the analog blocks. Mark it as a volatile register, so that sgtl5000 can be functional after a reset. Without this patch the following error happens after a audio playback followed by a reset: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b443..8b704ff 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -965,6 +965,7 @@ static bool sgtl5000_volatile(struct device *dev, unsigned int reg) switch (reg) { case SGTL5000_CHIP_ID: case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_ANA_POWER: case SGTL5000_CHIP_ANA_STATUS: return true; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs
by Takashi Iwai 08 May '13

08 May '13

This is a revised patch based on Mengdong Lin's fix patch, which is a supplement to a previous patch [1611a9c9: ALSA: hda - Add fixup for Haswell to enable all pin and convertor widgets]. Some Haswell BIOS will disable the 2nd and 3rd pin/covertor widgets when the HD-A controller changes state from D3 to D0. So when the controller resumes after a system or runtime suspend, these widgets are disabled and programming these widgets to D0 will cause H/W error and codec will not respond. In addition, we found out that some BIOS disables the pins at S3 although it shows up at boot. This confuses the driver utterly, and the hardware falls into the fatal communication error like the above. So in this patch, we apply intel_haswell_enable_all_pins() not only as a fixup to a certain device (with 8086:2010) but to all Haswell machines. The codec driver basically assumes that all pins are exposed, so it's anyway better to see them from the beginning. Even if all pins and converters are shown by this call, there should be no regression in practice: the pin default configurations are still kept, thus the disabled pins are handled as disabled by the driver properly. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 54 +++++++++++++++++++--------------------------- 1 file changed, 22 insertions(+), 32 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 32930e6..e12f7a0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1832,12 +1832,10 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, #define INTEL_EN_ALL_PIN_CVTS 0x01 /* enable 2nd & 3rd pins and convertors */ static void intel_haswell_enable_all_pins(struct hda_codec *codec, - const struct hda_fixup *fix, int action) + bool update_tree) { unsigned int vendor_param; - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, INTEL_GET_VENDOR_VERB, 0); if (vendor_param == -1 || vendor_param & INTEL_EN_ALL_PIN_CVTS) @@ -1849,8 +1847,8 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, if (vendor_param == -1) return; - snd_hda_codec_update_widgets(codec); - return; + if (update_tree) + snd_hda_codec_update_widgets(codec); } static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) @@ -1868,30 +1866,20 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) INTEL_SET_VENDOR_VERB, vendor_param); } +/* Haswell needs to re-issue the vendor-specific verbs before turning to D0. + * Otherwise you may get severe h/w communication errors. + */ +static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (power_state == AC_PWRST_D0) { + intel_haswell_enable_all_pins(codec, false); + intel_haswell_fixup_enable_dp12(codec); + } - -/* available models for fixup */ -enum { - INTEL_HASWELL, -}; - -static const struct hda_model_fixup hdmi_models[] = { - {.id = INTEL_HASWELL, .name = "Haswell"}, - {} -}; - -static const struct snd_pci_quirk hdmi_fixup_tbl[] = { - SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), - {} /* terminator */ -}; - -static const struct hda_fixup hdmi_fixups[] = { - [INTEL_HASWELL] = { - .type = HDA_FIXUP_FUNC, - .v.func = intel_haswell_enable_all_pins, - }, -}; - + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} static int patch_generic_hdmi(struct hda_codec *codec) { @@ -1904,11 +1892,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { + intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); + } if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; @@ -1916,6 +1903,9 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + if (codec->vendor_id == 0x80862807) + codec->patch_ops.set_power_state = haswell_set_power_state; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.8.2.1

1 0

[alsa-devel] [PATCH] ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs
by Takashi Iwai 08 May '13

08 May '13

This is a revised patch based on Mengdong Lin's fix patch, which is a supplement to a previous patch [1611a9c9: ALSA: hda - Add fixup for Haswell to enable all pin and convertor widgets]. Some Haswell BIOS will disable the 2nd and 3rd pin/covertor widgets when the HD-A controller changes state from D3 to D0. So when the controller resumes after a system or runtime suspend, these widgets are disabled and programming these widgets to D0 will cause H/W error and codec will not respond. In addition, we found out that some BIOS disables the pins at S3 although it shows up at boot. This confuses the driver utterly, and the hardware falls into the fatal communication error like the above. So in this patch, we apply intel_haswell_enable_all_pins() not only as a fixup to a certain device (with 8086:2010) but to all Haswell machines. The codec driver basically assumes that all pins are exposed, so it's anyway better to see them from the beginning. Even if all pins and converters are shown by this call, there should be no regression in practice: the pin default configurations are still kept, thus the disabled pins are handled as disabled by the driver properly. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 54 +++++++++++++++++++--------------------------- 1 file changed, 22 insertions(+), 32 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 32930e6..e12f7a0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1832,12 +1832,10 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, #define INTEL_EN_ALL_PIN_CVTS 0x01 /* enable 2nd & 3rd pins and convertors */ static void intel_haswell_enable_all_pins(struct hda_codec *codec, - const struct hda_fixup *fix, int action) + bool update_tree) { unsigned int vendor_param; - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, INTEL_GET_VENDOR_VERB, 0); if (vendor_param == -1 || vendor_param & INTEL_EN_ALL_PIN_CVTS) @@ -1849,8 +1847,8 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, if (vendor_param == -1) return; - snd_hda_codec_update_widgets(codec); - return; + if (update_tree) + snd_hda_codec_update_widgets(codec); } static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) @@ -1868,30 +1866,20 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) INTEL_SET_VENDOR_VERB, vendor_param); } +/* Haswell needs to re-issue the vendor-specific verbs before turning to D0. + * Otherwise you may get severe h/w communication errors. + */ +static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (power_state == AC_PWRST_D0) { + intel_haswell_enable_all_pins(codec, false); + intel_haswell_fixup_enable_dp12(codec); + } - -/* available models for fixup */ -enum { - INTEL_HASWELL, -}; - -static const struct hda_model_fixup hdmi_models[] = { - {.id = INTEL_HASWELL, .name = "Haswell"}, - {} -}; - -static const struct snd_pci_quirk hdmi_fixup_tbl[] = { - SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), - {} /* terminator */ -}; - -static const struct hda_fixup hdmi_fixups[] = { - [INTEL_HASWELL] = { - .type = HDA_FIXUP_FUNC, - .v.func = intel_haswell_enable_all_pins, - }, -}; - + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} static int patch_generic_hdmi(struct hda_codec *codec) { @@ -1904,11 +1892,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { + intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); + } if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; @@ -1916,6 +1903,9 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + if (codec->vendor_id == 0x80862807) + codec->patch_ops.set_power_state = haswell_set_power_state; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.8.2.1

1 0

Re: [alsa-devel] [RFC] ASoC: sgtl5000: Remove cache support
by Fabio Estevam 08 May '13

08 May '13

On Tue, May 7, 2013 at 5:06 PM, Matt Sealey <matt(a)genesi-usa.com> wrote: > Right, and in theory - I can't find an example of anyone implementing > VDDD as unconnected - if you float VDDD then the SGTL5000 will start Actually I can't find see anyone that implemented VDDD connected, except Efika. All these boards have sgtl5000 VDDD unconnected: mx51evk, mx28evk, mx53smd, mx35pdk, mx25pdk. mx6qsabrelite connects VDD to GND.

2 1

[alsa-devel] [PATCH v2] ALSA: hda - re-apply fixup when resuming haswell hdmi codec
by mengdong.lin＠intel.com 07 May '13

07 May '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is a supplement to a previous patch: "ALSA: hda - Add fixup for Haswell to enable all pin and convertor widgets". Some Haswell BIOS will disable the 2nd and 3rd pin/covertor widgets when the HD-A controller changes state from D3 to D0. So when the controller resumes after a system or runtime suspend, these widgets are disabled and programming these widgets to D0 will cause H/W error and codec will not respond. So this patch adds a "set_power_state" ops for Haswell codec. Before setting the codec to D0, this function will apply a BIOS-specific fixup to enable the 2nd & 3rd pins/cvts if need. And since BIOS will also disable DP1.2, so this function will check and enable DP1.2 mode if need. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index adb5dce..080398e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1842,7 +1842,7 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, { unsigned int vendor_param; - if (action != HDA_FIXUP_ACT_PRE_PROBE) + if (action != HDA_FIXUP_ACT_PRE_PROBE && action != HDA_FIXUP_ACT_INIT) return; vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, INTEL_GET_VENDOR_VERB, 0); @@ -1855,7 +1855,8 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, if (vendor_param == -1) return; - snd_hda_codec_update_widgets(codec); + if (action == HDA_FIXUP_ACT_PRE_PROBE) + snd_hda_codec_update_widgets(codec); return; } @@ -1874,7 +1875,24 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) INTEL_SET_VENDOR_VERB, vendor_param); } +static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (AC_PWRST_D0 == power_state) { + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); + + intel_haswell_fixup_enable_dp12(codec); + + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state); + + return; +} /* available models for fixup */ enum { @@ -1922,6 +1940,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + codec->patch_ops.set_power_state = haswell_set_power_state; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

3 5

Re: [alsa-devel] [PATCH] ASoC: wm0010: fix error return code in wm0010_boot()
by Mark Brown 07 May '13

07 May '13

On Tue, May 07, 2013 at 07:38:52PM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Fix to return -ENOMEM in the memory malloc of 'out' and 'img_swap' error > handling case instead of 0, as done elsewhere in this function. Applied, thanks. Please note my new e-mail address.

1 0

[alsa-devel] [PATCH] ASoC: sam9g20ek: Let device core handle pinctrl
by Mark Brown 07 May '13

07 May '13

Since commit ab78029 (drivers/pinctrl: grab default handles from device core) we can rely on device core for handling pinctrl so remove devm_pinctrl_get_select_default() from the driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/atmel/sam9g20_wm8731.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 2d6fbd0..802717e 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -38,8 +38,6 @@ #include <linux/platform_device.h> #include <linux/i2c.h> -#include <linux/pinctrl/consumer.h> - #include <linux/atmel-ssc.h> #include <sound/core.h> @@ -203,15 +201,8 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) struct device_node *codec_np, *cpu_np; struct clk *pllb; struct snd_soc_card *card = &snd_soc_at91sam9g20ek; - struct pinctrl *pinctrl; int ret; - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "Failed to request pinctrl for mck\n"); - return PTR_ERR(pinctrl); - } - if (!np) { if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc())) -- 1.7.10.4

3 3

[alsa-devel] [PATCH] ALSA: hda - re-apply fixup when resuming haswell hdmi codec
by mengdong.lin＠intel.com 07 May '13

07 May '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is a supplement to a previous patch: "ALSA: hda - Add fixup for Haswell to enable all pin and convertor widgets". Some Haswell BIOS will disable the 2nd and 3rd pin/covertor widgets when the HD-A controller changes state from D3 to D0. So when the controller resumes after a system or runtime suspend, these widgets are disabled and programming these widgets to D0 will cause H/W error and codec will not respond. So this patch adds a "set_power_state" ops for Haswell codec. Before setting the codec to D0, this function will apply a BIOS-specific fixup to enable the 2nd & 3rd pins/cvts if need. And since BIOS will also disable DP1.2, so this function will check and enable DP1.2 mode if need. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index adb5dce..9ecb55c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1842,7 +1842,7 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, { unsigned int vendor_param; - if (action != HDA_FIXUP_ACT_PRE_PROBE) + if (action != HDA_FIXUP_ACT_PRE_PROBE && action != HDA_FIXUP_ACT_INIT) return; vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, INTEL_GET_VENDOR_VERB, 0); @@ -1855,7 +1855,8 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, if (vendor_param == -1) return; - snd_hda_codec_update_widgets(codec); + if (action == HDA_FIXUP_ACT_PRE_PROBE) + snd_hda_codec_update_widgets(codec); return; } @@ -1874,7 +1875,27 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) INTEL_SET_VENDOR_VERB, vendor_param); } +static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + unsigned int state; + unsigned int vendor_param; + + if (AC_PWRST_D0 == power_state) { + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); + + intel_haswell_fixup_enable_dp12(codec); + + } + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); + + return; +} /* available models for fixup */ enum { @@ -1922,6 +1943,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + codec->patch_ops.set_power_state = haswell_set_power_state; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

2 1

[alsa-devel] [PATCH]ALSA: HDA: Fix Oops caused by dereference NULL pointer
by Wang YanQing 07 May '13

07 May '13

The interrupt handler azx_interrupt will call azx_update_rirb, which may call snd_hda_queue_unsol_event, snd_hda_queue_unsol_event will dereference chip->bus pointer. The problem is we alloc chip->bus in azx_codec_create which will be called after we enable IRQ and enable unsolicited event in azx_probe. This will cause Oops due dereference NULL pointer. I meet it, good luck:) Signed-off-by: Wang YanQing <udknight(a)gmail.com> --- sound/pci/hda/hda_codec.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 622f726..9c76752 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -618,6 +618,9 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) unsigned int wp; trace_hda_unsol_event(bus, res, res_ex); + if (!bus) + return 0; + unsol = bus->unsol; if (!unsol) return 0; -- 1.7.12.4.dirty

2 1

[alsa-devel] [PATCH 1/6] ASoC: Samsung: Remove redundant platform_set_drvdata()
by Sachin Kamat 07 May '13

07 May '13

Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/smdk_wm8580pcm.c | 1 - sound/soc/samsung/smdk_wm8994pcm.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index e43bd42..23a9204 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -176,7 +176,6 @@ static int snd_smdk_probe(struct platform_device *pdev) static int snd_smdk_remove(struct platform_device *pdev) { snd_soc_unregister_card(&smdk_pcm); - platform_set_drvdata(pdev, NULL); return 0; } diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 3688a32..0c84ca0 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -146,7 +146,6 @@ static int snd_smdk_probe(struct platform_device *pdev) static int snd_smdk_remove(struct platform_device *pdev) { snd_soc_unregister_card(&smdk_pcm); - platform_set_drvdata(pdev, NULL); return 0; } -- 1.7.9.5

4 12

[alsa-devel] [PATCH v2 12/13] ASoC: fsl: imx-audmux: Let device core handle pinctrl
by Fabio Estevam 07 May '13

07 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit ab78029 (drivers/pinctrl: grab default handles from device core), we can rely on device core for handling pinctrl. So remove devm_pinctrl_get_select_default() from the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Rebased against topic/fsl branch sound/soc/fsl/imx-audmux.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 47f046a..e260f1f 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -26,7 +26,6 @@ #include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <linux/pinctrl/consumer.h> #include "imx-audmux.h" @@ -247,7 +246,6 @@ EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); static int imx_audmux_probe(struct platform_device *pdev) { struct resource *res; - struct pinctrl *pinctrl; const struct of_device_id *of_id = of_match_device(imx_audmux_dt_ids, &pdev->dev); @@ -256,12 +254,6 @@ static int imx_audmux_probe(struct platform_device *pdev) if (IS_ERR(audmux_base)) return PTR_ERR(audmux_base); - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "setup pinctrl failed!"); - return PTR_ERR(pinctrl); - } - audmux_clk = devm_clk_get(&pdev->dev, "audmux"); if (IS_ERR(audmux_clk)) { dev_dbg(&pdev->dev, "cannot get clock: %ld\n", -- 1.7.9.5

2 1

Re: [alsa-devel] [RFC] ASoC: sgtl5000: Remove cache support
by Mark Brown 07 May '13

07 May '13

On Mon, May 06, 2013 at 03:19:27PM -0500, Matt Sealey wrote: > I have asked before and I will ask again; does anyone have any insight Not on the list you haven't as far as I can tell... this is the first I've heard of any concern about this code, I'd not be surprised if others were similarly unaware of any concerns. > The patch came from out of nowhere, and the comments seem to say that > the LDO gets enabled on every chip higher than a specific version. > Now, this is totally against the design documentation for the SGTL5000 > which specifically states that using the internal LDO is quite > possibly the *worst* power consuming case and NOT best practise for > putting the codec on a design, with the sole benefit being reduced > external component count. Are you sure it's purely about power consumption? A common application for LDOs in this situation is to clean up noise on the incoming power supply, improving rejection of noise. This is mostly irrelevant for digital supplies but can be desirable for analogue ones. Of course this looks like it's being used as a digital supply so... I have to say your explanation is a little curious, though - normally an LDO would require an external capacitor and so would either increase the component count (adding the capactior) or have no effect (if one is already present for decoupling). What is the supply for this LDO and where does the output go? > What happens if we just take out the internal LDO setup conditional on > version, and only initialize it if VDDD is missing? Do boards that If the LDO is enabled and connected to a supply which is also driven externally that's not a clever idea, the two regulators will both be trying to regulate the same supply. > I am not entirely sure how well the commit linked by Fabio would work > on mainline since if it uses hw_read/write it should also be updating > the cache with those values to be sure they can later be modified... > the BSP code does that then fills the cache with the existing values > afterwards. That commit is just bogus in the way it interacts with the cache, half of it looks awfully like it's trying to work around some breakage in the way the cache was being set up. > So maybe we don't disable the cache, or it gets ported to regmap, but > the order should be enable regulators in a non-cached way then fill > the cache AFTERWARDS - the code at ToT mainline doesn't fix the cache I'm sorry, this makes no sense to me. If we know the physical defaults for the device and we can get the device into that state then we should have no need to read from the cache. If we don't know what they are then they shouldn't be listed. > after it does regulator/power regs setup, I assume because cache > filling is now somehow automated in ASoC (and if this is the case, There is no cache filling to automate, if (as is the case here) register defaults are provided they're used. > then the cache io setup needs to be done after regulator/power setup, > the defaults list/cache needs to be updated for those registers before > any register caching io is done..). And maybe setting up the regulator > is kind of pointless anyway (meaning that whole caching/not-caching > thing just falls away?) Again this makes no sense, it seems like you're very confused here. The whole purpose of the register defaults is to reflect the default values of the registers. The infrastructure relies on this being the case, otherwise it will get confused for example during cache sync when it decides if a write to the device is required. I think the point that you lost me was your assumption that the regulator enable needs to be done without the cache. What makes you believe that this is the case? Note that as I said in another post the driver is currently totally broken with regard to anything that actually powers off the device, there's never any attempt made to restore the register cache.

1 0

[alsa-devel] [PATCH 13/13] ASoC: mxs: mxs-saif: Let device core handle pinctrl
by Fabio Estevam 06 May '13

06 May '13

Since commit ab78029 (drivers/pinctrl: grab default handles from device core), we can rely on device core for handling pinctrl. So remove devm_pinctrl_get_select_default() from the driver. Cc: Mark Brown <broonie(a)kernel.org> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 3a2aa1d..12723b0 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -27,7 +27,6 @@ #include <linux/delay.h> #include <linux/time.h> #include <linux/fsl/mxs-dma.h> -#include <linux/pinctrl/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -661,7 +660,6 @@ static int mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct pinctrl *pinctrl; int ret = 0; struct device_node *master; @@ -701,12 +699,6 @@ static int mxs_saif_probe(struct platform_device *pdev) mxs_saif[saif->id] = saif; - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - ret = PTR_ERR(pinctrl); - return ret; - } - saif->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { ret = PTR_ERR(saif->clk); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 12/13] ASoC: fsl: imx-audmux: Let device core handle pinctrl
by Fabio Estevam 06 May '13

06 May '13

Since commit ab78029 (drivers/pinctrl: grab default handles from device core), we can rely on device core for handling pinctrl. So remove devm_pinctrl_get_select_default() from the driver. Cc: Mark Brown <broonie(a)kernel.org> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 3f333e5..22967c6 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -26,7 +26,6 @@ #include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <linux/pinctrl/consumer.h> #include "imx-audmux.h" @@ -247,7 +246,6 @@ EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); static int imx_audmux_probe(struct platform_device *pdev) { struct resource *res; - struct pinctrl *pinctrl; const struct of_device_id *of_id = of_match_device(imx_audmux_dt_ids, &pdev->dev); @@ -256,12 +254,6 @@ static int imx_audmux_probe(struct platform_device *pdev) if (IS_ERR(audmux_base)) return PTR_ERR(audmux_base); - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "setup pinctrl failed!"); - return PTR_ERR(pinctrl); - } - audmux_clk = clk_get(&pdev->dev, "audmux"); if (IS_ERR(audmux_clk)) { dev_dbg(&pdev->dev, "cannot get clock: %ld\n", -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: mxs-sgtl5000: Remove unneeded 'ret' variable
by Fabio Estevam 06 May '13

06 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Variable 'ret' is not needed here, so just remove it. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index b1d9b5e..4f74b05 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -116,7 +116,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i, ret = 0; + int i; if (!np) return 1; /* no device tree */ @@ -142,7 +142,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return ret; + return 0; } static int mxs_sgtl5000_probe(struct platform_device *pdev) -- 1.7.9.5

3 5

[alsa-devel] [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
by mengdong.lin＠intel.com 06 May '13

06 May '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. This patch - adds a codec flag to delay resuming a codec. System resume will skip the codecs if this flag is set, and these codecs will be resumed on later codec access. - adds a set_power_state ops for Haswell HDMI codec. In a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. 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 04b5738..bcb7205 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5508,11 +5508,15 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (codec->support_delay_resume) + codec->resume_delayed = 1; + else + hda_call_codec_resume(codec); } return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 23ca172..5b5e5f4 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -886,6 +886,8 @@ struct hda_codec { unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ unsigned int in_pm:1; /* suspend/resume being performed */ + unsigned int support_delay_resume:1; /* codec support delay resume */ + unsigned int resume_delayed:1; /* resume delayed by PM */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78e1827..d116908 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -98,6 +98,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -977,6 +985,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(&spec->pins[pin_idx], 1); snd_hda_jack_report_sync(codec); } @@ -1846,6 +1861,63 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + spec->ready_to_resume = 0; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = &spec->pins[pin_idx]; + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printd(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static inline void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = + intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume NULL +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1868,6 +1940,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

4 51

[alsa-devel] [PATCH] alsactl: sprintf to snprintf prevent buffer overflow
by Doug Goldstein 06 May '13

06 May '13

sprintf() is a bit dangerous unless you explicitly know your type size and want to keep it in sync always. Its safer to just use snprintf() and ensure your string doesn't overflow and is NULL terminated. Signed-off-by: Doug Goldstein <cardoe(a)cardoe.com> --- This was written to fix the buffer overflow crash in alsactl with the 1.0.27 release, which is documented in Gentoo bug #468160 [1]. The issue is actually already fixed upstream by 95788fea25c1a59985828d4b91af0772d077600b however the fix just changes the size and continues to use sprintf() which IMHO is still dangerous. The attached patch will fix the issue in a safer manner for good. [1] https://bugs.gentoo.org/show_bug.cgi?id=468160 --- alsactl/lock.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/alsactl/lock.c b/alsactl/lock.c index fce208b..587a109 100644 --- a/alsactl/lock.c +++ b/alsactl/lock.c @@ -53,9 +53,9 @@ static int state_lock_(const char *file, int lock, int timeout) lck.l_len = 11; lck.l_pid = 0; if (lock) { - sprintf(lcktxt, "%10li\n", (long)getpid()); + snprintf(lcktxt, sizeof(lcktxt), "%10li\n", (long)getpid()); } else { - sprintf(lcktxt, "%10s\n", ""); + snprintf(lcktxt, sizeof(lcktxt), "%10s\n", ""); } while (fd < 0 && timeout-- > 0) { fd = open(nfile, O_RDWR); -- 1.8.2.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: mips/hal2: Remove redundant platform_set_drvdata()
by Sachin Kamat 06 May '13

06 May '13

Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> Cc: Thomas Bogendoerfer <tsbogend(a)alpha.fanken.de> --- sound/mips/hal2.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index 7420c59..2b7f6e8 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -922,7 +922,6 @@ static int hal2_remove(struct platform_device *pdev) struct snd_card *card = platform_get_drvdata(pdev); snd_card_free(card); - platform_set_drvdata(pdev, NULL); return 0; } -- 1.7.9.5

2 3

[alsa-devel] [PATCH resend 2/2] ALSA: mips/sgio2audio: Remove redundant platform_set_drvdata()
by Sachin Kamat 06 May '13

06 May '13

Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> Cc: Vivien Chappelier <vivien.chappelier(a)linux-mips.org> --- Resending becasue of a typo in patch title --- sound/mips/sgio2audio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 01a03ef..cfe99ae 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -963,7 +963,6 @@ static int snd_sgio2audio_remove(struct platform_device *pdev) struct snd_card *card = platform_get_drvdata(pdev); snd_card_free(card); - platform_set_drvdata(pdev, NULL); return 0; } -- 1.7.9.5

1 0

[alsa-devel] [RFC 26/26] HID/ALSA: wiimote: add speaker support
by David Herrmann 05 May '13

05 May '13

Classic Wii Remotes provide a speaker on the device. We can stream PCM data, mute and change volume via special protocol requests and remote audio registers. The device supports several different formats, but fully understood are only signed 8-bit PCM and something that looks like 4-bit Yamaha ADPCM. Theoretically, we can set data-rates from 183Hz up to 12MHz, but a realistic range for Bluetooth l2cap is 1000-4000 Hz. Data is streamed as 20bytes blocks and must be sent in a constant rate. There is no way to read the current position. It would be way too slow, anyway. However, if we stream data too fast, older devices will report overrun errors. It is nearly impossible to avoid them as we have no fine-grained control over bluetooth radio. But that's probably also the reason why newer devices no longer send these errors. Note that 8bit PCM at such low rates has quite bad sound quality. You also need to use user-space programs like "sox/play" as other applications require rates >4000/8000Hz. The 4bit Yamaha ADPCM should improve sound-quality a lot, however, the kernel currently doesn't support this. So we either need to add ALSA support or add an internal encoder. But this will be implemented in follow-up patches so we first get the infrastructure ready. Thanks to David Henningsson for very helpful advice on the driver layout. The driver is also very loosely based on the sgio2audio driver by Vivien Chappelier <vivien.chappelier(a)linux-mips.org> and the dummy alsa sound driver. Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: David Herrmann <dh.herrmann(a)gmail.com> --- Hi This is my second revision of the Wii Remote sound-driver. It is based on the hotplugging rework in this series. It's the only patch that is marked as RFC but I thought I'd just include it in this series for completenes. If someone wants to give it a try, I used the "sox" tools for testing as most other tools require rates >4000/8000Hz. Sample created via: sox input.wav -b 8 -e signed -c 1 -r 2000 output.raw Play via: (prefix with: SOX_OPTS="--buffer 128" for better responsiveness) AUDIODEV=hw:1 play -e signed -b 8 -c 1 -r 2000 output.raw Comments welcome! I tested this with 4 different devices and all worked well. I also tested hotplugging while playing audio and I never got any deadlocks or oopses. It's the first time I worked with the sound-layer, but I'm now pretty sure I got everything right. Maybe I could even relax the heavy-locking but better be safe. Cheers David drivers/hid/Kconfig | 4 + drivers/hid/Makefile | 3 + drivers/hid/hid-wiimote-core.c | 66 +++- drivers/hid/hid-wiimote-modules.c | 5 + drivers/hid/hid-wiimote-speaker.c | 662 ++++++++++++++++++++++++++++++++++++++ drivers/hid/hid-wiimote.h | 17 + 6 files changed, 755 insertions(+), 2 deletions(-) create mode 100644 drivers/hid/hid-wiimote-speaker.c diff --git a/drivers/hid/Kconfig b/drivers/hid/Kconfig index e8ef86c..3bc81c8 100644 --- a/drivers/hid/Kconfig +++ b/drivers/hid/Kconfig @@ -699,6 +699,7 @@ config HID_WIIMOTE tristate "Nintendo Wii / Wii U peripherals" depends on HID depends on LEDS_CLASS + depends on !SND || HIGH_RES_TIMERS select POWER_SUPPLY select INPUT_FF_MEMLESS ---help--- @@ -715,6 +716,9 @@ config HID_WIIMOTE the Wii U Gamepad) might be supported in the future. But currently support is limited to Bluetooth based devices. + If Alsa sound support (CONFIG_SND) is enabled, this driver also provides + a sound card interface for Wii Remote speakers. + If unsure, say N. To compile this driver as a module, choose M here: the diff --git a/drivers/hid/Makefile b/drivers/hid/Makefile index 8b7106b..d149e10 100644 --- a/drivers/hid/Makefile +++ b/drivers/hid/Makefile @@ -32,6 +32,9 @@ hid-wiimote-y := hid-wiimote-core.o hid-wiimote-modules.o ifdef CONFIG_DEBUG_FS hid-wiimote-y += hid-wiimote-debug.o endif +ifdef CONFIG_SND + hid-wiimote-y += hid-wiimote-speaker.o +endif obj-$(CONFIG_HID_A4TECH) += hid-a4tech.o obj-$(CONFIG_HID_ACRUX) += hid-axff.o diff --git a/drivers/hid/hid-wiimote-core.c b/drivers/hid/hid-wiimote-core.c index 279a07d..ea07fc3 100644 --- a/drivers/hid/hid-wiimote-core.c +++ b/drivers/hid/hid-wiimote-core.c @@ -309,8 +309,8 @@ void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags) #define wiiproto_req_weeprom(wdata, os, buf, sz) \ wiiproto_req_wmem((wdata), true, (os), (buf), (sz)) -static void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, - __u32 offset, const __u8 *buf, __u8 size) +void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size) { __u8 cmd[22]; @@ -359,6 +359,58 @@ void wiiproto_req_rmem(struct wiimote_data *wdata, bool eeprom, __u32 offset, wiimote_queue(wdata, cmd, sizeof(cmd)); } +void wiiproto_req_speaker(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_SPEAKER; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +void wiiproto_req_mute(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_MUTE; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +/* send audio data, possibly fragmented into two blocks */ +size_t wiiproto_req_audio(struct wiimote_data *wdata, const __u8 *b1, + size_t len1, const __u8 *b2, size_t len2) +{ + __u8 cmd[22]; + size_t pos, len; + + pos = 0; + len = min_t(size_t, len1, 20ULL); + memcpy(&cmd[2 + pos], b1, len); + pos += len; + + if (pos < 20 && len2) { + len = min_t(size_t, len2, 20ULL - pos); + memcpy(&cmd[2 + pos], b2, len); + pos += len; + } + + cmd[0] = WIIPROTO_REQ_AUDIO; + cmd[1] = (pos & 0xff) << 3; + + for ( ; pos < 20; ++pos) + cmd[pos] = 0; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); + + return pos; +} + /* requries the cmd-mutex to be held */ int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size) @@ -570,6 +622,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN10] = (const __u8[]){ @@ -582,6 +635,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN20] = (const __u8[]){ @@ -595,6 +649,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_ACCEL, WIIMOD_IR, WIIMOD_BUILTIN_MP, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_BALANCE_BOARD] = (const __u8[]) { @@ -1469,6 +1524,13 @@ static void handler_return(struct wiimote_data *wdata, const __u8 *payload) if (wiimote_cmd_pending(wdata, cmd, 0)) { wdata->state.cmd_err = err; wiimote_cmd_complete(wdata); + } else if (cmd == WIIPROTO_REQ_AUDIO && err == 4) { + /* If we stream audio data too fast, we get overrun errors + * from the device. However, on audio-period bounds, + * we have a very hard time sending audio at the exact rate + * as we have no control over the bluetooth l2cap IRQs. + * Hence, drop any overrun errors. */ + hid_dbg(wdata->hdev, "audio overrun error\n"); } else if (err) { hid_warn(wdata->hdev, "Remote error %hhu on req %hhu\n", err, cmd); diff --git a/drivers/hid/hid-wiimote-modules.c b/drivers/hid/hid-wiimote-modules.c index ab25a64..25483aa 100644 --- a/drivers/hid/hid-wiimote-modules.c +++ b/drivers/hid/hid-wiimote-modules.c @@ -2074,6 +2074,11 @@ const struct wiimod_ops *wiimod_table[WIIMOD_NUM] = { [WIIMOD_IR] = &wiimod_ir, [WIIMOD_BUILTIN_MP] = &wiimod_builtin_mp, [WIIMOD_NO_MP] = &wiimod_no_mp, +#ifdef CONFIG_SND + [WIIMOD_SPEAKER] = &wiimod_speaker, +#else + [WIIMOD_SPEAKER] = &wiimod_dummy, +#endif }; const struct wiimod_ops *wiimod_ext_table[WIIMOTE_EXT_NUM] = { diff --git a/drivers/hid/hid-wiimote-speaker.c b/drivers/hid/hid-wiimote-speaker.c new file mode 100644 index 0000000..89b7f9f --- /dev/null +++ b/drivers/hid/hid-wiimote-speaker.c @@ -0,0 +1,662 @@ +/* + * Driver for audio speakers of Nintendo Wii / Wii U peripherals + * Copyright (c) 2012-2013 David Herrmann <dh.herrmann(a)gmail.com> + */ + +/* + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + */ + +/* + * Audio Speakers + * Some Wii peripherals provide an audio speaker that supports 8bit PCM, 4bit + * Yamaha ADPCM and some other mostly unknown formats. Not all setup options + * are known, but we know how to setup an 8bit PCM or 4bit ADPCM stream and + * adjust volume. Data is sent as 20bytes chunks and needs to be streamed at + * a constant rate. + */ + +#include <linux/device.h> +#include <linux/hid.h> +#include <linux/hrtimer.h> +#include <linux/interrupt.h> +#include <linux/mutex.h> +#include <linux/spinlock.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "hid-wiimote.h" + +struct wiimote_speaker { + spinlock_t lock; + struct snd_card *card; + struct wiimote_data *wdata; + unsigned int online : 1; + unsigned int mute : 1; + __u8 volume; + + unsigned long pos; + struct snd_pcm_substream *subs; + struct mutex runlock; + + struct hrtimer timer; + struct tasklet_struct tasklet; +}; + +enum wiimod_speaker_mode { + WIIMOD_SPEAKER_MODE_PCM8, + /* TODO: 4bit Yamaha ADPCM is currently not implemented */ + WIIMOD_SPEAKER_MODE_ADPCM4, +}; + +static int wiimod_speaker_enable(struct wiimote_speaker *speaker) +{ + struct wiimote_data *wdata = speaker->wdata; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (!speaker->online) { + speaker->online = 1; + wiiproto_req_speaker(wdata, true); + wiiproto_req_mute(wdata, speaker->mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static void wiimod_speaker_disable(struct wiimote_speaker *speaker) +{ + struct wiimote_data *wdata = speaker->wdata; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->online) { + speaker->online = 0; + wiiproto_req_speaker(wdata, false); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_mute(struct wiimote_speaker *speaker, bool mute) +{ + struct wiimote_data *wdata = speaker->wdata; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->mute != mute) { + speaker->mute = mute; + if (speaker->online) + wiiproto_req_mute(wdata, mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_volume(struct wiimote_speaker *speaker, + __u8 volume) +{ + struct wiimote_data *wdata = speaker->wdata; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->volume != volume) { + speaker->volume = volume; + if (speaker->online) + wiiproto_req_wmem(wdata, false, 0xa20005, &volume, + sizeof(volume)); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +/* Change speaker configuration. \mode can be one of WIIMOD_SPEAKER_MODE_*, + * \rate is the PCM sample rate and \volume is the requested volume. */ +static int wiimod_speaker_setup(struct wiimote_speaker *speaker, + __u8 mode, __u16 rate, __s16 volume) +{ + struct wiimote_data *wdata = speaker->wdata; + unsigned long flags; + __u8 config[7], m, wmem; + __u16 r; + int ret; + + if (!rate) + return -EINVAL; + + switch (mode) { + case WIIMOD_SPEAKER_MODE_PCM8: + r = 12000000ULL / rate; + m = 0x40; + break; + case WIIMOD_SPEAKER_MODE_ADPCM4: + r = 6000000ULL / rate; + m = 0x00; + break; + default: + return -EINVAL; + } + + config[0] = 0x00; + config[1] = m; + config[2] = r & 0x00ff; + config[3] = (r & 0xff00) >> 8; + config[4] = volume & 0xff; + config[5] = 0x00; + config[6] = 0x00; + + wiimote_cmd_acquire_noint(wdata); + + /* mute speaker during setup and read/write volume field */ + spin_lock_irqsave(&wdata->state.lock, flags); + + wiiproto_req_mute(wdata, true); + if (volume < 0) + config[4] = speaker->volume; + else + speaker->volume = volume; + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + /* power speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20009, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* prepare setup */ + wmem = 0x08; + ret = wiimote_cmd_write(wdata, 0xa20001, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* write configuration */ + ret = wiimote_cmd_write(wdata, 0xa20001, config, sizeof(config)); + if (ret) + goto out_unlock; + + /* enable speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20008, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + +out_unlock: + wiimote_cmd_release(wdata); + return ret; +} + +/* returns true if a period has elapsed */ +static bool wiimod_speaker_push(struct wiimote_speaker *speaker) +{ + struct wiimote_data *wdata = speaker->wdata; + struct snd_pcm_runtime *runtime = speaker->subs->runtime; + unsigned long flags; + bool elapsed; + unsigned long buflen, plen, len; + __u8 *src; + + spin_lock_irqsave(&wdata->state.lock, flags); + + buflen = frames_to_bytes(runtime, runtime->buffer_size); + plen = frames_to_bytes(runtime, runtime->period_size); + src = runtime->dma_area; + + len = buflen - speaker->pos; + + /* no need to send data while muted or offline */ + if (speaker->online && !speaker->mute) { + if (len < 20) + wiiproto_req_audio(wdata, &src[speaker->pos], + len, src, 20 - len); + else + wiiproto_req_audio(wdata, &src[speaker->pos], + 20, NULL, 0); + } + + speaker->pos += 20; + speaker->pos %= buflen; + elapsed = (speaker->pos % plen) < 20; + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return elapsed; +} + +/* timer handling */ + +static void wiimod_speaker_task(unsigned long data) +{ + struct wiimote_speaker *speaker = (void*)data; + unsigned long flags; + bool elapsed = false; + + spin_lock_irqsave(&speaker->lock, flags); + if (speaker->wdata) + elapsed = wiimod_speaker_push(speaker); + spin_unlock_irqrestore(&speaker->lock, flags); + + if (elapsed) + snd_pcm_period_elapsed(speaker->subs); +} + +static enum hrtimer_restart wiimod_speaker_tick(struct hrtimer *timer) +{ + struct wiimote_speaker *speaker = container_of(timer, + struct wiimote_speaker, + timer); + unsigned long missed; + __u64 ival; + + tasklet_schedule(&speaker->tasklet); + + ival = 1000000000ULL * 20U; + ival /= speaker->subs->runtime->rate; + + missed = hrtimer_forward_now(timer, ns_to_ktime(ival)); + if (missed > 1) + snd_printdd("wiimote: speaker: missed %lu timer interrupts\n", + missed - 1); + + return HRTIMER_RESTART; +} + +/* PCM layer */ + +static const struct snd_pcm_hardware wiimod_speaker_playback_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 500, + .rate_max = 2000, /* TODO: >2000 only supported by 2nd gen */ + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = 32768, + .period_bytes_min = 128, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, +}; + +static int wiimod_speaker_playback_open(struct snd_pcm_substream *substream) +{ + struct wiimote_speaker *speaker = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long flags; + int ret; + + runtime->hw = wiimod_speaker_playback_hw; + + spin_lock_irqsave(&speaker->lock, flags); + ret = -ENODEV; + if (speaker->wdata) { + runtime->private_data = speaker->wdata; + ret = wiimod_speaker_enable(speaker); + } + spin_unlock_irqrestore(&speaker->lock, flags); + + return ret; +} + +static int wiimod_speaker_playback_close(struct snd_pcm_substream *substream) +{ + struct wiimote_speaker *speaker = snd_pcm_substream_chip(substream); + unsigned long flags; + + spin_lock_irqsave(&speaker->lock, flags); + if (speaker->wdata) + wiimod_speaker_disable(speaker); + spin_unlock_irqrestore(&speaker->lock, flags); + + hrtimer_cancel(&speaker->timer); + tasklet_kill(&speaker->tasklet); + + return 0; +} + +static int wiimod_speaker_playback_hw_params(struct snd_pcm_substream *subs, + struct snd_pcm_hw_params *hw) +{ + return snd_pcm_lib_alloc_vmalloc_buffer(subs, params_buffer_bytes(hw)); +} + +static int wiimod_speaker_playback_hw_free(struct snd_pcm_substream *subs) +{ + return snd_pcm_lib_free_vmalloc_buffer(subs); +} + +static int wiimod_speaker_playback_prepare(struct snd_pcm_substream *subs) +{ + struct wiimote_speaker *speaker = snd_pcm_substream_chip(subs); + struct wiimote_data *wdata; + struct snd_pcm_runtime *runtime = subs->runtime; + int ret, online; + unsigned long flags; + + /* runlock synchronizes with device hotplugging */ + mutex_lock(&speaker->runlock); + + spin_lock_irqsave(&speaker->lock, flags); + wdata = speaker->wdata; + online = speaker->online; + spin_unlock_irqrestore(&speaker->lock, flags); + + if (!wdata || !online) { + ret = -ENODEV; + goto unlock; + } + + spin_lock_irqsave(&wdata->state.lock, flags); + speaker->pos = 0; + speaker->subs = subs; + spin_unlock_irqrestore(&wdata->state.lock, flags); + + /* Perform speaker setup. This may take a few milliseconds and the + * handlers perform synchronous network operations so this + * may sleep. */ + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S8: + ret = wiimod_speaker_setup(speaker, WIIMOD_SPEAKER_MODE_PCM8, + runtime->rate, -1); + break; + default: + ret = -EINVAL; + } + +unlock: + mutex_unlock(&speaker->runlock); + return ret; +} + +static int wiimod_speaker_playback_trigger(struct snd_pcm_substream *subs, + int cmd) +{ + struct wiimote_speaker *speaker = snd_pcm_substream_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + struct wiimote_data *wdata; + unsigned long flags; + __u64 ival; + + spin_lock_irqsave(&speaker->lock, flags); + + wdata = speaker->wdata; + if (!wdata || !speaker->online) + goto unlock; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* unmute device on start if not muted by user-space */ + spin_lock(&wdata->state.lock); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock(&wdata->state.lock); + + ival = 20U * 1000000000ULL / runtime->rate; + hrtimer_start(&speaker->timer, ktime_set(0, ival), + HRTIMER_MODE_REL); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* mute device when stopping transmission */ + spin_lock(&wdata->state.lock); + wiiproto_req_mute(wdata, true); + spin_unlock(&wdata->state.lock); + + hrtimer_cancel(&speaker->timer); + break; + case SNDRV_PCM_TRIGGER_RESUME: + /* unmute device on start if not muted by user-space */ + spin_lock(&wdata->state.lock); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock(&wdata->state.lock); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + /* mute device when stopping transmission */ + spin_lock(&wdata->state.lock); + wiiproto_req_mute(wdata, true); + spin_unlock(&wdata->state.lock); + break; + } + +unlock: + spin_unlock_irqrestore(&speaker->lock, flags); + return 0; +} + +static snd_pcm_uframes_t wiimod_speaker_playback_pointer(struct snd_pcm_substream *subs) +{ + struct wiimote_speaker *speaker = snd_pcm_substream_chip(subs); + + return bytes_to_frames(subs->runtime, speaker->pos); +} + +static struct snd_pcm_ops wiimod_speaker_playback_ops = { + .open = wiimod_speaker_playback_open, + .close = wiimod_speaker_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = wiimod_speaker_playback_hw_params, + .hw_free = wiimod_speaker_playback_hw_free, + .prepare = wiimod_speaker_playback_prepare, + .trigger = wiimod_speaker_playback_trigger, + .pointer = wiimod_speaker_playback_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, +}; + +/* volume control */ + +static int wiimod_speaker_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 1; + info->value.integer.min = 0; + info->value.integer.max = 0xff; + + return 0; +} + +static int wiimod_speaker_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_speaker *speaker = snd_kcontrol_chip(kcontrol); + + val->value.integer.value[0] = speaker->volume; + + return 0; +} + +static int wiimod_speaker_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_speaker *speaker = snd_kcontrol_chip(kcontrol); + unsigned long value, flags; + + value = val->value.integer.value[0]; + if (value > 0xff) + value = 0xff; + + spin_lock_irqsave(&speaker->lock, flags); + if (speaker->wdata) + wiimod_speaker_set_volume(speaker, value); + spin_unlock_irqrestore(&speaker->lock, flags); + + return 0; +} + +static const struct snd_kcontrol_new wiimod_speaker_volume = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = wiimod_speaker_volume_info, + .get = wiimod_speaker_volume_get, + .put = wiimod_speaker_volume_put, +}; + +/* mute control */ + +static int wiimod_speaker_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_speaker *speaker = snd_kcontrol_chip(kcontrol); + + val->value.integer.value[0] = !speaker->mute; + + return 0; +} + +static int wiimod_speaker_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_speaker *speaker = snd_kcontrol_chip(kcontrol); + unsigned long flags; + + spin_lock_irqsave(&speaker->lock, flags); + if (speaker->wdata) + wiimod_speaker_set_mute(speaker, + !val->value.integer.value[0]); + spin_unlock_irqrestore(&speaker->lock, flags); + + return 0; +} + +static const struct snd_kcontrol_new wiimod_speaker_mute = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = wiimod_speaker_mute_get, + .put = wiimod_speaker_mute_put, +}; + +/* initialization and setup */ + +static int wiimod_speaker_probe(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + int ret; + struct wiimote_speaker *speaker; + struct snd_card *card; + struct snd_kcontrol *kcontrol; + struct snd_pcm *pcm; + + /* create sound card device */ + ret = snd_card_create(-1, NULL, THIS_MODULE, + sizeof(struct wiimote_speaker), &card); + if (ret) + return ret; + speaker = card->private_data; + + wdata->speaker = speaker; + speaker->wdata = wdata; + speaker->card = card; + speaker->mute = 1; + speaker->volume = 0xff; + strcpy(card->driver, "hid-wiimote"); + strcpy(card->shortname, "wiimote"); + strcpy(card->longname, "Nintendo Wii Remote speaker"); + + spin_lock_init(&speaker->lock); + mutex_init(&speaker->runlock); + hrtimer_init(&speaker->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + speaker->timer.function = wiimod_speaker_tick; + tasklet_init(&speaker->tasklet, wiimod_speaker_task, + (unsigned long)speaker); + + /* create volume control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_volume, speaker); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create mute control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_mute, speaker); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create PCM sub-device for playback */ + ret = snd_pcm_new(card, "Speaker", 0, 1, 0, &pcm); + if (ret) + goto err_free; + + pcm->private_data = speaker; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &wiimod_speaker_playback_ops); + + /* register sound card */ + snd_card_set_dev(card, &wdata->hdev->dev); + ret = snd_card_register(card); + if (ret) + goto err_free; + + return 0; + +err_free: + snd_card_free(card); + wdata->speaker = NULL; + return ret; +} + +static void wiimod_speaker_remove(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + if (!speaker) + return; + + mutex_lock(&speaker->runlock); + spin_lock_irqsave(&speaker->lock, flags); + + wdata->speaker = NULL; + speaker->online = 0; + speaker->wdata = NULL; + + spin_lock(&wdata->state.lock); + wiiproto_req_mute(wdata, true); + wiiproto_req_speaker(wdata, false); + spin_unlock(&wdata->state.lock); + + spin_unlock_irqrestore(&speaker->lock, flags); + mutex_unlock(&speaker->runlock); + + hrtimer_cancel(&speaker->timer); + tasklet_kill(&speaker->tasklet); + snd_card_free_when_closed(speaker->card); +} + +const struct wiimod_ops wiimod_speaker = { + .flags = 0, + .arg = 0, + .probe = wiimod_speaker_probe, + .remove = wiimod_speaker_remove, +}; diff --git a/drivers/hid/hid-wiimote.h b/drivers/hid/hid-wiimote.h index f1474f3..63a9244 100644 --- a/drivers/hid/hid-wiimote.h +++ b/drivers/hid/hid-wiimote.h @@ -147,6 +147,7 @@ struct wiimote_data { struct input_dev *mp; struct timer_list timer; struct wiimote_debug *debug; + struct wiimote_speaker *speaker; union { struct input_dev *input; @@ -171,6 +172,7 @@ enum wiimod_module { WIIMOD_IR, WIIMOD_BUILTIN_MP, WIIMOD_NO_MP, + WIIMOD_SPEAKER, WIIMOD_NUM, WIIMOD_NULL = WIIMOD_NUM, }; @@ -207,9 +209,12 @@ enum wiiproto_reqs { WIIPROTO_REQ_LED = 0x11, WIIPROTO_REQ_DRM = 0x12, WIIPROTO_REQ_IR1 = 0x13, + WIIPROTO_REQ_SPEAKER = 0x14, WIIPROTO_REQ_SREQ = 0x15, WIIPROTO_REQ_WMEM = 0x16, WIIPROTO_REQ_RMEM = 0x17, + WIIPROTO_REQ_AUDIO = 0x18, + WIIPROTO_REQ_MUTE = 0x19, WIIPROTO_REQ_IR2 = 0x1a, WIIPROTO_REQ_STATUS = 0x20, WIIPROTO_REQ_DATA = 0x21, @@ -263,6 +268,12 @@ extern void wiiproto_req_status(struct wiimote_data *wdata); extern void wiiproto_req_accel(struct wiimote_data *wdata, __u8 accel); extern void wiiproto_req_ir1(struct wiimote_data *wdata, __u8 flags); extern void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags); +extern void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size); +extern void wiiproto_req_speaker(struct wiimote_data *wdata, bool on); +extern void wiiproto_req_mute(struct wiimote_data *wdata, bool on); +extern size_t wiiproto_req_audio(struct wiimote_data *wdata, const __u8 *b1, + size_t len1, const __u8 *b2, size_t len2); extern int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size); extern ssize_t wiimote_cmd_read(struct wiimote_data *wdata, __u32 offset, @@ -287,6 +298,12 @@ static inline void wiidebug_deinit(void *u) { } #endif +#ifdef CONFIG_SND + +extern const struct wiimod_ops wiimod_speaker; + +#endif + /* requires the state.lock spinlock to be held */ static inline bool wiimote_cmd_pending(struct wiimote_data *wdata, int cmd, __u32 opt) -- 1.8.2.2

1 0

[alsa-devel] [RFC] Sound/HID: wiimote: add speaker support
by David Herrmann 05 May '13

05 May '13

Classic Wii Remotes provide a speaker on the device. We can stream PCM data, mute and change volume via special protocol requests and remote audio registers. The device supports several different formats, but fully understood are only signed 8-bit PCM and something that looks like 4-bit Yamaha ADPCM. Theoretically, we can set data-rates from 183Hz up to 12MHz, but a realistic range for Bluetooth l2cap is 1500-4000 Hz. Data is streamed as 20bytes blocks and must be sent in a constant rate. There is no way to read the current position. It would be way too slow, anyway. Signed-off-by: David Herrmann <dh.herrmann(a)gmail.com> --- Hi I'm reworking large parts of the HID Nintendo Wii Remote driver and wanted to add speaker support. I have never worked with the sound subsystem until now and need some help. The Wii Remote is a Bluetooth gamepad that features a small speaker on the remote. All information was gathered via reverse-engineering so I cannot tell you any specifics about the hardware, sorry. What I figured out so far is how to set up the speaker, stream data, set volume and mute the speaker. Supported formats are: signed 8-bit PCM 4-bit Yamaha ADPCM I implemented only the 8-bit PCM as I couldn't figure out whether the kernel supports the other format? The wiiproto_* helpers in hid-wiimote-core.c are used to send the requests. Data is streamed as 20-bytes packets and must be sent at a constant rate. I would really appreciate if someone can have a look at hid-wiimote-speaker.c and help me fill the gaps. I figured out how to write the basic snd_card management and PCM setup, but I am stuck with buffer management now. I have a function called wiiproto_req_audio() which sends up to 20bytes of PCM data to the device. However, I cannot figure out how to integrate that into the snd_pcm object. The problem is that all other drivers I found have large DMA buffers which are filled whenever an interrupt signals more data. However, I don't have this setup but instead I need a timer (does the sound-core provide that?) which causes the "copy" callback to be called with 20bytes of data at a constant rate. Any help welcome! And if this whole idea is stupid, please tell me! ;) Thanks David drivers/hid/Kconfig | 3 + drivers/hid/Makefile | 3 + drivers/hid/hid-wiimote-core.c | 54 +++- drivers/hid/hid-wiimote-modules.c | 5 + drivers/hid/hid-wiimote-speaker.c | 539 ++++++++++++++++++++++++++++++++++++++ drivers/hid/hid-wiimote.h | 17 ++ 6 files changed, 619 insertions(+), 2 deletions(-) create mode 100644 drivers/hid/hid-wiimote-speaker.c diff --git a/drivers/hid/Kconfig b/drivers/hid/Kconfig index e8ef86c..0b4d7e2 100644 --- a/drivers/hid/Kconfig +++ b/drivers/hid/Kconfig @@ -715,6 +715,9 @@ config HID_WIIMOTE the Wii U Gamepad) might be supported in the future. But currently support is limited to Bluetooth based devices. + If Alsa sound support (CONFIG_SND) is enabled, this driver also provides + a sound card interface for Wii Remote speakers. + If unsure, say N. To compile this driver as a module, choose M here: the diff --git a/drivers/hid/Makefile b/drivers/hid/Makefile index 8b7106b..d149e10 100644 --- a/drivers/hid/Makefile +++ b/drivers/hid/Makefile @@ -32,6 +32,9 @@ hid-wiimote-y := hid-wiimote-core.o hid-wiimote-modules.o ifdef CONFIG_DEBUG_FS hid-wiimote-y += hid-wiimote-debug.o endif +ifdef CONFIG_SND + hid-wiimote-y += hid-wiimote-speaker.o +endif obj-$(CONFIG_HID_A4TECH) += hid-a4tech.o obj-$(CONFIG_HID_ACRUX) += hid-axff.o diff --git a/drivers/hid/hid-wiimote-core.c b/drivers/hid/hid-wiimote-core.c index 906c146..53f1e88 100644 --- a/drivers/hid/hid-wiimote-core.c +++ b/drivers/hid/hid-wiimote-core.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/mutex.h> #include <linux/spinlock.h> +#include <linux/uaccess.h> #include "hid-ids.h" #include "hid-wiimote.h" @@ -309,8 +310,8 @@ void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags) #define wiiproto_req_weeprom(wdata, os, buf, sz) \ wiiproto_req_wmem((wdata), true, (os), (buf), (sz)) -static void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, - __u32 offset, const __u8 *buf, __u8 size) +void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size) { __u8 cmd[22]; @@ -359,6 +360,52 @@ void wiiproto_req_rmem(struct wiimote_data *wdata, bool eeprom, __u32 offset, wiimote_queue(wdata, cmd, sizeof(cmd)); } +void wiiproto_req_speaker(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_SPEAKER; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +void wiiproto_req_mute(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_MUTE; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +/* must not be called from atomic contexts */ +int wiiproto_req_audio_user(struct wiimote_data *wdata, + void __user *buf, size_t len) +{ + unsigned long flags; + __u8 cmd[22]; + + if (len > 20) + len = 20; + + cmd[0] = WIIPROTO_REQ_AUDIO; + cmd[1] = (len & 0xff) << 3; + + if (copy_from_user(&cmd[2], buf, len)) + return -EFAULT; + + spin_lock_irqsave(&wdata->state.lock, flags); + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + /* requries the cmd-mutex to be held */ int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size) @@ -570,6 +617,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN10] = (const __u8[]){ @@ -582,6 +630,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN20] = (const __u8[]){ @@ -595,6 +644,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_ACCEL, WIIMOD_IR, WIIMOD_BUILTIN_MP, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_BALANCE_BOARD] = (const __u8[]) { diff --git a/drivers/hid/hid-wiimote-modules.c b/drivers/hid/hid-wiimote-modules.c index 4ce0e02..7acf721 100644 --- a/drivers/hid/hid-wiimote-modules.c +++ b/drivers/hid/hid-wiimote-modules.c @@ -2072,6 +2072,11 @@ const struct wiimod_ops *wiimod_table[WIIMOD_NUM] = { [WIIMOD_IR] = &wiimod_ir, [WIIMOD_BUILTIN_MP] = &wiimod_builtin_mp, [WIIMOD_NO_MP] = &wiimod_no_mp, +#ifdef CONFIG_SND + [WIIMOD_SPEAKER] = &wiimod_speaker, +#else + [WIIMOD_SPEAKER] = &wiimod_dummy, +#endif }; const struct wiimod_ops *wiimod_ext_table[WIIMOTE_EXT_NUM] = { diff --git a/drivers/hid/hid-wiimote-speaker.c b/drivers/hid/hid-wiimote-speaker.c new file mode 100644 index 0000000..ecec5d9 --- /dev/null +++ b/drivers/hid/hid-wiimote-speaker.c @@ -0,0 +1,539 @@ +/* + * Driver for audio speakers of Nintendo Wii / Wii U peripherals + * Copyright (c) 2012-2013 David Herrmann <dh.herrmann(a)gmail.com> + */ + +/* + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + */ + +/* + * Audio Speakers + * Some Wii peripherals provide an audio speaker that supports 8bit PCM and + * some other mostly unknown formats. Not all setup options are known, but we + * know how to setup an 8bit PCM or 4bit ADPCM stream and adjust volume. Data + * is sent as 20bytes chunks and needs to be streamed at a constant rate. + */ + +#include <linux/device.h> +#include <linux/hid.h> +#include <linux/mutex.h> +#include <linux/spinlock.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "hid-wiimote.h" + +struct wiimote_speaker { + struct snd_card *card; + unsigned int online : 1; + unsigned int mute : 1; + __u8 volume; +}; + +enum wiimod_speaker_mode { + WIIMOD_SPEAKER_MODE_PCM8, + WIIMOD_SPEAKER_MODE_ADPCM4, +}; + +static int wiimod_speaker_enable(struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (!speaker->online) { + speaker->online = 1; + wiiproto_req_speaker(wdata, true); + wiiproto_req_mute(wdata, speaker->mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static void wiimod_speaker_disable(struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->online) { + speaker->online = 0; + wiiproto_req_speaker(wdata, false); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_mute(struct wiimote_data *wdata, bool mute) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->mute != mute) { + speaker->mute = mute; + if (speaker->online) + wiiproto_req_mute(wdata, mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_volume(struct wiimote_data *wdata, __u8 volume) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->volume != volume) { + speaker->volume = volume; + if (speaker->online) + wiiproto_req_wmem(wdata, false, 0xa20005, &volume, + sizeof(volume)); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +/* Change speaker configuration. \mode can be one of WIIMOD_SPEAKER_MODE_*, + * \rate is the PCM sample rate and \volume is the requested volume. */ +static int wiimod_speaker_setup(struct wiimote_data *wdata, + __u8 mode, __u16 rate, __s16 volume) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + __u8 config[7], m, wmem; + __u16 r; + int ret; + + if (!rate) + return -EINVAL; + + switch (mode) { + case WIIMOD_SPEAKER_MODE_PCM8: + r = 12000000ULL / rate; + m = 0x40; + break; + case WIIMOD_SPEAKER_MODE_ADPCM4: + r = 6000000ULL / rate; + m = 0x00; + break; + default: + return -EINVAL; + } + + config[0] = 0x00; + config[1] = m; + config[2] = r & 0x00ff; + config[3] = (r & 0xff00) >> 8; + config[4] = volume & 0xff; + config[5] = 0x00; + config[6] = 0x00; + + wiimote_cmd_acquire_noint(wdata); + + /* mute speaker during setup and read/write volume field */ + spin_lock_irqsave(&wdata->state.lock, flags); + + wiiproto_req_mute(wdata, true); + if (volume < 0) + config[4] = speaker->volume; + else + speaker->volume = volume; + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + /* power speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20009, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* prepare setup */ + wmem = 0x08; + ret = wiimote_cmd_write(wdata, 0xa20001, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* write configuration */ + ret = wiimote_cmd_write(wdata, 0xa20001, config, sizeof(config)); + if (ret) + goto out_unlock; + + /* enable speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20008, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* unmute speaker after setup if not muted */ + spin_lock_irqsave(&wdata->state.lock, flags); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock_irqrestore(&wdata->state.lock, flags); + +out_unlock: + wiimote_cmd_release(wdata); + return ret; +} + +/* PCM layer */ + +static const struct snd_pcm_hardware wiimod_speaker_playback_hw = { + .info = SNDRV_PCM_INFO_NONINTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 1500, + .rate_max = 4000, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = 20, + .period_bytes_min = 20, + .period_bytes_max = 20, + .periods_min = 1, + .periods_max = 1, +}; + +static int wiimod_speaker_playback_open(struct snd_pcm_substream *substream) +{ + struct wiimote_data *wdata = snd_pcm_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = wiimod_speaker_playback_hw; + runtime->private_data = wdata; + + return wiimod_speaker_enable(wdata); +} + +static int wiimod_speaker_playback_close(struct snd_pcm_substream *substream) +{ + struct wiimote_data *wdata = snd_pcm_chip(substream); + + wiimod_speaker_disable(wdata); + + return 0; +} + +static int wiimod_speaker_playback_hw_params(struct snd_pcm_substream *subs, + struct snd_pcm_hw_params *hw) +{ + /* TODO: anything to do here? */ + + return 0; +} + +static int wiimod_speaker_playback_hw_free(struct snd_pcm_substream *subs) +{ + return 0; +} + +static int wiimod_speaker_playback_prepare(struct snd_pcm_substream *subs) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + size_t buf_size, period_size, periods; + int ret; + + /* TODO: If I have set the hw-params to a fixed 20 bytes buffer, do + * I actually need to do any computations here? */ + buf_size = frames_to_bytes(runtime, runtime->buffer_size); + period_size = frames_to_bytes(runtime, runtime->period_size); + periods = runtime->periods; + + switch (runtime->format) { + case SNDRV_PCM_FMTBIT_S8: + ret = wiimod_speaker_setup(wdata, WIIMOD_SPEAKER_MODE_PCM8, + runtime->rate, -1); + break; + default: + return -EINVAL; + } + + return ret; +} + +static int wiimod_speaker_playback_trigger(struct snd_pcm_substream *subs, + int cmd) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + /* unmute device on start if not muted by user-space */ + spin_lock_irqsave(&wdata->state.lock, flags); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock_irqrestore(&wdata->state.lock, flags); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + /* mute device when stopping transmission */ + spin_lock_irqsave(&wdata->state.lock, flags); + wiiproto_req_mute(wdata, true); + spin_unlock_irqrestore(&wdata->state.lock, flags); + break; + } + + return 0; +} + +static snd_pcm_uframes_t wiimod_speaker_playback_pointer(struct snd_pcm_substream *subs) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + unsigned long pos; + + /* There is no way to read the current position. Even if there was a + * way to do that, it would be so slow that it would be useless. + * TODO: Should we interpolate via a timer? Is this function actually + * needed? When is it called? */ + pos = 0; + + pos %= 20; + + return bytes_to_frames(subs->runtime, pos); +} + +static int wiimod_speaker_playback_copy(struct snd_pcm_substream *subs, + int channel, snd_pcm_uframes_t pos, + void __user *buf, + snd_pcm_uframes_t count) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + + count = frames_to_bytes(runtime, count); + pos = frames_to_bytes(runtime, pos); + + /* TODO: copy from "buf" "count" bytes to "protobuf + pos" + * With the given hw-params, can pos be non-zero? Can count != 20? + * If not, should I simply send a 20byte frame to the device here? */ + + return 0; +} + +static int wiimod_speaker_playback_silence(struct snd_pcm_substream *subs, + int channel, snd_pcm_uframes_t pos, + snd_pcm_uframes_t count) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + + count = frames_to_bytes(runtime, count); + pos = frames_to_bytes(runtime, pos); + + /* TODO: set "count" bytes of "protobuf + pos" to zero + * Can "pos" actually be non-zero with the given hw-params? If not, + * can I simply send a 20byte zeroed frame to the remote device? */ + + return 0; +} + +/* TODO: is there a reason this cannot be "const"? */ +static struct snd_pcm_ops wiimod_speaker_playback_ops = { + .open = wiimod_speaker_playback_open, + .close = wiimod_speaker_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = wiimod_speaker_playback_hw_params, + .hw_free = wiimod_speaker_playback_hw_free, + .prepare = wiimod_speaker_playback_prepare, + .trigger = wiimod_speaker_playback_trigger, + .pointer = wiimod_speaker_playback_pointer, + .copy = wiimod_speaker_playback_copy, + .silence = wiimod_speaker_playback_silence, +}; + +/* volume control */ + +static int wiimod_speaker_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 1; + info->value.integer.min = 0; + info->value.integer.max = 0xff; + + return 0; +} + +static int wiimod_speaker_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + val->value.integer.value[0] = speaker->volume; + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static int wiimod_speaker_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + unsigned long value; + + value = val->value.integer.value[0]; + if (value > 0xff) + value = 0xff; + + wiimod_speaker_set_volume(wdata, value); + + return 0; +} + +static const struct snd_kcontrol_new wiimod_speaker_volume = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = wiimod_speaker_volume_info, + .get = wiimod_speaker_volume_get, + .put = wiimod_speaker_volume_put, +}; + +/* mute control */ + +static int wiimod_speaker_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + val->value.integer.value[0] = !!speaker->mute; + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static int wiimod_speaker_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + + wiimod_speaker_set_mute(wdata, val->value.integer.value[0]); + + return 0; +} + +/* TODO: Is *_IFACE_CARD the right interface? */ +static const struct snd_kcontrol_new wiimod_speaker_mute = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "PCM Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = wiimod_speaker_mute_get, + .put = wiimod_speaker_mute_put, +}; + +/* initialization and setup */ + +static int wiimod_speaker_probe(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + int ret; + struct wiimote_speaker *speaker; + struct snd_card *card; + struct snd_kcontrol *kcontrol; + struct snd_pcm *pcm; + + /* create sound card device */ + ret = snd_card_create(-1, NULL, THIS_MODULE, + sizeof(struct wiimote_speaker), &card); + if (ret) + return ret; + speaker = card->private_data; + + wdata->speaker = speaker; + speaker->card = card; + speaker->mute = 1; + speaker->volume = 0xff; + strcpy(card->driver, "hid-wiimote"); + strcpy(card->shortname, "wiimote"); + strcpy(card->longname, "Nintendo Wii Remote speaker"); + + /* create volume control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_volume, wdata); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create mute control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_mute, wdata); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create PCM sub-device for playback */ + ret = snd_pcm_new(card, "Speaker", 0, 1, 0, &pcm); + if (ret) + goto err_free; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &wiimod_speaker_playback_ops); + pcm->private_data = wdata; + + /* register sound card */ + snd_card_set_dev(card, &wdata->hdev->dev); + ret = snd_card_register(card); + if (ret) + goto err_free; + + return 0; + +err_free: + snd_card_free(card); + wdata->speaker = NULL; + return ret; +} + +static void wiimod_speaker_remove(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + + if (!speaker) + return; + + snd_card_free_when_closed(speaker->card); + wdata->speaker = NULL; +} + +const struct wiimod_ops wiimod_speaker = { + .flags = 0, + .arg = 0, + .probe = wiimod_speaker_probe, + .remove = wiimod_speaker_remove, +}; diff --git a/drivers/hid/hid-wiimote.h b/drivers/hid/hid-wiimote.h index 30caef5..4907bc7 100644 --- a/drivers/hid/hid-wiimote.h +++ b/drivers/hid/hid-wiimote.h @@ -148,6 +148,7 @@ struct wiimote_data { struct timer_list timer; struct wiimote_ext *ext; struct wiimote_debug *debug; + struct wiimote_speaker *speaker; union { struct input_dev *input; @@ -172,6 +173,7 @@ enum wiimod_module { WIIMOD_IR, WIIMOD_BUILTIN_MP, WIIMOD_NO_MP, + WIIMOD_SPEAKER, WIIMOD_NUM, WIIMOD_NULL = WIIMOD_NUM, }; @@ -208,9 +210,12 @@ enum wiiproto_reqs { WIIPROTO_REQ_LED = 0x11, WIIPROTO_REQ_DRM = 0x12, WIIPROTO_REQ_IR1 = 0x13, + WIIPROTO_REQ_SPEAKER = 0x14, WIIPROTO_REQ_SREQ = 0x15, WIIPROTO_REQ_WMEM = 0x16, WIIPROTO_REQ_RMEM = 0x17, + WIIPROTO_REQ_AUDIO = 0x18, + WIIPROTO_REQ_MUTE = 0x19, WIIPROTO_REQ_IR2 = 0x1a, WIIPROTO_REQ_STATUS = 0x20, WIIPROTO_REQ_DATA = 0x21, @@ -241,6 +246,12 @@ extern void wiiproto_req_status(struct wiimote_data *wdata); extern void wiiproto_req_accel(struct wiimote_data *wdata, __u8 accel); extern void wiiproto_req_ir1(struct wiimote_data *wdata, __u8 flags); extern void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags); +extern void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size); +extern void wiiproto_req_speaker(struct wiimote_data *wdata, bool on); +extern void wiiproto_req_mute(struct wiimote_data *wdata, bool on); +extern int wiiproto_req_audio_user(struct wiimote_data *wdata, + void __user *buf, size_t len); extern int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size); extern ssize_t wiimote_cmd_read(struct wiimote_data *wdata, __u32 offset, @@ -283,6 +294,12 @@ static inline void wiidebug_deinit(void *u) { } #endif +#ifdef CONFIG_SND + +extern const struct wiimod_ops wiimod_speaker; + +#endif + /* requires the state.lock spinlock to be held */ static inline bool wiimote_cmd_pending(struct wiimote_data *wdata, int cmd, __u32 opt) -- 1.8.2.1

2 6

Re: [alsa-devel] [PATCH] ASoC: fix kernel message grepability
by Mark Brown 05 May '13

05 May '13

On Sat, May 04, 2013 at 10:21:38PM +0200, Michał Mirosław wrote: > Signed-off-by: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: Convert to use regmap directly
by Fabio Estevam 05 May '13

05 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 145 +++++++++++++++++++++++++++++++++---------- 1 file changed, 111 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec..96b65b1 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -16,6 +16,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/clk.h> +#include <linux/regmap.h> #include <linux/regulator/driver.h> #include <linux/regulator/machine.h> #include <linux/regulator/consumer.h> @@ -34,30 +35,30 @@ #define SGTL5000_MAX_REG_OFFSET 0x013A /* default value of sgtl5000 registers */ -static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] = { - [SGTL5000_CHIP_CLK_CTRL] = 0x0008, - [SGTL5000_CHIP_I2S_CTRL] = 0x0010, - [SGTL5000_CHIP_SSS_CTRL] = 0x0008, - [SGTL5000_CHIP_DAC_VOL] = 0x3c3c, - [SGTL5000_CHIP_PAD_STRENGTH] = 0x015f, - [SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818, - [SGTL5000_CHIP_ANA_CTRL] = 0x0111, - [SGTL5000_CHIP_LINE_OUT_VOL] = 0x0404, - [SGTL5000_CHIP_ANA_POWER] = 0x7060, - [SGTL5000_CHIP_PLL_CTRL] = 0x5000, - [SGTL5000_DAP_BASS_ENHANCE] = 0x0040, - [SGTL5000_DAP_BASS_ENHANCE_CTRL] = 0x051f, - [SGTL5000_DAP_SURROUND] = 0x0040, - [SGTL5000_DAP_EQ_BASS_BAND0] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND1] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND2] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND3] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND4] = 0x002f, - [SGTL5000_DAP_MAIN_CHAN] = 0x8000, - [SGTL5000_DAP_AVC_CTRL] = 0x0510, - [SGTL5000_DAP_AVC_THRESHOLD] = 0x1473, - [SGTL5000_DAP_AVC_ATTACK] = 0x0028, - [SGTL5000_DAP_AVC_DECAY] = 0x0050, +static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, + { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, + { SGTL5000_CHIP_SSS_CTRL, 0x0008 }, + { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, + { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, + { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, + { SGTL5000_CHIP_ANA_POWER, 0x7060 }, + { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, + { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_SURROUND, 0x0040 }, + { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND2, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, + { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, + { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, + { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, + { SGTL5000_DAP_AVC_DECAY, 0x0050 }, }; /* regulator supplies for sgtl5000, VDDD is an optional external supply */ @@ -112,6 +113,7 @@ struct sgtl5000_priv { int fmt; /* i2s data format */ struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; struct ldo_regulator *ldo; + struct regmap *regmap; }; /* @@ -958,17 +960,76 @@ static struct snd_soc_dai_driver sgtl5000_dai = { .symmetric_rates = 1, }; -static int sgtl5000_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool sgtl5000_volatile(struct device *dev, unsigned int reg) { switch (reg) { case SGTL5000_CHIP_ID: case SGTL5000_CHIP_ADCDAC_CTRL: case SGTL5000_CHIP_ANA_STATUS: - return 1; + return true; } - return 0; + return false; +} + +static bool sgtl5000_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SGTL5000_CHIP_ID: + case SGTL5000_CHIP_DIG_POWER: + case SGTL5000_CHIP_CLK_CTRL: + case SGTL5000_CHIP_I2S_CTRL: + case SGTL5000_CHIP_SSS_CTRL: + case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_DAC_VOL: + case SGTL5000_CHIP_PAD_STRENGTH: + case SGTL5000_CHIP_ANA_ADC_CTRL: + case SGTL5000_CHIP_ANA_HP_CTRL: + case SGTL5000_CHIP_ANA_CTRL: + case SGTL5000_CHIP_LINREG_CTRL: + case SGTL5000_CHIP_REF_CTRL: + case SGTL5000_CHIP_MIC_CTRL: + case SGTL5000_CHIP_LINE_OUT_CTRL: + case SGTL5000_CHIP_LINE_OUT_VOL: + case SGTL5000_CHIP_ANA_POWER: + case SGTL5000_CHIP_PLL_CTRL: + case SGTL5000_CHIP_CLK_TOP_CTRL: + case SGTL5000_CHIP_ANA_STATUS: + case SGTL5000_CHIP_SHORT_CTRL: + case SGTL5000_CHIP_ANA_TEST2: + case SGTL5000_DAP_CTRL: + case SGTL5000_DAP_PEQ: + case SGTL5000_DAP_BASS_ENHANCE: + case SGTL5000_DAP_BASS_ENHANCE_CTRL: + case SGTL5000_DAP_AUDIO_EQ: + case SGTL5000_DAP_SURROUND: + case SGTL5000_DAP_FLT_COEF_ACCESS: + case SGTL5000_DAP_COEF_WR_B0_MSB: + case SGTL5000_DAP_COEF_WR_B0_LSB: + case SGTL5000_DAP_EQ_BASS_BAND0: + case SGTL5000_DAP_EQ_BASS_BAND1: + case SGTL5000_DAP_EQ_BASS_BAND2: + case SGTL5000_DAP_EQ_BASS_BAND3: + case SGTL5000_DAP_EQ_BASS_BAND4: + case SGTL5000_DAP_MAIN_CHAN: + case SGTL5000_DAP_MIX_CHAN: + case SGTL5000_DAP_AVC_CTRL: + case SGTL5000_DAP_AVC_THRESHOLD: + case SGTL5000_DAP_AVC_ATTACK: + case SGTL5000_DAP_AVC_DECAY: + case SGTL5000_DAP_COEF_WR_B1_MSB: + case SGTL5000_DAP_COEF_WR_B1_LSB: + case SGTL5000_DAP_COEF_WR_B2_MSB: + case SGTL5000_DAP_COEF_WR_B2_LSB: + case SGTL5000_DAP_COEF_WR_A1_MSB: + case SGTL5000_DAP_COEF_WR_A1_LSB: + case SGTL5000_DAP_COEF_WR_A2_MSB: + case SGTL5000_DAP_COEF_WR_A2_LSB: + return true; + + default: + return false; + } } #ifdef CONFIG_SUSPEND @@ -1300,7 +1361,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); /* setup i2c data ops */ - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + codec->control_data = sgtl5000->regmap; + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1391,11 +1453,6 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .suspend = sgtl5000_suspend, .resume = sgtl5000_resume, .set_bias_level = sgtl5000_set_bias_level, - .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = sgtl5000_regs, - .volatile_register = sgtl5000_volatile_register, .controls = sgtl5000_snd_controls, .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), .dapm_widgets = sgtl5000_dapm_widgets, @@ -1404,6 +1461,19 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .num_dapm_routes = ARRAY_SIZE(sgtl5000_dapm_routes), }; +static const struct regmap_config sgtl5000_regmap = { + .reg_bits = 16, + .val_bits = 16, + + .max_register = SGTL5000_MAX_REG_OFFSET, + .volatile_reg = sgtl5000_volatile, + .readable_reg = sgtl5000_readable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = sgtl5000_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(sgtl5000_reg_defaults), +}; + static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -1415,6 +1485,13 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (!sgtl5000) return -ENOMEM; + sgtl5000->regmap = devm_regmap_init_i2c(client, &sgtl5000_regmap); + if (IS_ERR(sgtl5000->regmap)) { + ret = PTR_ERR(sgtl5000->regmap); + dev_err(&client->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + i2c_set_clientdata(client, sgtl5000); ret = snd_soc_register_codec(&client->dev, -- 1.7.9.5

2 1

Re: [alsa-devel] [RFC] ASoC: sgtl5000: Remove cache support
by Fabio Estevam 05 May '13

05 May '13

Hi Mark, On Fri, May 3, 2013 at 7:36 PM, Mark Brown <broonie(a)sirena.org.uk> wrote: > On Fri, May 03, 2013 at 04:49:09PM -0300, Fabio Estevam wrote: > >> .set_bias_level = sgtl5000_set_bias_level, >> - .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), >> - .reg_word_size = sizeof(u16), >> - .reg_cache_step = 2, >> - .reg_cache_default = sgtl5000_regs, > > So, the most obvious thing here is to convert the driver to regmap. It > has a much more baked in idea of sparse register maps which should help > if nothing else, and will allow the device initialisation to be moved to > the normal device probe which is better practice. Thanks for your suggestion. I have just sent a patch converting sgtl5000 to use regmap. Unfortunately the result is the same: the codec cannot probe after a reset. > Otherwise the normal issue with this sort of thing is a failure to power > down the device properly, the LDO workaround looks suspicous here as > does the fact that the driver doesn't restore the register cache when > powering on the device in resume (nor does it makr the device as cache > only). I will try to investigate the power down and LDO workaround as suggested. Thanks, Fabio Estevam

2 2

Re: [alsa-devel] Conexant CX20588 breakage in Kernel 3.9
by Takashi Iwai 04 May '13

04 May '13

At Sat, 4 May 2013 00:56:10 +0200, Christopher K. wrote: > > Works great, thanks for the help! OK, now applied. Thanks for a quick test. Takashi > > > On Fri, May 3, 2013 at 4:58 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Fri, 3 May 2013 15:51:20 +0200, > > Christopher K. wrote: > > > > > > Hi, > > > > > > Kernel 3.9 breaks audio on internal speakers for the Conexant CX20588. > > > EAPD is disabled for Node 0x1d. Enabling in hda_analyzer works fine. > > > > > > Someone already postet on > > > https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1173618 > > > > > > Is this the right place to report this bug? > > > > The kernel bugzilla or in alsa-devel ML, at best with Cc to > > maintainer. > > > > Does the patch below fix your problem? If so, I'll queue it to the > > upstream and Cc to stable kernel. > > > > > > thanks, > > > > Takashi > > > > --- > > diff --git a/sound/pci/hda/patch_conexant.c > > b/sound/pci/hda/patch_conexant.c > > index 84b81c8..e08cd33 100644 > > --- a/sound/pci/hda/patch_conexant.c > > +++ b/sound/pci/hda/patch_conexant.c > > @@ -63,6 +63,7 @@ struct conexant_spec { > > > > /* extra EAPD pins */ > > unsigned int num_eapds; > > + bool dynamic_eapd; > > hda_nid_t eapds[4]; > > > > #ifdef ENABLE_CXT_STATIC_QUIRKS > > @@ -3155,7 +3156,7 @@ static void cx_auto_parse_eapd(struct hda_codec > > *codec) > > * thus it might control over all pins. > > */ > > if (spec->num_eapds > 2) > > - spec->gen.own_eapd_ctl = 1; > > + spec->dynamic_eapd = 1; > > } > > > > static void cx_auto_turn_eapd(struct hda_codec *codec, int num_pins, > > @@ -3194,10 +3195,19 @@ static int cx_auto_build_controls(struct hda_codec > > *codec) > > return 0; > > } > > > > +static int cx_auto_init(struct hda_codec *codec) > > +{ > > + struct conexant_spec *spec = codec->spec; > > + snd_hda_gen_init(codec); > > + if (!spec->dynamic_eapd) > > + cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, > > true); > > + return 0; > > +} > > + > > static const struct hda_codec_ops cx_auto_patch_ops = { > > .build_controls = cx_auto_build_controls, > > .build_pcms = snd_hda_gen_build_pcms, > > - .init = snd_hda_gen_init, > > + .init = cx_auto_init, > > .free = snd_hda_gen_free, > > .unsol_event = snd_hda_jack_unsol_event, > > #ifdef CONFIG_PM > > @@ -3348,7 +3358,8 @@ static int patch_conexant_auto(struct hda_codec > > *codec) > > > > cx_auto_parse_beep(codec); > > cx_auto_parse_eapd(codec); > > - if (spec->gen.own_eapd_ctl) > > + spec->gen.own_eapd_ctl = 1; > > + if (spec->dynamic_eapd) > > spec->gen.vmaster_mute.hook = cx_auto_vmaster_hook; > > > > switch (codec->vendor_id) { > > > [2 <text/html; UTF-8 (quoted-printable)>] >

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[alsa-devel] [PATCH] ASoC: sgtl5000: Fix comment about register addresses
by Fabio Estevam 04 May '13

04 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> The list below the comment relates to sgtl5000 registers addresses, so change the comment to improve the description. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 8a9f435..4b69229 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -12,7 +12,7 @@ #define _SGTL5000_H /* - * Register values. + * Registers addresses */ #define SGTL5000_CHIP_ID 0x0000 #define SGTL5000_CHIP_DIG_POWER 0x0002 -- 1.7.9.5

2 1

[alsa-devel] [RFC] ASoC: sgtl5000: Remove cache support
by Fabio Estevam 03 May '13

03 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently sgtl5000 driver is probed correctly after a power-on reset, but after audio is played and the board is reset, we are no longer able to probe again: sgtl5000 0-000a: Device with ID register ffff is not a sgtl5000 sgtl5000 0-000a: ASoC: failed to probe CODEC -19 imx-sgtl5000 sound.12: ASoC: failed to instantiate card -19 imx-sgtl5000 sound.12: snd_soc_register_card failed (-19) Removing the cache support mechanism allows us to probe sgtl5000 driver successfully in subsequent board resets. Tested on a mx6qsabrelite board. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- I would like to fix it properly and looking for some ideas/feedbacks as to what a proper fix would be. Should I complete the sgtl5000_regs array with all the sgtl5000 registers? Thanks! sound/soc/codecs/sgtl5000.c | 31 ------------------------------- 1 file changed, 31 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec..3884162 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -33,33 +33,6 @@ #define SGTL5000_DAP_REG_OFFSET 0x0100 #define SGTL5000_MAX_REG_OFFSET 0x013A -/* default value of sgtl5000 registers */ -static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] = { - [SGTL5000_CHIP_CLK_CTRL] = 0x0008, - [SGTL5000_CHIP_I2S_CTRL] = 0x0010, - [SGTL5000_CHIP_SSS_CTRL] = 0x0008, - [SGTL5000_CHIP_DAC_VOL] = 0x3c3c, - [SGTL5000_CHIP_PAD_STRENGTH] = 0x015f, - [SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818, - [SGTL5000_CHIP_ANA_CTRL] = 0x0111, - [SGTL5000_CHIP_LINE_OUT_VOL] = 0x0404, - [SGTL5000_CHIP_ANA_POWER] = 0x7060, - [SGTL5000_CHIP_PLL_CTRL] = 0x5000, - [SGTL5000_DAP_BASS_ENHANCE] = 0x0040, - [SGTL5000_DAP_BASS_ENHANCE_CTRL] = 0x051f, - [SGTL5000_DAP_SURROUND] = 0x0040, - [SGTL5000_DAP_EQ_BASS_BAND0] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND1] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND2] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND3] = 0x002f, - [SGTL5000_DAP_EQ_BASS_BAND4] = 0x002f, - [SGTL5000_DAP_MAIN_CHAN] = 0x8000, - [SGTL5000_DAP_AVC_CTRL] = 0x0510, - [SGTL5000_DAP_AVC_THRESHOLD] = 0x1473, - [SGTL5000_DAP_AVC_ATTACK] = 0x0028, - [SGTL5000_DAP_AVC_DECAY] = 0x0050, -}; - /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { VDDA, @@ -1391,10 +1364,6 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .suspend = sgtl5000_suspend, .resume = sgtl5000_resume, .set_bias_level = sgtl5000_set_bias_level, - .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = sgtl5000_regs, .volatile_register = sgtl5000_volatile_register, .controls = sgtl5000_snd_controls, .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), -- 1.7.9.5

1 0

[alsa-devel] [PATCH] drm/i915: Add private api for power well usage -- alignment between graphic team and audio team
by Li, Jocelyn 03 May '13

03 May '13

Hi Guys, I apologize that I may have misunderstood on the agreement we have made on power well work. I assumed that audio team is expected to create private gfx driver API to control the power well between gfx and audio for internal releases. That's why I asked Xingchao work on that. So the correct implementation approach between gfx and audio should be, #1, Daniel will work on the existing private API in gfx driver for power well control that can be used by the audio driver to enable/disable the power well. #2, Once #1 is done, Audio team will integrate Daniel's patch into audio driver and add/improve on anything that is missing Please correct me if my understanding is not correct. Thanks, Jocelyn -----Original Message----- From: Daniel Vetter [mailto:daniel.vetter@ffwll.ch] Sent: Thursday, April 25, 2013 3:52 PM To: Wang, Xingchao Cc: Zanoni, Paulo R; ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; alsa-devel(a)alsa-project.org; Wang Xingchao; Takashi Iwai; Barnes, Jesse; Wysocki, Rafael J Subject: Re: [PATCH] drm/i915: Add private api for power well usage On Thu, Apr 25, 2013 at 4:36 AM, Wang, Xingchao <xingchao.wang(a)intel.com> wrote: > Hi Daniel/Paulo, > > Any comments on this? > Add Jesse and Rafael in loop. Well I've figured we want to get the power well integration working before we waste time reviewing an interface which might or might not quite work. But if you want, I can drop two quick comments: - I think the functions needs some prefix like i915_ - Imo the power well should just be reference counted (the current interface name suggests it's generally useable, but actually can only cope with one request from the audio driver). - Is the spin_lock (irq-disabling even) really required for the audio driver to work? power well enabling can take up to 20ms, so this won't work as-is. Can't we just use a mutex? - I'd need to see the audio driver side of this to check how this works for coping with arbitrary module load ordering. Cheers, Daniel > > thanks > --xingchao > > >> -----Original Message----- >> From: Wang, Xingchao >> Sent: Wednesday, April 24, 2013 3:29 PM >> To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R; 'Takashi Iwai' >> Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; >> Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; alsa-devel(a)alsa-project.org; 'Wang Xingchao' >> Subject: RE: [PATCH] drm/i915: Add private api for power well usage >> >> Hi Guys, >> >> Sorry I should Add Takashi and alsa maillist in loop. >> >> I tested this patch against Daniel's "drm-intel-next-queued" Branch, >> on Haswell Mobile C3 stepping machines. >> >> Test steps: >> 1. only connect one monitor through HDMI cable 2. echo 1 > >> /sys/module/i915/parameters/disable_power_well ==> I think this could >> be set as 1 by default if applied my patch. >> 3. plug out HDMI cable >> 4. cat /proc/asound/card0/codec#0 >> >> At step 3, i915 will try to disable power well but rejected as >> display audio is using power well. >> Without the patch, display audio would crash at step 4. >> >> I did not list suspend/resume test here, as during suspend sequence, >> i915 will _ENABLE_ power well, that's a BUG? >> >> For me above test case is the only one to reproduce the issue, if you >> have more ideas and need more test, please let me know. >> >> I'm afraid there's some risk because the dependency between alsa >> driver and drm driver. For example, if i915 module was not loaded, >> display audio Will call >> request_power_well() and meet symbol error. Even if i915 module >> loaded after snd-hda-intel module, there's such risk too. Any comment on this? >> >> Another risk is the pointer "power_well_device", it should be set >> NULL when >> i915 module unloaded, I added the caller but have no idea where to >> call it, if anyone has suggestion, that would be fine. >> >> thanks >> --xingchao >> >> >> > -----Original Message----- >> > From: Wang Xingchao [mailto:xingchao.wang@linux.intel.com] >> > Sent: Thursday, April 24, 2014 10:20 PM >> > To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R >> > Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; >> > Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; Wang, Xingchao; Wang >> > Xingchao >> > Subject: [PATCH] drm/i915: Add private api for power well usage >> > >> > Display audio depends on power well in graphic side, it should >> > request power well before use it and release power well after use. >> > I915 will not shutdown power well if it detects audio is using. >> > This patch protects display audio crash for Intel Haswell mobile >> > C3 stepping board. >> > >> > Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> >> > --- >> > drivers/gpu/drm/i915/intel_pm.c | 73 >> > +++++++++++++++++++++++++++++++++++---- >> > include/drm/audio_drm.h | 39 +++++++++++++++++++++ >> > sound/pci/hda/hda_intel.c | 8 +++++ >> > 3 files changed, 113 insertions(+), 7 deletions(-) create mode >> > 100644 include/drm/audio_drm.h >> > >> > diff --git a/drivers/gpu/drm/i915/intel_pm.c >> > b/drivers/gpu/drm/i915/intel_pm.c index 2557926..9983f56 100644 >> > --- a/drivers/gpu/drm/i915/intel_pm.c >> > +++ b/drivers/gpu/drm/i915/intel_pm.c >> > @@ -4298,18 +4298,12 @@ bool intel_using_power_well(struct >> > drm_device >> > *dev) >> > return true; >> > } >> > >> > -void intel_set_power_well(struct drm_device *dev, bool enable) >> > +void set_power_well(struct drm_device *dev, bool enable) >> > { >> > struct drm_i915_private *dev_priv = dev->dev_private; >> > bool is_enabled, enable_requested; >> > uint32_t tmp; >> > >> > - if (!HAS_POWER_WELL(dev)) >> > - return; >> > - >> > - if (!i915_disable_power_well && !enable) >> > - return; >> > - >> > tmp = I915_READ(HSW_PWR_WELL_DRIVER); >> > is_enabled = tmp & HSW_PWR_WELL_STATE; >> > enable_requested = tmp & HSW_PWR_WELL_ENABLE; @@ -4332,6 >> > +4326,71 @@ void intel_set_power_well(struct drm_device *dev, bool >> > +enable) >> > } >> > } >> > >> > +/* Global drm_device for display audio drvier usage */ static >> > +struct drm_device *power_well_device; static bool >> > +audio_power_well_using, i915_power_well_using; >> > +/* Lock protecting power well setting */ >> > +DEFINE_SPINLOCK(powerwell_lock); >> > + >> > +void request_power_well(void) >> > +{ >> > + DRM_DEBUG_KMS("Display audio requesting power well\n"); >> > + spin_lock_irq(&powerwell_lock); >> > + if (!power_well_device) >> > + goto out; >> > + if (i915_power_well_using == false) >> > + set_power_well(power_well_device, true); >> > + audio_power_well_using = true; >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > +EXPORT_SYMBOL_GPL(request_power_well); >> > + >> > +void release_power_well(void) >> > +{ >> > + DRM_DEBUG_KMS("Display audio releasing power well\n"); >> > + spin_lock_irq(&powerwell_lock); >> > + if (!power_well_device) >> > + goto out; >> > + if (i915_power_well_using == false) >> > + set_power_well(power_well_device, false); >> > + audio_power_well_using = false; >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > +EXPORT_SYMBOL_GPL(release_power_well); >> > + >> > +/* TODO: Call this when i915 module unload */ void >> > +remove_power_well(void) { >> > + spin_lock_irq(&powerwell_lock); >> > + audio_power_well_using = false; >> > + i915_power_well_using = false; >> > + power_well_device = NULL; >> > + spin_unlock_irq(&powerwell_lock); } >> > + >> > +void intel_set_power_well(struct drm_device *dev, bool enable) { >> > + if (!HAS_POWER_WELL(dev)) >> > + return; >> > + >> > + spin_lock_irq(&powerwell_lock); >> > + power_well_device = dev; >> > + i915_power_well_using = enable; >> > + if (!enable && audio_power_well_using) >> > + goto out; >> > + >> > + if (!i915_disable_power_well && !enable) >> > + goto out; >> > + set_power_well(dev, enable); >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > + >> > /* >> > * Starting with Haswell, we have a "Power Down Well" that can be >> > turned >> off >> > * when not needed anymore. We have 4 registers that can request >> > the power well diff --git a/include/drm/audio_drm.h >> > b/include/drm/audio_drm.h new file mode 100644 index >> > 0000000..4412b36 >> > --- /dev/null >> > +++ b/include/drm/audio_drm.h >> > @@ -0,0 +1,39 @@ >> > >> +/************************************************************** >> > ******** >> > +**** >> > + * >> > + * Copyright 2013 Intel Inc. >> > + * All Rights Reserved. >> > + * >> > + * Permission is hereby granted, free of charge, to any person >> > +obtaining a >> > + * copy of this software and associated documentation files (the >> > + * "Software"), to deal in the Software without restriction, >> > +including >> > + * without limitation the rights to use, copy, modify, merge, >> > +publish, >> > + * distribute, sub license, and/or sell copies of the Software, >> > +and to >> > + * permit persons to whom the Software is furnished to do so, >> > +subject to >> > + * the following conditions: >> > + * >> > + * The above copyright notice and this permission notice >> > +(including the >> > + * next paragraph) shall be included in all copies or substantial >> > +portions >> > + * of the Software. >> > + * >> > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY >> KIND, >> > +EXPRESS OR >> > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF >> > +MERCHANTABILITY, >> > + * FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT. IN NO >> > EVENT >> > +SHALL >> > + * THE COPYRIGHT HOLDERS, AUTHORS AND/OR ITS SUPPLIERS BE LIABLE >> > FOR >> > +ANY CLAIM, >> > + * DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, >> > TORT >> > +OR >> > + * OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE >> > SOFTWARE >> > +OR THE >> > + * USE OR OTHER DEALINGS IN THE SOFTWARE. >> > + * >> > + * >> > + >> > >> +*************************************************************** >> > ******** >> > +***/ >> > +/* >> > + * Authors: >> > + * Wang Xingchao <xingchao.wang(a)intel.com> */ >> > + >> > +#ifndef _AUDIO_DRM_H_ >> > +#define _AUDIO_DRM_H_ >> > + >> > +extern void request_power_well(void); extern void >> > +release_power_well(void); >> > + >> > +#endif /* _AUDIO_DRM_H_ */ >> > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c >> > index 418bfc0..59a85de 100644 >> > --- a/sound/pci/hda/hda_intel.c >> > +++ b/sound/pci/hda/hda_intel.c >> > @@ -50,6 +50,7 @@ >> > #include <linux/clocksource.h> >> > #include <linux/time.h> >> > #include <linux/completion.h> >> > +#include <drm/audio_drm.h> >> > >> > #ifdef CONFIG_X86 >> > /* for snoop control */ >> > @@ -2869,6 +2870,7 @@ static int azx_suspend(struct device *dev) >> > pci_disable_device(pci); >> > pci_save_state(pci); >> > pci_set_power_state(pci, PCI_D3hot); >> > + release_power_well(); >> > return 0; >> > } >> > >> > @@ -2881,6 +2883,7 @@ static int azx_resume(struct device *dev) >> > if (chip->disabled) >> > return 0; >> > >> > + request_power_well(); >> > pci_set_power_state(pci, PCI_D0); >> > pci_restore_state(pci); >> > if (pci_enable_device(pci) < 0) { @@ -2913,6 +2916,7 @@ static >> > int azx_runtime_suspend(struct device >> > *dev) >> > >> > azx_stop_chip(chip); >> > azx_clear_irq_pending(chip); >> > + release_power_well(); >> > return 0; >> > } >> > >> > @@ -2921,6 +2925,7 @@ static int azx_runtime_resume(struct device *dev) >> > struct snd_card *card = dev_get_drvdata(dev); >> > struct azx *chip = card->private_data; >> > >> > + request_power_well(); >> > azx_init_pci(chip); >> > azx_init_chip(chip, 1); >> > return 0; >> > @@ -3671,6 +3676,8 @@ static int azx_probe(struct pci_dev *pci, >> > return -ENOENT; >> > } >> > >> > + request_power_well(); >> > + >> > err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); >> > if (err < 0) { >> > snd_printk(KERN_ERR "hda-intel: Error creating >> > card!\n"); @@ >> > -3806,6 +3813,7 @@ static void azx_remove(struct pci_dev *pci) >> > if (card) >> > snd_card_free(card); >> > pci_set_drvdata(pci, NULL); >> > + release_power_well(); >> > } >> > >> > /* PCI IDs */ >> > -- >> > 1.7.9.5 > -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

9 23

[alsa-devel] Conexant CX20588 breakage in Kernel 3.9
by Christopher K. 03 May '13

03 May '13

Hi, Kernel 3.9 breaks audio on internal speakers for the Conexant CX20588. EAPD is disabled for Node 0x1d. Enabling in hda_analyzer works fine. Someone already postet on https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1173618 Is this the right place to report this bug? Regards, Chris

2 1

[alsa-devel] Audio miniconference (probably) moved to Edinburgh
by David Henningsson 03 May '13

03 May '13

Just to keep the wider crowd informed: We're currently discussing that we won't have any Audio miniconference on LPC (Linux Plumber's Conference) this year. Instead, we'll have an audio miniconference day in Edinburgh in October, colocated with LinuxCon Europe, Kernel Summit, ELCE and GStreamer conferences. The exact day for this miniconference has not been decided yet. The main reason for this move is proximity, as several participants are located in northern Europe. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

1 0

[alsa-devel] [PATCH]: x86_64, audio: Fix system panic when DMA > 40 bits
by Mike Travis 03 May '13

03 May '13

The audio driver mistakenly allows 64 bit addresses to be created for the audio driver on Nvidia GPUs. Unfortunately, the hardware normally only supports up to 40 bits of DMA. This can cause system panics as well as misdirected data when the address is > 40 bits as the upper part the address is truncated. Signed-off-by: Mike Travis <travis(a)sgi.com> Reviewed-by: Mike Habeck <habeck(a)sgi.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bcd40ee..45eb165 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -615,7 +615,7 @@ enum { /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\ - AZX_DCAPS_ALIGN_BUFSIZE) + AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_NO_64BIT) #define AZX_DCAPS_PRESET_CTHDA \ (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY) -- 1.8.2.1

4 7

[alsa-devel] ASoC updates for v3.10
by Mark Brown 03 May '13

03 May '13

The following changes since commit c1be5a5b1b355d40e6cf79cc979eb66dafa24ad1: Linux 3.9 (2013-04-28 17:36:01 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-4 for you to fetch changes up to 9631c02e240236ec206394fb52380647d0e70e4f: Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus (2013-05-03 09:30:29 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.10 A few more bug fixes, the DAPM clock fix is actually a driver specific one since currently there's only one user of the clock support due to the problems relying on the clock API. ---------------------------------------------------------------- Dan Carpenter (1): ASoC: wm8994: missing break in wm8994_aif3_hw_params() Fabio Baltieri (1): ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare Marek Belisko (2): ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode. Mark Brown (2): Merge remote-tracking branch 'asoc/fix/davinci' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus sound/soc/codecs/wm8994.c | 1 + sound/soc/davinci/davinci-mcasp.c | 7 +++++-- sound/soc/soc-dapm.c | 4 ++-- 3 files changed, 8 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH -next 0/3] mop500_ab8500 fixes
by Fabio Baltieri 03 May '13

03 May '13

Hi, these are small fixes to the mop500_ab8500 asoc driver, developed on top of today's ASoC-next. The first patch is actually non ab8500 specific, but as ab8500 seems to be the only codec to use the SND_SOC_DAPM_CLOCK_SUPPLY widget I'm posting it here anyway. The other two patches sort out a mixup of card and codec controls in mop500_ab8500. Thanks, Fabio Fabio Baltieri (3): ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare ASoC: ux500: move clock controls to ab8500-codec ASoC: ux500: register controls to card instead of codec sound/soc/codecs/ab8500-codec.c | 10 ++++++++++ sound/soc/soc-dapm.c | 4 ++-- sound/soc/ux500/mop500_ab8500.c | 20 +++++--------------- 3 files changed, 17 insertions(+), 17 deletions(-) -- 1.8.2

3 11

[alsa-devel] [PATCH] ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode.
by Marek Belisko 03 May '13

03 May '13

According documentation bit ACLKRPOL is set to 0 (receiver samples data on falling edge) and when set to 1 (receiver samples data on rising edge). I2S data are always sampled on falling edge and valid during rising edge of bit clock. So in case of capture data transmitter sample data on falling edge and macsp must read then on rising edge. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 849a187..c46112e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -568,7 +568,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); - mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); break; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ALSA: atmel: Remove redundant platform_set_drvdata()
by Sachin Kamat 03 May '13

03 May '13

Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/atmel/abdac.c | 2 -- sound/atmel/ac97c.c | 2 -- 2 files changed, 4 deletions(-) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 071ce1b..872d59e 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -583,8 +583,6 @@ static int atmel_abdac_remove(struct platform_device *pdev) free_irq(dac->irq, dac); snd_card_free(card); - platform_set_drvdata(pdev, NULL); - return 0; } diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 6b7e2b5..ae63d22 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1199,8 +1199,6 @@ static int atmel_ac97c_remove(struct platform_device *pdev) snd_card_set_dev(card, NULL); snd_card_free(card); - platform_set_drvdata(pdev, NULL); - return 0; } -- 1.7.9.5

4 3

[alsa-devel] [PATCH] ASoC: Added the condition for the channels_min of cpu_dai
by JongHo Kim 02 May '13

02 May '13

Only playback device, the pcm substream for capture does not need to be generated. If codec dai driver has both playback.channels_min and capture.channels_min, and cpu dai driver has only playback.channels_min, soc_new_pcm() will make the pcm substream for capture although DataIN is not connected. So added the condition for the channels_min of cpu_dai in soc_new_pcm(). Signed-off-by: JongHo Kim <furmuwon(a)gmail.com> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cf191e6..c405e39 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2018,9 +2018,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->capture.channels_min && + cpu_dai->driver->capture.channels_min) capture = 1; } -- 1.8.1.1

2 4

[alsa-devel] [PATCH] ASoC: max98090: Fix build warning when CONFIG_PM_RUNTIME is not set
by Axel Lin 02 May '13

02 May '13

Fix below build warning when CONFIG_PM_RUNTIME is not set. CC [M] sound/soc/codecs/max98090.o sound/soc/codecs/max98090.c:2339:12: warning: 'max98090_runtime_resume' defined but not used [-Wunused-function] sound/soc/codecs/max98090.c:2350:12: warning: 'max98090_runtime_suspend' defined but not used [-Wunused-function] Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98090.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ce0d364..4f39a50 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2336,6 +2336,7 @@ static int max98090_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME static int max98090_runtime_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); @@ -2355,6 +2356,7 @@ static int max98090_runtime_suspend(struct device *dev) return 0; } +#endif static const struct dev_pm_ops max98090_pm = { SET_RUNTIME_PM_OPS(max98090_runtime_suspend, -- 1.8.1.2

2 1

[alsa-devel] [PATCH v2 1/4] ASoC: imx-sgtl5000: Do not enter the error path on success
by Fabio Estevam 02 May '13

02 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Return on success instead of entering the error path. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Keep of_node_put in the success path. sound/soc/fsl/imx-sgtl5000.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 9584e78..5a6aaa3 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -174,6 +174,11 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + clk_fail: clk_put(data->codec_clk); fail: -- 1.7.9.5

3 12

[alsa-devel] Mono output on emu10k1 with alsa broken
by Thomas Richter 01 May '13

01 May '13

Trying to configure mono (one channel) output on emu10k1 hardware using the alsa version that comes with kernel 3.9 is broken. Instead of expecting mono samples, the alsa callback expects twice as many stereo samples. Not getting the required number of samples then causes a buffer underrun, and sound gets interrupted or messed up. In fact, what the audio hardware does is that it plays not only the buffer, but also the data "behind the buffer", causing wierd noises. The channel count is setup as channels = 1; snd_pcm_hw_params_malloc(&HWParms); snd_pcm_open(&SoundStream,CardName,SND_PCM_STREAM_PLAYBACK,SND_PCM_NONBLOCK | SND_PCM_ASYNC); snd_pcm_hw_params_set_channels_min(SoundStream,HWParms,&channels); snd_pcm_hw_params_set_channels_max(SoundStream,HWParms,&channels); snd_pcm_hw_params_set_channels_first(SoundStream,HWParms,&channels); plus error checking. Then, a bit later, the number of configured channels is verified: snd_pcm_hw_params_get_channels(HWParms,&channels); channels is at this point still 1, i.e. the driver accepted the mono output, but does then not play correctly. Note that the same call sequence works for stereo, and it also works for mono for other underlying hardware drivers, i.e. mono output works on snd-hda-intel, for example. It does not on emu10k1.

1 0

[alsa-devel] [PATCH] MAINTAINERS: Add linux-samsung-soc list to all related entries
by Tomasz Figa 01 May '13

01 May '13

Several entries in MAINTAINERS file related to Samsung SoCs do not point to linux-samsung-soc mailing list, which is supposed to collect all Samsung SoC-related threads, to ease following of Samsung SoC-related work. This leads to a problem with people skipping this mailing list in their posts, even though they are related to Samsung SoCs. This patch adds pointers to linux-samsung-soc mailing list to affected entries of MAINTAINERS file. Signed-off-by: Tomasz Figa <tomasz.figa(a)gmail.com> --- MAINTAINERS | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 6c0d68b..07cb8da 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1139,6 +1139,7 @@ F: arch/arm/mach-exynos*/ ARM/SAMSUNG MOBILE MACHINE SUPPORT M: Kyungmin Park <kyungmin.park(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/mach-s5pv210/mach-aquila.c F: arch/arm/mach-s5pv210/mach-goni.c @@ -1150,6 +1151,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Kamil Debski <k.debski(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/media/platform/s5p-g2d/ @@ -1158,6 +1160,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Sylwester Nawrocki <s.nawrocki(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/plat-samsung/include/plat/*fimc* F: drivers/media/platform/s5p-fimc/ @@ -1168,6 +1171,7 @@ M: Kamil Debski <k.debski(a)samsung.com> M: Jeongtae Park <jtp.park(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/plat-samsung/s5p-dev-mfc.c F: drivers/media/platform/s5p-mfc/ @@ -1177,6 +1181,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Tomasz Stanislawski <t.stanislaws(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/media/platform/s5p-tv/ @@ -2679,6 +2684,7 @@ M: Joonyoung Shim <jy0922.shim(a)samsung.com> M: Seung-Woo Kim <sw0312.kim(a)samsung.com> M: Kyungmin Park <kyungmin.park(a)samsung.com> L: dri-devel(a)lists.freedesktop.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/daeinki/drm-exynos.git S: Supported F: drivers/gpu/drm/exynos @@ -3142,6 +3148,7 @@ F: Documentation/extcon/ EXYNOS DP DRIVER M: Jingoo Han <jg1.han(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/exynos/exynos_dp* F: include/video/exynos_dp* @@ -3151,6 +3158,7 @@ M: Inki Dae <inki.dae(a)samsung.com> M: Donghwa Lee <dh09.lee(a)samsung.com> M: Kyungmin Park <kyungmin.park(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/exynos/exynos_mipi* F: include/video/exynos_mipi* @@ -6892,12 +6900,14 @@ F: drivers/platform/x86/samsung-laptop.c SAMSUNG AUDIO (ASoC) DRIVERS M: Sangbeom Kim <sbkim73(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Supported F: sound/soc/samsung SAMSUNG FRAMEBUFFER DRIVER M: Jingoo Han <jg1.han(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/s3c-fb.c @@ -7087,6 +7097,7 @@ F: drivers/mmc/host/sdhci-pltfm.[ch] SECURE DIGITAL HOST CONTROLLER INTERFACE (SDHCI) SAMSUNG DRIVER M: Ben Dooks <ben-linux(a)fluff.org> L: linux-mmc(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/mmc/host/sdhci-s3c.c -- 1.8.2.1

4 3

[alsa-devel] [PATCH] ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format.
by Marek Belisko 30 Apr '13

30 Apr '13

When McASP is bit clock and frame clock master enable pin output for rx clocks. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d54e0d9..2953c55 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -505,7 +505,10 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, ACLKX | AFSX); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + ACLKX | ACLKR); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + AFSX | AFSR); break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: Add devicetree support for spdif dummy codecs.
by Marek Belisko 30 Apr '13

30 Apr '13

changes from v1 (for patch 2 and 3): - add prefix linux for compatible property (for both tx and rx) - rename spdif_transceiver.c to spdif_transmitter.c - add bindings documentation Marek Belisko (3): ASoC: spdif_transceiver: Change driver filename to spdif_transmitter.c. ASoC: spdif_transmitter: Add DT support. ASoC: spdif_receiver: Add DT support. .../devicetree/bindings/sound/spdif-receiver.txt | 10 ++++++++++ .../bindings/sound/spdif-transmitter.txt | 10 ++++++++++ sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/spdif_receiver.c | 10 ++++++++++ .../{spdif_transciever.c => spdif_transmitter.c} | 10 ++++++++++ 5 files changed, 41 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/spdif-receiver.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-transmitter.txt rename sound/soc/codecs/{spdif_transciever.c => spdif_transmitter.c} (88%) -- 1.7.9.5

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Re: [alsa-devel] linux-next: Tree for Apr 30 (sound)
by Randy Dunlap 30 Apr '13

30 Apr '13

On 04/29/13 23:57, Stephen Rothwell wrote: > Hi all, > > Please do not add any v3.11 destined work to your linux-next included > branches until after v3.10-rc1 is released. > > Changes since 20130429: > when CONFIG_PROC_FS is not enabled: sound/core/memalloc.c: In function 'snd_mem_exit': sound/core/memalloc.c:525:20: error: 'SND_MEM_PROC_FILE' undeclared (first use in this function) -- ~Randy

1 0

[alsa-devel] [patch] ASoC: wm8994: missing break in wm8994_aif3_hw_params()
by Dan Carpenter 30 Apr '13

30 Apr '13

The missing break here means that we always return early and the function is a no-op. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 14094f5..1eb152c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2882,6 +2882,7 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, default: return 0; } + break; default: return 0; }

3 3

Re: [alsa-devel] what is the difference between SNDRV_PCM_FMTBIT_S24_LE and SNDRV_PCM_FMTBIT_S24_3LE
by Clemens Ladisch 30 Apr '13

30 Apr '13

(please do not drop the mailing list from replies.) Zhang wei wrote: > I do not really understand. For example,an codec is wm8985 ... The SNDRV_PCM_* symbols specify the format in which samples are stored in memory. I do not know how ASoC maps these symbols to on-the-wire formats. >>Most devices that support 24 bits use SNDRV_PCM_FORMAT_S32_LE, which >>stores the sample's bits in the upper bits of a 32-bit word. > > But how to distinguish the SNDRV_PCM_FORMAT_S24_LE and the REAL > SNDRV_PCM_FORMAT_S32_LE? If a 24-bit sample with the value 0x123456 is stored in memory, it looks like this: SNDRV_PCM_FORMAT_S24_LE: 56 34 12 00 SNDRV_PCM_FORMAT_S32_LE: 00 56 34 12 Regards, Clemens

3 2

[alsa-devel] [PATCH, for-next] ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
by Eldad Zack 30 Apr '13

30 Apr '13

Current code does this: be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]) Which is effectively (neglecting the index): be16_to_cpu(be16_to_cpu(*((u16 *) buf))) This means the int16 in the buffer is not converted at all. Daniel Mack confirmed that the driver works on little endian CPUs, leading to the conclusion that the device-side structure is actually little endian. This changes the code to use le16_to_cpu(). Caught by sparse. Acked-by: Daniel Mack <zonque(a)gmail.com> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/caiaq/input.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index efc70ae..4b3fb91 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -488,13 +488,13 @@ static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, unsigned int len) { unsigned int i, pad_id; - uint16_t pressure; + __le16 *pressure = (__le16 *) buf; for (i = 0; i < MASCHINE_PADS; i++) { - pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]); - pad_id = pressure >> 12; - - input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff); + pad_id = le16_to_cpu(*pressure) >> 12; + input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), + le16_to_cpu(*pressure) & 0xfff); + pressure++; } input_sync(cdev->input_dev); -- 1.8.1.5

3 2

[alsa-devel] ASoC : playback and capture counts in soc_new_pcm
by JongHo Kim 29 Apr '13

29 Apr '13

Hi I am developing the ALSA ASoC part. I has some problem. Now my situation must only use the cpu_dai(AP I2S IP) to half duplex. So, I want to allocate either a capture substream or a playback substream. But ASoC(soc-pcm.c) referred to the playback(capture)_channel_min of codec_dai at the calling snd_pcm_new in soc_new_pcm. I do not want to modify the codec_dai codes. As a result, hope to see below /dev/snd/pcmC0D0p /dev/snd/pcmC0D1c attach the ASCII schematic diagram. Slaves Master +--------------+ +-----------------+ |I2S.0 data_out|---------------|data_in(AIF1) | |I2S.0 bclk |-------+-------|bclk | |I2S.0 lrclk |-------|--+----|lrclk | +--------------+ | | | | +--------------+ | | | AUDIO CODEC | |I2S.1 bclk |-------+ | | | |I2S.1 lrclk |----------+ | | |I2S.1 data_in |---------------|data_out(AIF1) | +--------------+ +-----------------+ attach the i2s and dai_link codes. ----------------------------------------------------------------------------- static struct snd_soc_dai_driver board_i2s_dai_hf_playback_template = { .probe = board_i2s_probe, .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &board_i2s_dai_ops, }; static struct snd_soc_dai_driver board_i2s_dai_hf_capture_template = { .probe = board_i2s_probe, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &board_i2s_dai_ops, }; static struct snd_soc_dai_link board_dai[] = { { .name = "audcodec_l0", .stream_name = "audcodec HiFi", .cpu_dai_name = "board-i2s.0", .codec_dai_name = "audcodec-aif1", .platform_name = "board-pcm-audio", .codec_name = "audcodec-codec", .ops = &board_ops, }, { .name = "audcodec_l1", .stream_name = "audcodec HiFi", .cpu_dai_name = "board-i2s.1", .codec_dai_name = "audcodec-aif1", .platform_name = "board-pcm-audio", .codec_name = "audcodec-codec", .ops = &board_ops, }, }; ----------------------------------------------------------------------------- So, I suggest the next diff. --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2014,9 +2014,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) capture = 1; } Thanks.

2 2

[alsa-devel] ALSA SOC Driver (mxs-wm8753)
by John Graham 29 Apr '13

29 Apr '13

Hi there, I'm completely new to both kernel/ALSA driver programming (at least at a serious level) and was looking for some help on a project I'm working on - I'm part way there but I've kind of hit a brick wall and don't know where to turn. I'm trying to get a codec working on a new platform (new for the codec - the platform itself is not new). Here are the details: The audio codec is the Wolfson WM8753, which already has a codec driver in the kernel mainline tree. We also have a vendor-supplied (but GPL) patch to make the wm8753 work on a CC9M2443 (based on the Samsung S3C2443). This includes the glue code ("platform" code?) in sound/soc/s3c24xx/cc9m2443js_wm8753.c. We have a product with this combination fully working. The platform is the Karo TX28 (based on the Freescale i.MX28), and there are already drivers in the kernel tree to make the TX28 work with an SGTL5000 codec in sound/soc/mxs/mxs-sgtl5000.c. We also have a product with this combination fully working. Now we want to move the WM8753 over from the CC9M2443 to work on the TX28. To simplify, what I *have* is: CC9M2443 <----> WM8753 TX28 <----> SGTL5000 and what I *want* is: /-> WM8753 TX28 <-/ What I was hoping I could do was just "mash" the existing drivers together - i.e. copy sound/soc/mxs/mxs-sgtl5000.c into sound/soc/mxs/mxs-wm8753.c, change the parts in it that were specific for the SGTL5000 to the parts that drive the wm8753 in sound/soc/s3c24xx/cc9m2443js_wm8753.c, tell the TX28 to probe for it and that it would "magically work" (hey, crazier things have happened...). Obviously, it does not work, but I've got it a little way down the line. The WM8752 is probed for by the TX28 and recognized - the I2C control works, and you can change the mixer settings with amixer. When I try to record (using arecord) I can see with an oscilloscope the I2S clock lines look correct (in that they look the same as when the CC9M2443 talks to the WM8753 at the same audio rate). The problem now is that the I2S data lines don't look right - the line from codec to TX28 looks like a (more-or-less) square wave with the same period as the I2S clock, but only going from ~3.1 - ~3.5 V, as opposed to the 0 - ~3.3 V I expect. Furthermore, it's a pretty regular square wave - it doesn't have all the breaks you see on a working I2S data line. I know I probably haven't provided the right detail to get to the bottom of this problem, but part of my problem is that I'm so new to this that I'm not entirely sure where the problem might lie, or what extra detail to include to ask for more help. Is anyone able to point me in the right direction/suggest some ideas as to what might be wrong/anything else at all? Any help would be much appreciated - many thanks, John Graham

2 1

[alsa-devel] Configure a CODEC which has I2C connected to CPU, but no PCM bus connection
by Craig McQueen 29 Apr '13

29 Apr '13

I have an embedded device with somewhat unusual configuration: there is a second audio CODEC (Wolfson WM8940) which is connected to the PCM interface of another device (a radio module, which is acting as PCM bus master), so that radio device can do analog audio I/O. So there is no CPU<-->CODEC PCM link, but the CODEC _is_ connected to the CPU's I2C, and the CPU needs to configure the CODEC registers to interface to the radio module. See the attached diagram which shows a simplified view of the configuration. This configuration came about because both the CPU and the radio module can _only_ operate as PCM bus masters. I am not sure how I could fit this into the ALSA ASoC architecture. Is the ALSA ASoC driver architecture able to accommodate this scenario? Can I use the WM8940 driver to configure the second CODEC through I2C? Could I simply set up a machine driver that defines a DAI including a pointer to the WM8940 CODEC DAI, but NULL pointer for CPU DAI? Can I set it up so the Wolfson CODEC gains could appear as ALSA mixer widgets? Regards, Craig McQueen

2 1

[alsa-devel] [PATCH] ASoC: wm8962: Convert to devm_input_allocate_device()
by Mark Brown 29 Apr '13

29 Apr '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 50b0983..1cee9b3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3189,7 +3189,7 @@ static void wm8962_init_beep(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int ret; - wm8962->beep = input_allocate_device(); + wm8962->beep = devm_input_allocate_device(codec->dev); if (!wm8962->beep) { dev_err(codec->dev, "Failed to allocate beep device\n"); return; @@ -3210,7 +3210,6 @@ static void wm8962_init_beep(struct snd_soc_codec *codec) ret = input_register_device(wm8962->beep); if (ret != 0) { - input_free_device(wm8962->beep); wm8962->beep = NULL; dev_err(codec->dev, "Failed to register beep device\n"); } @@ -3227,7 +3226,6 @@ static void wm8962_free_beep(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); device_remove_file(codec->dev, &dev_attr_beep); - input_unregister_device(wm8962->beep); cancel_work_sync(&wm8962->beep_work); wm8962->beep = NULL; -- 1.7.10.4

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[alsa-devel] ALSA: usb-audio: caiaq: endianness bug
by Eldad Zack 29 Apr '13

29 Apr '13

Hi Daniel, I think I found a endian usage issue in the caiaq driver. I have a patch to fix it (below), but it assumes that the driver works currently on little endian CPUs - I don't have a device to test it with. Cheers, Eldad -- >8 -- Subject: [PATCH] ALSA: usb-audio: caiaq: fix endianness bug Current code does this: be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]) Which is effectively: be16_to_cpu(be16_to_cpu(*((u16 *) buf))) This means the int16 in the buffer is not converted at all. Assuming the device-side structure is actually little endian, change the code to use le16_to_cpu(). Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/caiaq/input.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index efc70ae..2f4ae4d 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -488,13 +488,12 @@ static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, unsigned int len) { unsigned int i, pad_id; - uint16_t pressure; + __le16 *pressure = (__le16 *) buf; for (i = 0; i < MASCHINE_PADS; i++) { - pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]); - pad_id = pressure >> 12; - - input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff); + pad_id = le16_to_cpu(*pressure) >> 12; + input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), le16_to_cpu(*pressure) & 0xfff); + pressure++; } input_sync(cdev->input_dev); -- 1.8.1.5

2 2

Re: [alsa-devel] linux-next: Tree for Apr 29 (sound/soc)
by Randy Dunlap 29 Apr '13

29 Apr '13

On 04/29/13 02:17, Stephen Rothwell wrote: > Hi all, > > Changes since 20130426: > on i386: CONFIG_I2C=m sound/built-in.o: In function `snd_soc_codec_set_cache_io': (.text+0x1ebe4): undefined reference to `regmap_init_i2c' make[1]: *** [vmlinux] Error 1 -- ~Randy

1 0

[alsa-devel] [PATCH 0/3] snd-firewire-lib: add handling CMP output connection
by Takashi Sakamoto 29 Apr '13

29 Apr '13

Current implementation of snd-firewire-lib can handle only CMP input connection. This series of patch enable it to handle both CMP output and input connection according to IEC 61883-1. Note that this series of patch still supports only a point-to-point connection without overlaying. The overlaying and any broadcast connection are not supported. Each connections gain its own isochronous resource. Takashi Sakamoto (3): rename macros, variables and functions related to CMP plug Add "direction" member to cmp_connection structure Add handling CMP output connection sound/firewire/cmp.c | 192 +++++++++++++++++++++++++++++++++------------ sound/firewire/cmp.h | 13 ++- sound/firewire/speakers.c | 2 +- 3 files changed, 152 insertions(+), 55 deletions(-) -- 1.7.10.4

2 12

[alsa-devel] Missing patches for 3.10 merge?
by Takashi Iwai 29 Apr '13

29 Apr '13

Hi, if anyone posted fix patches that should be applied to 3.10, but still unapplied, please let me know ASAP. I got a complete home (and system) FS crash on Friday (due to too much experiment with btrfs) and might have lost some incoming mails. thanks, Takashi

2 2

[alsa-devel] [PATCH] ALSA: compress: fix the states to check for allowing read
by Vinod Koul 29 Apr '13

29 Apr '13

for reading compressed data, we need to allow when we are paused, draining or stopped. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Cc: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Cc: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a0bc47f..5389b9a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -311,8 +311,14 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); - /* read is allowed when stream is running */ - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + /* read is allowed when stream is running, paused, draining and setup + * (yes setup is state which we transistion to after stop, so if user + * wants to read data after stop we allow that + */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING || + stream->runtime->state != SNDRV_PCM_STATE_DRAINING || + stream->runtime->state != SNDRV_PCM_STATE_PAUSED || + stream->runtime->state != SNDRV_PCM_STATE_SETUP) { retval = -EBADFD; goto out; } -- 1.7.0.4

4 8

[alsa-devel] [PATCH] compress: fix the states to check for allowing read
by Vinod Koul 29 Apr '13

29 Apr '13

for reading compressed data, we need to allow when we are paused, draining or stopped. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Cc: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Cc: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 12 ++++++++++-- 1 files changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a0bc47f..7b774b7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -311,8 +311,16 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); - /* read is allowed when stream is running */ - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + /* read is allowed when stream is running, paused, draining and setup + * (yes setup is state which we transition to after stop, so if user + * wants to read data after stop we allow that + */ + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_XRUN: + case SNDRV_PCM_STATE_SUSPENDED: + case SNDRV_PCM_STATE_DISCONNECTED: retval = -EBADFD; goto out; } -- 1.7.0.4

2 1

[alsa-devel] [PATCH 0/4] snd-firewire-lib: add handling AMDTP receive stream
by Takashi Sakamoto 29 Apr '13

29 Apr '13

Current implementation of snd-firewire-lib can handle only AMDTP transmit stream with PCM. This series of patch enable it to handle both AMDTP receive and transmit stream with PCM according to IEC 61883-6. I develop with Echo Audio's Fireworks (AudioFirePre8). Unfortunately it's not fully compliant to IEC 61883-6. The last patch includes some work arounds for Fireworks. The snd-fireworks is still under development and not prepared for commit. Takashi Sakamoto (4): Rename functions, structure and member name for AMDTP Add macros for fixed value related to AMDTP Add "direction" member to amdtp_stream structure for AMDTP Add handling AMDTP receive stream sound/firewire/amdtp.c | 422 ++++++++++++++++++++++++++++++++++++--------- sound/firewire/amdtp.h | 64 +++---- sound/firewire/speakers.c | 33 ++-- 3 files changed, 393 insertions(+), 126 deletions(-) -- 1.7.10.4

2 13

[alsa-devel] [PATCH 1/1] ALSA: compress_core: Ensure device is stopped when stream closed
by Richard Fitzgerald 29 Apr '13

29 Apr '13

If the stream state is running or paused when device file is closed snd_compr_free() will send a SNDRV_PCM_TRIGGER_STOP to the stream. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 91e6bcb..a81c5e4 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -132,6 +132,11 @@ static int snd_compr_open(struct inode *inode, struct file *f) static int snd_compr_free(struct inode *inode, struct file *f) { struct snd_compr_file *data = f->private_data; + + if ((data->stream.runtime->state == SNDRV_PCM_STATE_RUNNING) || + (data->stream.runtime->state == SNDRV_PCM_STATE_PAUSED)) + data->stream.ops->trigger(&data->stream, SNDRV_PCM_TRIGGER_STOP); + data->stream.ops->free(&data->stream); kfree(data->stream.runtime->buffer); kfree(data->stream.runtime); -- 1.7.2.5

3 5

[alsa-devel] [PATCH] ALSA: hda - Move Thinkpad X220 to use auto parser
by David Henningsson 29 Apr '13

29 Apr '13

This enables better volume controls than the current model parser. Also, because the original quirk for X220 was added to fix docking station support, add the TP410 fixup instead. Reported-by: Willian Jon McCann <william.jon.mccann(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5499643..84b81c8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2946,7 +2946,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), @@ -3308,6 +3307,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: USB: adjust for changed 3.8 USB API
by Clemens Ladisch 29 Apr '13

29 Apr '13

The recent changes in the USB API ("implement new semantics for URB_ISO_ASAP") made the former meaning of the URB_ISO_ASAP flag the default, and changed this flag to mean that URBs can be delayed. This is not the behaviour wanted by any of the audio drivers because it leads to discontinuous playback with very small period sizes. Therefore, our URBs need to be submitted without this flag. Reported-by: Joe Rayhawk <jrayhawk(a)fairlystable.org> Cc: <stable(a)vger.kernel.org> # 3.8 only Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/6fire/pcm.c | 1 - sound/usb/caiaq/audio.c | 3 --- sound/usb/endpoint.c | 5 ++--- sound/usb/misc/ua101.c | 3 +-- sound/usb/usx2y/usb_stream.c | 1 - sound/usb/usx2y/usbusx2yaudio.c | 1 - sound/usb/usx2y/usx2yhwdeppcm.c | 1 - 7 files changed, 3 insertions(+), 12 deletions(-) diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index e2ca12f..40dd50a 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -575,7 +575,6 @@ static void usb6fire_pcm_init_urb(struct pcm_urb *urb, urb->instance.pipe = in ? usb_rcvisocpipe(chip->dev, ep) : usb_sndisocpipe(chip->dev, ep); urb->instance.interval = 1; - urb->instance.transfer_flags = URB_ISO_ASAP; urb->instance.complete = handler; urb->instance.context = urb; urb->instance.number_of_packets = PCM_N_PACKETS_PER_URB; diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index fde9a7a..b45e29b 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -670,7 +670,6 @@ static void read_completed(struct urb *urb) if (send_it) { out->number_of_packets = outframe; - out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } else { struct snd_usb_caiaq_cb_info *oinfo = out->context; @@ -686,7 +685,6 @@ requeue: } urb->number_of_packets = FRAMES_PER_URB; - urb->transfer_flags = URB_ISO_ASAP; usb_submit_urb(urb, GFP_ATOMIC); } @@ -751,7 +749,6 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) * BYTES_PER_FRAME; urbs[i]->context = &dev->data_cb_info[i]; urbs[i]->interval = 1; - urbs[i]->transfer_flags = URB_ISO_ASAP; urbs[i]->number_of_packets = FRAMES_PER_URB; urbs[i]->complete = (dir == SNDRV_PCM_STREAM_CAPTURE) ? read_completed : write_completed; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 21049b8..63cca3a 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -677,7 +677,7 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, if (!u->urb->transfer_buffer) goto out_of_memory; u->urb->pipe = ep->pipe; - u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; + u->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP; u->urb->interval = 1 << ep->datainterval; u->urb->context = u; u->urb->complete = snd_complete_urb; @@ -716,8 +716,7 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep, u->urb->transfer_dma = ep->sync_dma + i * 4; u->urb->transfer_buffer_length = 4; u->urb->pipe = ep->pipe; - u->urb->transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; + u->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP; u->urb->number_of_packets = 1; u->urb->interval = 1 << ep->syncinterval; u->urb->context = u; diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 8b81cb5..6ad617b 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -1120,8 +1120,7 @@ static int alloc_stream_urbs(struct ua101 *ua, struct ua101_stream *stream, usb_init_urb(&urb->urb); urb->urb.dev = ua->dev; urb->urb.pipe = stream->usb_pipe; - urb->urb.transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; + urb->urb.transfer_flags = URB_NO_TRANSFER_DMA_MAP; urb->urb.transfer_buffer = addr; urb->urb.transfer_dma = dma; urb->urb.transfer_buffer_length = max_packet_size; diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 1e7a47a..bf618e1 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -69,7 +69,6 @@ static void init_pipe_urbs(struct usb_stream_kernel *sk, unsigned use_packsize, ++u, transfer += transfer_length) { struct urb *urb = urbs[u]; struct usb_iso_packet_descriptor *desc; - urb->transfer_flags = URB_ISO_ASAP; urb->transfer_buffer = transfer; urb->dev = dev; urb->pipe = pipe; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 520ef96..b376532 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -503,7 +503,6 @@ static int usX2Y_urbs_start(struct snd_usX2Y_substream *subs) if (0 == i) atomic_set(&subs->state, state_STARTING3); urb->dev = usX2Y->dev; - urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * pack; urb->iso_frame_desc[pack].length = subs->maxpacksize; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index cc56007..f2a1acd 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -443,7 +443,6 @@ static int usX2Y_usbpcm_urbs_start(struct snd_usX2Y_substream *subs) if (0 == u) atomic_set(&subs->state, state_STARTING3); urb->dev = usX2Y->dev; - urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * (pack + u * nr_of_packs()); urb->iso_frame_desc[pack].length = subs->maxpacksize;

3 2

[alsa-devel] [TINYCOMPRESS][PATCH] compress: check for config first
by Vinod Koul 26 Apr '13

26 Apr '13

if invalid config is passed, then we return error for memory Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- compress.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/compress.c b/compress.c index e80e598..ee792c3 100644 --- a/compress.c +++ b/compress.c @@ -218,8 +218,13 @@ struct compress *compress_open(unsigned int card, unsigned int device, struct snd_compr_caps caps; char fn[256]; + if (!config) { + oops(&bad_compress, -EINVAL, "passed bad config"); + return &bad_compress; + } + compress = calloc(1, sizeof(struct compress)); - if (!compress || !config) { + if (!compress) { oops(&bad_compress, errno, "cannot allocate compress object"); return &bad_compress; } -- 1.7.0.4

1 0

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] compress: Add support for capture streams
by Richard Fitzgerald 26 Apr '13

26 Apr '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 64 +++++++++++++++++++++++++++++++--- include/tinycompress/tinycompress.h | 2 +- 2 files changed, 59 insertions(+), 7 deletions(-) diff --git a/compress.c b/compress.c index 86cca88..e80e598 100644 --- a/compress.c +++ b/compress.c @@ -240,13 +240,12 @@ struct compress *compress_open(unsigned int card, unsigned int device, oops(&bad_compress, -EINVAL, "can't deduce device direction from given flags"); goto config_fail; } + if (flags & COMPRESS_OUT) { - /* this should be removed once we have capture tested */ - oops(&bad_compress, -EINVAL, "this version doesnt support capture"); - goto config_fail; + compress->fd = open(fn, O_RDONLY); + } else { + compress->fd = open(fn, O_WRONLY); } - - compress->fd = open(fn, O_WRONLY); if (compress->fd < 0) { oops(&bad_compress, errno, "cannot open device '%s'", fn); goto config_fail; @@ -403,7 +402,60 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size int compress_read(struct compress *compress, void *buf, unsigned int size) { - return oops(compress, -ENOTTY, "Not supported yet in lib"); + struct snd_compr_avail avail; + struct pollfd fds; + int to_read = 0; + int num_read, total = 0, ret; + char* cbuf = buf; + const unsigned int frag_size = compress->config->fragment_size; + + if (!(compress->flags & COMPRESS_OUT)) + return oops(compress, -EINVAL, "Invalid flag set"); + if (!is_compress_ready(compress)) + return oops(compress, -ENODEV, "device not ready"); + fds.fd = compress->fd; + fds.events = POLLIN; + + while (size) { + if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &avail)) + return oops(compress, errno, "cannot get avail"); + + if ( (avail.avail < frag_size) && (avail.avail < size) ) { + /* Less than one fragment available and not at the + * end of the read, so poll + */ + ret = poll(&fds, 1, compress->max_poll_wait_ms); + /* A pause will cause -EBADFD or zero. + * This is not an error, just stop reading */ + if ((ret == 0) || (ret == -EBADFD)) + break; + if (ret < 0) + return oops(compress, errno, "poll error"); + if (fds.revents & POLLIN) { + continue; + } + if (fds.revents & POLLERR) { + return oops(compress, -EIO, "poll returned error!"); + } + } + /* read avail bytes */ + if (size > avail.avail) + to_read = avail.avail; + else + to_read = size; + num_read = read(compress->fd, cbuf, to_read); + /* If play was paused the read returns -EBADFD */ + if (num_read == -EBADFD) + break; + if (num_read < 0) + return oops(compress, errno, "read failed!"); + + size -= num_read; + cbuf += num_read; + total += num_read; + } + + return total; } int compress_start(struct compress *compress) diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index ba9f3b6..9867afc 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -76,7 +76,7 @@ struct compr_gapless_mdata { __u32 encoder_padding; }; -#define COMPRESS_OUT 0x00000000 +#define COMPRESS_OUT 0x20000000 #define COMPRESS_IN 0x10000000 struct compress; -- 1.7.2.5

2 1

[alsa-devel] [PATCH 00/21] more arm build fixes
by Arnd Bergmann 26 Apr '13

26 Apr '13

Hi subsystem maintainers, Here is another set of patches that resulted from build testing on linux-next. Please apply directly into your trees if you agree, or let me know if I made a mistake. I can take whatever remains through the arm-soc tree if you prefer that or I don't hear back. Arnd Arnd Bergmann (21): ARM: topology: export cpu_topology ARM: default machine descriptor for multiplatform ARM: shmobile: don't call irqchip_init unconditionally ARM: orion5x: include linux/cpu.h atm: he: use mdelay instead of large udelay constants ALSA: ali5451: use mdelay instead of large udelay constants oss/dmabuf: use dma_map_single drm/nouveau: use mdelay instead of large udelay constants drm: export drm_vm_open_locked [SCSI] nsp32: use mdelay instead of large udelay constants irqdomain: export irq_domain_add_simple irqchip: s3c24xx: add missing __init annotations iommu: tegra: print dma_addr_t using %lld cpufreq: pxa2xx: initialize variables thermal: cooling: avoid uninitialied used gcc warning OF: remove #ifdef from linux/of_platform.h X.509: do not emit any informational output USB: ehci-msm: USB_MSM_OTG needs USB_PHY USB: lpc32xx: ISP1301 needs USB_PHY USB: OMAP: ISP1301 needs USB_PHY USB: OHCI: avoid conflicting platform drivers arch/arm/Kconfig | 1 - arch/arm/configs/lpc32xx_defconfig | 1 + arch/arm/configs/msm_defconfig | 1 + arch/arm/configs/omap1_defconfig | 1 + arch/arm/kernel/devtree.c | 7 ++ arch/arm/kernel/setup.c | 11 +- arch/arm/kernel/topology.c | 1 + arch/arm/mach-orion5x/common.c | 1 + arch/arm/mach-shmobile/intc-r8a7740.c | 13 +- drivers/atm/he.c | 2 +- drivers/cpufreq/pxa2xx-cpufreq.c | 5 +- drivers/gpu/drm/drm_vm.c | 1 + drivers/gpu/drm/nouveau/core/engine/disp/dacnv50.c | 3 +- drivers/iommu/tegra-gart.c | 3 +- drivers/iommu/tegra-smmu.c | 2 +- drivers/irqchip/irq-s3c24xx.c | 4 +- drivers/scsi/nsp32.c | 2 +- drivers/thermal/cpu_cooling.c | 17 +-- drivers/usb/gadget/Kconfig | 2 + drivers/usb/host/Kconfig | 1 + drivers/usb/host/ohci-hcd.c | 136 ++++++++++++++++++--- drivers/usb/phy/Makefile | 2 +- include/linux/of_platform.h | 13 +- kernel/irq/irqdomain.c | 1 + lib/build_OID_registry | 2 - sound/oss/dmabuf.c | 3 +- sound/pci/ali5451/ali5451.c | 8 +- 27 files changed, 187 insertions(+), 57 deletions(-) -- 1.8.1.2 Cc: "James E.J. Bottomley" <JBottomley(a)parallels.com> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> Cc: David Airlie <airlied(a)linux.ie> Cc: Felipe Balbi <balbi(a)ti.com> Cc: Grant Likely <grant.likely(a)linaro.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Inki Dae <inki.dae(a)samsung.com> Cc: Jason Cooper <jason(a)lakedaemon.net> Cc: Joerg Roedel <joro(a)8bytes.org> Cc: Nicolas Pitre <nico(a)linaro.org> Cc: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Cc: Rob Herring <robherring2(a)gmail.com> Cc: Russell King <rmk+kernel(a)arm.linux.org.uk> Cc: Simon Horman <horms+renesas(a)verge.net.au> Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Viresh Kumar <viresh.kumar(a)linaro.org> Cc: Will Deacon <will.deacon(a)arm.com> Cc: alsa-devel(a)alsa-project.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-atm-general(a)lists.sourceforge.net Cc: linux-scsi(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: netdev(a)vger.kernel.org

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[alsa-devel] [PATCH] alsa-lib: conf - Add Scarlett 2i2 USB to S/PDIF blacklist
by David Henningsson 26 Apr '13

26 Apr '13

The Scarlett 2i2 USB interface does not have any S/PDIF connections. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 71c2971..0d93175 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -47,6 +47,7 @@ USB-Audio.pcm.iec958_device { "Plantronics USB Headset" 999 "Plantronics Wireless Audio" 999 "SB WoW Headset" 999 + "Scarlett 2i2 USB" 999 "Sennheiser USB headset" 999 "USB Device 0x46d:0x992" 999 } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
by David Henningsson 26 Apr '13

26 Apr '13

The Scarlett 2i2 seems to take almost 500 ms to set the sample rate, even if the clock is currently set to that value. This patch speeds up prepare of the device, by avoiding setting the clock to something it already is. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/usb/clock.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b0ec364..3a2ce39 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -356,6 +356,8 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return clock; prev_rate = get_sample_rate_v2(chip, iface, fmt->altsetting, clock); + if (prev_rate == rate) + return 0; cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, clock); writeable = uac2_control_is_writeable(cs_desc->bmControls, UAC2_CS_CONTROL_SAM_FREQ - 1); -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH 04/28] proc: Supply PDE attribute setting accessor functions [RFC]
by Vasant Hegde 25 Apr '13

25 Apr '13

On 04/16/2013 11:56 PM, David Howells wrote: > Supply accessor functions to set attributes in proc_dir_entry structs. > > The following are supplied: proc_set_size() and proc_set_user(). > > Signed-off-by: David Howells<dhowells(a)redhat.com> > cc: linuxppc-dev(a)lists.ozlabs.org > cc: linux-media(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: linux-wireless(a)vger.kernel.org > cc: linux-pci(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org > --- > > arch/powerpc/kernel/proc_powerpc.c | 2 +- > arch/powerpc/platforms/pseries/reconfig.c | 2 +- arch/powerpc side changes looks good. -Vasant > drivers/media/pci/ttpci/av7110_ir.c | 2 +- > drivers/net/irda/vlsi_ir.c | 2 +- > drivers/net/wireless/airo.c | 34 +++++++++-------------------- > drivers/pci/proc.c | 2 +- > fs/proc/generic.c | 13 +++++++++++ > include/linux/proc_fs.h | 5 ++++ > kernel/configs.c | 2 +- > kernel/profile.c | 2 +- > net/netfilter/xt_recent.c | 3 +-- > sound/core/info.c | 2 +- > 12 files changed, 38 insertions(+), 33 deletions(-) > > diff --git a/arch/powerpc/kernel/proc_powerpc.c b/arch/powerpc/kernel/proc_powerpc.c > index 41d8ee9..feb8580 100644 > --- a/arch/powerpc/kernel/proc_powerpc.c > +++ b/arch/powerpc/kernel/proc_powerpc.c > @@ -83,7 +83,7 @@ static int __init proc_ppc64_init(void) > &page_map_fops, vdso_data); > if (!pde) > return 1; > - pde->size = PAGE_SIZE; > + proc_set_size(pde, PAGE_SIZE); > > return 0; > } > diff --git a/arch/powerpc/platforms/pseries/reconfig.c b/arch/powerpc/platforms/pseries/reconfig.c > index d6491bd..f93cdf5 100644 > --- a/arch/powerpc/platforms/pseries/reconfig.c > +++ b/arch/powerpc/platforms/pseries/reconfig.c > @@ -452,7 +452,7 @@ static int proc_ppc64_create_ofdt(void) > > ent = proc_create("powerpc/ofdt", S_IWUSR, NULL,&ofdt_fops); > if (ent) > - ent->size = 0; > + proc_set_size(ent, 0); > > return 0; > } > diff --git a/drivers/media/pci/ttpci/av7110_ir.c b/drivers/media/pci/ttpci/av7110_ir.c > index eb82286..0e763a7 100644 > --- a/drivers/media/pci/ttpci/av7110_ir.c > +++ b/drivers/media/pci/ttpci/av7110_ir.c > @@ -375,7 +375,7 @@ int av7110_ir_init(struct av7110 *av7110) > if (av_cnt == 1) { > e = proc_create("av7110_ir", S_IWUSR, NULL,&av7110_ir_proc_fops); > if (e) > - e->size = 4 + 256 * sizeof(u16); > + proc_set_size(e, 4 + 256 * sizeof(u16)); > } > > tasklet_init(&av7110->ir.ir_tasklet, av7110_emit_key, (unsigned long)&av7110->ir); > diff --git a/drivers/net/irda/vlsi_ir.c b/drivers/net/irda/vlsi_ir.c > index e22cd4e..5f47584 100644 > --- a/drivers/net/irda/vlsi_ir.c > +++ b/drivers/net/irda/vlsi_ir.c > @@ -1678,7 +1678,7 @@ vlsi_irda_probe(struct pci_dev *pdev, const struct pci_device_id *id) > IRDA_WARNING("%s: failed to create proc entry\n", > __func__); > } else { > - ent->size = 0; > + proc_set_size(ent, 0); > } > idev->proc_entry = ent; > } > diff --git a/drivers/net/wireless/airo.c b/drivers/net/wireless/airo.c > index 66e398d..21d0233 100644 > --- a/drivers/net/wireless/airo.c > +++ b/drivers/net/wireless/airo.c > @@ -4507,73 +4507,63 @@ static int setup_proc_entry( struct net_device *dev, > airo_entry); > if (!apriv->proc_entry) > goto fail; > - apriv->proc_entry->uid = proc_kuid; > - apriv->proc_entry->gid = proc_kgid; > + proc_set_user(apriv->proc_entry, proc_kuid, proc_kgid); > > /* Setup the StatsDelta */ > entry = proc_create_data("StatsDelta", S_IRUGO& proc_perm, > apriv->proc_entry,&proc_statsdelta_ops, dev); > if (!entry) > goto fail_stats_delta; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Stats */ > entry = proc_create_data("Stats", S_IRUGO& proc_perm, > apriv->proc_entry,&proc_stats_ops, dev); > if (!entry) > goto fail_stats; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Status */ > entry = proc_create_data("Status", S_IRUGO& proc_perm, > apriv->proc_entry,&proc_status_ops, dev); > if (!entry) > goto fail_status; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Config */ > entry = proc_create_data("Config", proc_perm, > apriv->proc_entry,&proc_config_ops, dev); > if (!entry) > goto fail_config; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the SSID */ > entry = proc_create_data("SSID", proc_perm, > apriv->proc_entry,&proc_SSID_ops, dev); > if (!entry) > goto fail_ssid; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the APList */ > entry = proc_create_data("APList", proc_perm, > apriv->proc_entry,&proc_APList_ops, dev); > if (!entry) > goto fail_aplist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the BSSList */ > entry = proc_create_data("BSSList", proc_perm, > apriv->proc_entry,&proc_BSSList_ops, dev); > if (!entry) > goto fail_bsslist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the WepKey */ > entry = proc_create_data("WepKey", proc_perm, > apriv->proc_entry,&proc_wepkey_ops, dev); > if (!entry) > goto fail_wepkey; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > - > + proc_set_user(entry, proc_kuid, proc_kgid); > return 0; > > fail_wepkey: > @@ -5695,10 +5685,8 @@ static int __init airo_init_module( void ) > > airo_entry = proc_mkdir_mode("driver/aironet", airo_perm, NULL); > > - if (airo_entry) { > - airo_entry->uid = proc_kuid; > - airo_entry->gid = proc_kgid; > - } > + if (airo_entry) > + proc_set_user(airo_entry, proc_kuid, proc_kgid); > > for (i = 0; i< 4&& io[i]&& irq[i]; i++) { > airo_print_info("", "Trying to configure ISA adapter at irq=%d " > diff --git a/drivers/pci/proc.c b/drivers/pci/proc.c > index 12e4fb5..7cde7c1 100644 > --- a/drivers/pci/proc.c > +++ b/drivers/pci/proc.c > @@ -419,7 +419,7 @@ int pci_proc_attach_device(struct pci_dev *dev) > &proc_bus_pci_operations, dev); > if (!e) > return -ENOMEM; > - e->size = dev->cfg_size; > + proc_set_size(e, dev->cfg_size); > dev->procent = e; > > return 0; > diff --git a/fs/proc/generic.c b/fs/proc/generic.c > index 1c07cad..5f6f6c3 100644 > --- a/fs/proc/generic.c > +++ b/fs/proc/generic.c > @@ -498,6 +498,19 @@ out: > return NULL; > } > EXPORT_SYMBOL(proc_create_data); > + > +void proc_set_size(struct proc_dir_entry *de, loff_t size) > +{ > + de->size = size; > +} > +EXPORT_SYMBOL(proc_set_size); > + > +void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) > +{ > + de->uid = uid; > + de->gid = gid; > +} > +EXPORT_SYMBOL(proc_set_user); > > static void free_proc_entry(struct proc_dir_entry *de) > { > diff --git a/include/linux/proc_fs.h b/include/linux/proc_fs.h > index 805edac..28a4d7e 100644 > --- a/include/linux/proc_fs.h > +++ b/include/linux/proc_fs.h > @@ -130,6 +130,9 @@ static inline struct proc_dir_entry *proc_create(const char *name, umode_t mode, > extern struct proc_dir_entry *proc_net_mkdir(struct net *net, const char *name, > struct proc_dir_entry *parent); > > +extern void proc_set_size(struct proc_dir_entry *, loff_t); > +extern void proc_set_user(struct proc_dir_entry *, kuid_t, kgid_t); > + > extern struct file *proc_ns_fget(int fd); > extern bool proc_ns_inode(struct inode *inode); > > @@ -158,6 +161,8 @@ static inline struct proc_dir_entry *proc_mkdir(const char *name, > struct proc_dir_entry *parent) {return NULL;} > static inline struct proc_dir_entry *proc_mkdir_mode(const char *name, > umode_t mode, struct proc_dir_entry *parent) { return NULL; } > +static inline void proc_set_size(struct proc_dir_entry *de, loff_t size) {} > +static inline void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) {} > > struct tty_driver; > static inline void proc_tty_register_driver(struct tty_driver *driver) {}; > diff --git a/kernel/configs.c b/kernel/configs.c > index 42e8fa0..c18b1f1 100644 > --- a/kernel/configs.c > +++ b/kernel/configs.c > @@ -79,7 +79,7 @@ static int __init ikconfig_init(void) > if (!entry) > return -ENOMEM; > > - entry->size = kernel_config_data_size; > + proc_set_size(entry, kernel_config_data_size); > > return 0; > } > diff --git a/kernel/profile.c b/kernel/profile.c > index 524ce5e..0bf4007 100644 > --- a/kernel/profile.c > +++ b/kernel/profile.c > @@ -600,7 +600,7 @@ int __ref create_proc_profile(void) /* false positive from hotcpu_notifier */ > NULL,&proc_profile_operations); > if (!entry) > return 0; > - entry->size = (1+prof_len) * sizeof(atomic_t); > + proc_set_size(entry, (1 + prof_len) * sizeof(atomic_t)); > hotcpu_notifier(profile_cpu_callback, 0); > return 0; > } > diff --git a/net/netfilter/xt_recent.c b/net/netfilter/xt_recent.c > index 3db2d38..1e657cf 100644 > --- a/net/netfilter/xt_recent.c > +++ b/net/netfilter/xt_recent.c > @@ -401,8 +401,7 @@ static int recent_mt_check(const struct xt_mtchk_param *par, > ret = -ENOMEM; > goto out; > } > - pde->uid = uid; > - pde->gid = gid; > + proc_set_user(pde, uid, gid); > #endif > spin_lock_bh(&recent_lock); > list_add_tail(&t->list,&recent_net->tables); > diff --git a/sound/core/info.c b/sound/core/info.c > index 3aa8864..c7f41c3 100644 > --- a/sound/core/info.c > +++ b/sound/core/info.c > @@ -970,7 +970,7 @@ int snd_info_register(struct snd_info_entry * entry) > mutex_unlock(&info_mutex); > return -ENOMEM; > } > - p->size = entry->size; > + proc_set_size(p, entry->size); > } > entry->p = p; > if (entry->parent) > > _______________________________________________ > Linuxppc-dev mailing list > Linuxppc-dev(a)lists.ozlabs.org > https://lists.ozlabs.org/listinfo/linuxppc-dev >

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[alsa-devel] [PATCH v5 00/10] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 25 Apr '13

25 Apr '13

Hi, This series adds DT support for phycore-ac97. It is now based on the pcm dma cleanups from Lars. Beside ac97 support, the series adds fsl-ssi imx-pcm-fiq and generic DMA binding handling. Regards, Markus Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (10): ASoC: phycore-ac97: Add DT support ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 510 +++++++++++++++++---- sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 16 +- sound/soc/fsl/imx-pcm.h | 9 + sound/soc/fsl/imx-ssi.c | 7 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97.c | 166 ++++++- 12 files changed, 631 insertions(+), 122 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt

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[alsa-devel] ASoC updates for v3.10
by Mark Brown 25 Apr '13

25 Apr '13

The following changes since commit 22f38f792ec53e2a93be13ecb609bbe911ed8ff9: ASoC: ux500: Use generic dmaengine PCM (2013-04-18 15:04:44 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-3 for you to fetch changes up to 5cc50fc858a5ab37dc2744d72d7ffed96f23afd8: Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next (2013-04-23 19:26:15 +0100) Diffstat and log are inflated by Linus' tree again, plus git has apparently got itself a bit confused about what to diff against and picked the wrong side of a merge. ---------------------------------------------------------------- ASoC: More updates for v3.10 A few more fixes, nothing too major though the DMA changes fix modular builds. ---------------------------------------------------------------- Aaro Koskinen (1): ARM: 7692/1: iop3xx: move IOP3XX_PERIPHERAL_VIRT_BASE Abhijeet Dharmapurikar (1): irqchip: gic: fix irq_trigger return Akinobu Mita (2): sparc/srmmu: clear trailing edge of bitmap properly sparc/iommu: fix typo s/265KB/256KB/ Alban Bedel (1): ASoC: wm8903: Add the DAC boost control Alex Elder (1): rbd: do a safe list traversal in rbd_img_request_submit() Alexander Duyck (1): igb: Revert support for build_skb in igb Alistair Popple (1): powerpc: Fix audit crash due to save/restore PPR changes Andreas Schwab (1): kbuild: generate generic headers before recursing into scripts Andrei Epure (1): drivers/video/mmp/core.c: fix use-after-free bug Andrey Smirnov (2): ASoC: si476x: Convert SI476X codec to use regmap ASoC: si476x: Cosmetic changes to SI476X codec driver Andy Lutomirski (2): userns: Check uid_map's opener's fsuid, not the current fsuid userns: Changing any namespace id mappings should require privileges Antonio Quartulli (1): batman-adv: make is_my_mac() check for the current mesh only Arend van Spriel (2): brcmfmac: remove advertising P2P device support brcmsmac: request firmware in .start() callback Ariel Elior (1): bnx2x: Fix status blocks configuration Arnd Bergmann (6): ASoC: samsung: use irq resource for idma ASoC: samsung: fix module_device_table ASoC: samsung: export idma_reg_addr_init ASoC: samsung: move plat/ headers to local directory net: ax88796: avoid 64 bit arithmetic ASoC: ux500: forward declare msp_i2s_platform_data Bastian Hecht (1): ARM: 7697/1: hw_breakpoint: do not use __cpuinitdata for dbg_cpu_pm_nb Ben Greear (1): net: rate-limit warn-bad-offload splats. Bjørn Mork (4): net: cdc_mbim: remove bogus sizeof() net: qmi_wwan: fixup missing ethernet header (firmware bug workaround) net: qmi_wwan: fixup destination address (firmware bug workaround) net: qmi_wwan: prevent duplicate mac address on link (firmware bug workaround) Charles Keepax (12): ASoC: arizona: Move selection of FLL REFCLK into init ASoC: arizona: Tidy up SYNCCLK selection and cache values ASoC: arizona: Factor out check for enabled FLL ASoC: arizona: Factor out FLL disable ASoC: arizona: Factor out FLL enable ASoC: arizona: Improve suppression of noop FLL updates ASoC: arizona: Add support for directly setting the FLL REFCLK ASoC: arizona: Add convience define for clearing SYNCCLK ASoC: arizona: Don't enable FLL on REFCLK configuration ASoC: wm5102: Correctly use SOC_VALUE_ENUM for ISRC FSL controls ASoC: wm0010: Constify usage of firmware filenames ASoC: wm0010: Report filename when we fail to load firmware Christian Ruppert (1): stmmac: prevent interrupt loop with MMC RX IPC Counter Christoph Fritz (1): can: sja1000: fix handling on dt properties on little endian systems Cong Wang (1): ipv6: statically link register_inet6addr_notifier() Dan Carpenter (2): ASoC: tas5086: signedness bug in tas5086_hw_params() irda: small read past the end of array in debug code Daniel Mack (5): ASoC: ak4104: convert to direct regmap API usage ASoC: ak4104: correct tranceiver enable handling ASoC: Add codec driver for AK5386 ALSA: ASoC: add codec driver for TI TAS5086 ASoC: davinci-mcasp: clean up davinci_hw_common_param() David S. Miller (10): Merge branch 'master' of git://git.kernel.org/.../pablo/nf into netfilter Merge branch 'fixes-for-3.9' of git://gitorious.org/linux-can/linux-can Merge branch 'fixes' of git://git.kernel.org/.../jesse/openvswitch Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge Merge branch 'qlogic' Merge branch 'for-davem' of git://git.kernel.org/.../linville/wireless Merge branch 'master' of git://git.kernel.org/.../pablo/nf sparc64: Fix race in TLB batch processing. Merge branch 'bonding' Merge branch 'qmi_wwan' David Ward (1): net/802/mrp: fix possible race condition when calling mrp_pdu_queue() Dmitry Kravkov (1): bnx2x: Prevent UNDI FW illegal host access Dmitry Popov (1): tcp: incoming connections might use wrong route under synflood Emese Revfy (1): kernel/signal.c: stop info leak via the tkill and the tgkill syscalls Eric Dumazet (3): net: drop dst before queueing fragments bonding: fix l23 and l34 load balancing in forwarding path tcp: call tcp_replace_ts_recent() from tcp_ack() Eric W. Biederman (1): userns: Don't let unprivileged users trick privileged users into setting the id_map Felix Fietkau (2): ath9k_htc: accept 1.x firmware newer than 1.3 ath9k_hw: change AR9580 initvals to fix a stability issue Florian Westphal (2): netfilter: nf_nat: fix race when unloading protocol modules netfilter: xt_rpfilter: skip locally generated broadcast/multicast, too Greg Rose (1): ixgbe: Fix a bug in setting VF VLAN via PF H. Peter Anvin (2): x86, microcode: Verify the family before dispatching microcode patching Merge remote-tracking branch 'efi/urgent' into x86/urgent Heiko Stübner (1): ASoC: samsung: fix neo1973-wm8753 compilation Himanshu Madhani (1): qlcnic: fix beaconing test for 82xx adapter Hong Zhiguo (1): openvswitch: correct an invalid BUG_ON Illia Ragozin (1): ARM: 7696/1: Fix kexec by setting outer_cache.inv_all for Feroceon Ivan Vecera (1): be2net: take care of __vlan_put_tag return value Jason Gerecke (2): Input: wacom - fix "can not retrieve extra class descriptor" for 24HDT Input: wacom - correct reported resolution for Intuos4 Wireless Jeeja KP (1): ASoC: compress - add support for metadata apis Jesse Gross (1): openvswitch: Preallocate reply skb in ovs_vport_cmd_set(). Jitendra Kalsaria (3): qlcnic: Stop traffic before performing loopback test qlge: Fix ethtool autoneg advertising. qlge: Update version to 1.00.00.32. Joe Perches (1): checkpatch: fix stringification macro defect Johannes Berg (2): mac80211: fix LED in idle handling mac80211: fix cfg80211 interaction on auth/assoc request John W. Linville (3): Merge branch 'for-john' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'for-john' of git://x-git.kernel.org/.../jberg/mac80211 Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Joonsoo Kim (1): ARM: 7694/1: ARM, TCM: initialize TCM in paging_init(), instead of setup_arch() Josh Wu (1): avr32: fix build error in atstk1006_defconfig Jozsef Kadlecsik (3): netfilter: ipset: list:set: fix reference counter update netfilter: ipset: hash:*net*: nomatch flag not excluded on set resize netfilter: ipset: bitmap:ip,mac: fix listing with timeout K. Y. Srinivasan (1): x86, hyperv: Handle Xen emulation of Hyper-V more gracefully Kefeng Wang (2): sparc:remove unused declaration smp_boot_cpus() sparc:cleanup unused code in smp_32.h Kevin Hao (1): powerpc: add a missing label in resume_kernel Kuninori Morimoto (58): ASoC: fsi: remove unused irq ASoC: add snd_soc_register_component() ASoC: add .name for snd_soc_component_driver ASoC: fsi: use snd_soc_register_component() instead of snd_soc_register_dais() ASoC: pxa2xx-ac97: move EXPORT_SYMBOL_GPL() next to definition ASoC: snd_soc_register_component() uses properly snd_soc_register_dai[s]() ASoC: switch over to use snd_soc_register_component() on davinci i2s ASoC: switch over to use snd_soc_register_component() on davinci vcif ASoC: switch over to use snd_soc_register_component() on davinci mcasp ASoC: switch over to use snd_soc_register_component() on dw i2s ASoC: switch over to use snd_soc_register_component() on mpc5200 ac97 ASoC: switch over to use snd_soc_register_component() on atmel ssc ASoC: switch over to use snd_soc_register_component() on au1x i2sc ASoC: switch over to use snd_soc_register_component() on au1x psc-ac97 ASoC: switch over to use snd_soc_register_component() on au1x psc-i2s ASoC: switch over to use snd_soc_register_component() on au1x ac97c ASoC: switch over to use snd_soc_register_component() on bf6xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx ac97 ASoC: switch over to use snd_soc_register_component() on bf5xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx tdm ASoC: switch over to use snd_soc_register_component() on ep93xx ac97 ASoC: switch over to use snd_soc_register_component() on ep93xx i2s ASoC: switch over to use snd_soc_register_component() on mpc5200 i2s ASoC: switch over to use snd_soc_register_component() on fsl ssi ASoC: switch over to use snd_soc_register_component() on imx ssi ASoC: switch over to use snd_soc_register_component() on jz4740 i2s ASoC: switch over to use snd_soc_register_component() on kirkwood i2s ASoC: switch over to use snd_soc_register_component() on sst ASoC: switch over to use snd_soc_register_component() on mxs saif ASoC: switch over to use snd_soc_register_component() on nuc900 ac97 ASoC: switch over to use snd_soc_register_component() on omap mcbsp ASoC: switch over to use snd_soc_register_component() on omap mcpdm ASoC: switch over to use snd_soc_register_component() on omap dmic ASoC: switch over to use snd_soc_register_component() on omap hdmi ASoC: switch over to use snd_soc_register_component() on pxa ssp ASoC: switch over to use snd_soc_register_component() on pxa2xx ac97 ASoC: switch over to use snd_soc_register_component() on pxa2xx i2s ASoC: switch over to use snd_soc_register_component() on mmp sspa ASoC: switch over to use snd_soc_register_component() on s6000 i2s ASoC: switch over to use snd_soc_register_component() on s3c24xx i2s ASoC: switch over to use snd_soc_register_component() on s3c i2s ASoC: switch over to use snd_soc_register_component() on s3c ac97 ASoC: switch over to use snd_soc_register_component() on samsung spdif ASoC: switch over to use snd_soc_register_component() on samsung i2s ASoC: switch over to use snd_soc_register_component() on samsung pcm ASoC: switch over to use snd_soc_register_component() on goni_wm8994 ASoC: switch over to use snd_soc_register_component() on sh4 ssi ASoC: switch over to use snd_soc_register_component() on sh4 hac ASoC: switch over to use snd_soc_register_component() on spear spdif out ASoC: switch over to use snd_soc_register_component() on spear spdif in ASoC: switch over to use snd_soc_register_component() on tegra30 i2s ASoC: switch over to use snd_soc_register_component() on tegra20 spdif ASoC: switch over to use snd_soc_register_component() on tegra20 ac97 ASoC: switch over to use snd_soc_register_component() on tegra20 i2s ASoC: switch over to use snd_soc_register_component() on txx9aclc ac97 ASoC: switch over to use snd_soc_register_component() on ux500 msp ASoC: snd_soc_[un]register_dai[s]() become non global function ASoC: switch over to use snd_soc_register_component() on sh4 siu Lars-Peter Clausen (5): ASoC: spear_pcm: Staticize non-exported structs ASoC: generic-dmaengine-pcm: Add support for half-duplex ASoC: mxs-pcm: Set SNDRV_PCM_INFO_HALF_DUPLEX ASoC: mxs: Setup dma data in DAI probe ASoC: mxs: Use generic dmaengine PCM Len Brown (2): intel_idle: additional Haswell CPU-id tools/power turbostat: additional Haswell CPU-id Linus Torvalds (32): Merge branches 'timers-urgent-for-linus', 'irq-urgent-for-linus' and 'core-urgent-for-linus' of git://git.kernel.org/.../tip/tip vm: add vm_iomap_memory() helper function Merge branch 'for-linus' of git://git.kernel.org/.../dtor/input Merge branch 'rc-fixes' of git://git.kernel.org/.../mmarek/kbuild Merge tag 'for-linus' of git://git.kernel.org/.../sfr/next-fixes Merge git://git.kernel.org/pub/scm/virt/kvm/kvm Merge branch 'fixes' of git://git.linaro.org/people/rmk/linux-arm s390: move dummy io_remap_pfn_range() to asm/pgtable.h Merge git://git.kernel.org/.../davem/net Merge branch 'for-linus' of git://git.kernel.org/.../sage/ceph-client Merge branch 'akpm' (fixes from Andrew) Merge branch 'release' of git://git.kernel.org/.../lenb/linux Revert "block: add missing block_bio_complete() tracepoint" Merge branch 'for_linus' of git://cavan.codon.org.uk/platform-drivers-x86 Merge branch 'userns-fixes' of git://git.kernel.org/.../luto/linux Merge branch 'for-linus' of git://git.kernel.org/.../mszeredi/fuse Merge branch 'for-linus' of git://git.kernel.org/.../dtor/input vm: convert HPET mmap to vm_iomap_memory() helper vm: convert mtdchar mmap to vm_iomap_memory() helper vm: convert fb_mmap to vm_iomap_memory() helper vm: convert snd_pcm_lib_mmap_iomem() to vm_iomap_memory() helper mtdchar: remove no-longer-used vma helpers Merge tag 'fixes-for-linus' of git://git.kernel.org/.../arm/arm-soc net: fix incorrect credentials passing Merge git://git.kernel.org/.../davem/net Merge git://git.kernel.org/.../davem/sparc Merge branch 'fixes' of git://git.linaro.org/people/rmk/linux-arm Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-kdump-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'vm_ioremap_memory-examples' Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Linux 3.9-rc8 Lucas Stach (1): net: fec: fix regression in link change accounting Ludovic Desroches (1): dmaengine: at_hdmac: fix race condition in atc_advance_work() Marc Kleine-Budde (1): can: mcp251x: add missing IRQF_ONESHOT to request_threaded_irq Marc Zyngier (3): ARM: KVM: fix KVM_CAP_ARM_SET_DEVICE_ADDR reporting ARM: KVM: fix L_PTE_S2_RDWR to actually be Read/Write ARM: KVM: fix unbalanced get_cpu() in access_dcsw Marcelo Tosatti (1): Merge branch 'kvm-arm-fixes-3.9' of git://github.com/columbia/linux-kvm-arm Mark Brown (76): ASoC: arizona: Only allow input volume updates when inputs are enabled ASoC: wm8960: Add input boost volume control ASoC: arizona: If we only have a clock to synchronise with make it REFCLK mfd: arizona: Define additional FLL control registers mfd: wm5102: Map in additional FLL control registers ASoC: arizona: Increase FLL synchroniser bandwidth for high frequencies ASoC: arizona: Optimise FLL loop gains ASoC: arizona: Ensure synchroniser is disabled when not needed ASoC: arizona: Suppress reference calculations when setting REFCLK to 0 ASoC: arizona: Provide defines for the clock rates ASoC: wm_adsp: Handle old .bin files ASoC: arizona: Ensure we clock two channels for I2S mode ASoC: wm8994: Support constraining the maximum number of channels clocked mfd: wm5102: Add additional speaker control registers ASoC: arizona: Factor out speaker widgets from CODEC drivers ASoC: arizona: Log thermal events ASoC: arizona: Suppress speaker enable if thermal shutdown is flagged ASoC: arizona: Basic support for ISRC rate selection ASoC: wm8994: Restore AIFnCLK after reducing it for low clock rates extcon: arizona: Factor out magic application ASoC: arizona: Fix interaction between headphone outputs and identification extcon: arizona: Fix interaction between headphone outputs and identification Merge tag 'arizona-extcon-asoc' of git://git.kernel.org/.../broonie/misc into asoc-arizona ASoC: arizona: Add delay after powering up line level outputs Merge branch 'topic/pxa' of git://git.kernel.org/.../broonie/sound into asoc-component ASoC: wm5102: Implement OSR support ASoC: wm2000: Expose some more registers for diagnostics ASoC: wm_adsp: Provide defines for firmwares ASoC: wm_adsp: Split ADSP1 and ADSP2 firmware controls Merge branch 'fix/samsung' of git://git.kernel.org/.../broonie/sound into asoc-samsung ASoC: core: Merge Samsung fixes to avoid trivial context conflict ASoC: wm8994: Remove duplicate revision cache ASoC: wm_adsp: Add support for firmware wide coefficient blocks Merge remote-tracking branch 'asoc/fix/compress' into asoc-next Merge remote-tracking branch 'asoc/fix/core' into asoc-next Merge remote-tracking branch 'asoc/fix/samsung' into asoc-next Merge remote-tracking branch 'asoc/fix/tegra' into asoc-next Merge remote-tracking branch 'asoc/fix/wm8903' into asoc-next Merge remote-tracking branch 'asoc/topic/adau1373' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4104' into asoc-next Merge remote-tracking branch 'asoc/topic/ak5386' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42l73' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/fsi' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/maintainers' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/si476x' into asoc-next Merge remote-tracking branch 'asoc/topic/spear' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Merge remote-tracking branch 'asoc/topic/wm-hubs' into asoc-next Merge remote-tracking branch 'asoc/topic/wm0010' into asoc-next Merge remote-tracking branch 'asoc/topic/wm2000' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8903' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8960' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Masami Hiramatsu (1): kprobes: Fix a double lock bug of kprobe_mutex Matt Fleming (1): x86, efivars: firmware bug workarounds should be in platform code Matthew Garrett (4): Move utf16 functions to kernel core and rename efi: Pass boot services variable info to runtime code efi: Distinguish between "remaining space" and actually used space Revert "hp-wmi: Add support for SMBus hotkeys" Michael Bohan (1): hrtimer: Don't reinitialize a cpu_base lock on CPU_UP Michael Neuling (1): powerpc: fix compiling CONFIG_PPC_TRANSACTIONAL_MEM when CONFIG_ALTIVEC=n Michal Bachraty (3): ASoC: davinci-mcasp: Add support for multichannel playback ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes Miklos Szeredi (1): fuse: fix type definitions in uapi header Mugunthan V N (1): drivers: net: ethernet: cpsw: get slave VLAN id from slave node instead of cpsw node Naoya Horiguchi (3): hugetlbfs: stop setting VM_DONTDUMP in initializing vma(VM_HUGETLB) fs/binfmt_elf.c: fix hugetlb memory check in vma_dump_size() hugetlbfs: add swap entry check in follow_hugetlb_page() Oleg Nesterov (1): thinkpad-acpi: kill hotkey_thread_mutex Patrick McHardy (1): netfilter: nf_ct_sip: don't drop packets with offsets pointing outside the packet Paul Bolle (2): ASoC: samsung: remove last traces of neo1973-gta01 ASoC: Ux500: remove test for undefined Kconfig macro Paul Handrigan (1): ASoC: cs42l73: If Internal MCLK is >= 6.4MHz, then set SCLK to 64*Fs. Ping Cheng (2): Input: wacom - DTH2242 Grip Pen id was off by one bit Input: wacom - fix "can not retrieve extra class descriptor" for DTH2242 Rafał Miłecki (1): ssb: implement spurious tone avoidance Richard Weinberger (2): x86,efi: Check max_size only if it is non-zero. x86,efi: Implement efi_no_storage_paranoia parameter Rob Herring (1): ARM: highbank: fix cache flush ordering for cpu hotplug Russell King (2): ARM: Do 15e0d9e37c (ARM: pm: let platforms select cpu_suspend support) properly Merge branch 'for-rmk/740t' of git://git.kernel.org/.../will/linux into fixes Ryo Tsutsui (2): ASoC: dapm: Remove redundant clear_walk() for supply widgets ASoC: dapm: Only clear paths we've walked Sachin Kamat (5): ASoC: max98090: Remove unneeded version.h header include ASoC: max98090: Convert to devm_regmap_init_i2c() ASoC: max98090: Make struct dev_pm_ops const ASoC: max98090: Fix checkpatch errors related to spacing ASoC: max98090: Remove unnecessary braces Sam Ravnborg (3): sparc: use generic headers bbc_i2c: fix section mismatch warning sparc: use asm-generic version of types.h Scott Wood (4): kvm/powerpc/e500mc: fix tlb invalidation on cpu migration kvm/ppc/e500: h2g_tlb1_rmap: esel 0 is valid kvm/ppc/e500: g2h_tlb1_map: clear old bit before setting new bit kvm/ppc/e500: eliminate tlb_refs Sergey Vlasov (2): x86/Kconfig: Make EFI select UCS2_STRING efi: Export efi_query_variable_store() for efivars.ko Shahed Shaikh (1): qlcnic: Fix typo in logs Shawn Guo (2): ASoC: generic-dmaengine-pcm: use a more common dma name ASoC: generic-dmaengine-pcm: call dma_request_slave_channel() Sritej Velaga (2): qlcnic: fix TSO race condition qlge: Fix receive path to drop error frames Stephane Eranian (1): perf/x86: Fix offcore_rsp valid mask for SNB/IVB Stephen Boyd (1): ARM: 7699/1: sched_clock: Add more notrace to prevent recursion Stephen Warren (10): ASoC: tegra: assume CONFIG_OF in tegra_asoc_utils_init ASoC: tegra_wm8753: minor cleanup ASoC: tegra_wm9712: assume CONFIG_OF ASoC: tegra_alc5632: assume CONFIG_OF, and other cleanup ASoC: tegra trimslice: assume CONFIG_OF, and other cleanup ASoC: tegra_wm8903: assume CONFIG_OF, remove platform data ASoC: tegra: add Tegra114 support to the AHUB driver ASoC: tegra: add Tegra114 support to tegra_asoc_utils.c ASoC: export snd_soc_register_component ASoC: dapm: Implement mixer control sharing Thomas Gleixner (1): kthread: Prevent unpark race which puts threads on the wrong cpu Thomas Graf (1): tcp: Reallocate headroom if it would overflow csum_start Thomas Petazzoni (1): net: mvmdio: add select PHYLIB Tim Gardner (2): ASoC: adau1373: adau1373_hw_params: Silence overflow warning ASoC: wm_hubs: Silence reg_r and reg_l 'may be used uninitialized' warnings Tommi Rantala (1): perf: Treat attr.config as u64 in perf_swevent_init() Vinod Koul (1): ASoC: mid-x86 - add support for meaadata apis Vyacheslav Dubeyko (1): hfsplus: fix potential overflow in hfsplus_file_truncate() Wei Yongjun (5): ASoC: tas5086: use module_i2c_driver to simplify the code ASoC: arizona: remove duplicated include from arizona.c tuntap: fix error return code in tun_set_iff() esp4: fix error return code in esp_output() pkt_sched: fix error return code in fw_change_attrs() Will Deacon (6): ARM: tlbflush: remove ARMv3 support ARM: cache: remove ARMv3 support code ARM: mm: fix numerous hideous errors in proc-arm740.S ARM: mm: remove broken condition check for v4 flushing ARM: modules: don't export cpu_set_pte_ext when !MMU ARM: 7698/1: perf: fix group validation when using enable_on_exec Willy Tarreau (1): net: mvneta: fix improper tx queue usage in mvneta_tx() Xishi Qiu (1): mm/vmscan: fix error return in kswapd_run() Yegor Yefremov (2): ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode ASoC: davinci-mcasp: don't overwrite DIT settings Yinghai Lu (4): x86, kdump: Set crashkernel_low automatically x86, kdump: Retore crashkernel= to allocate under 896M x86, kdump: Change crashkernel_high/low= to crashkernel=,high/low kexec: use Crash kernel for Crash kernel low Zefir Kurtisi (1): ath9k: fix DFS detector called in softirq context Zoltan Puskas (1): ASoC: atmel: Add slave mode support to SSC in DSP Mode A nikolay(a)redhat.com (7): bonding: fix netdev event NULL pointer dereference bonding: IFF_BONDING is not stripped on enslave failure bonding: mc addresses don't get deleted on enslave failure bonding: vlans don't get deleted on enslave failure bonding: primary_slave & curr_active_slave are not cleaned on enslave failure bonding: disable netpoll on enslave failure bonding: in bond_mc_swap() bond's mc addr list is walked without lock stephen hemminger (1): bridge: make user modified path cost sticky Documentation/devicetree/bindings/sound/ak5386.txt | 19 + .../bindings/sound/nvidia,tegra30-ahub.txt | 26 +- .../devicetree/bindings/sound/ti,tas5086.txt | 32 ++ Documentation/kernel-parameters.txt | 29 +- Makefile | 5 +- arch/arm/include/asm/glue-cache.h | 8 - arch/arm/include/asm/hardware/iop3xx.h | 2 +- arch/arm/include/asm/pgtable-3level.h | 2 +- arch/arm/include/asm/tlbflush.h | 11 +- arch/arm/kernel/hw_breakpoint.c | 2 +- arch/arm/kernel/perf_event.c | 5 +- arch/arm/kernel/sched_clock.c | 4 +- arch/arm/kernel/setup.c | 3 - arch/arm/kernel/tcm.c | 1 - arch/arm/kvm/arm.c | 1 + arch/arm/kvm/coproc.c | 4 +- arch/arm/mach-highbank/hotplug.c | 10 +- arch/arm/mach-s3c24xx/dma-s3c2410.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2412.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2440.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2443.c | 2 - arch/arm/mm/Kconfig | 5 +- arch/arm/mm/Makefile | 1 - arch/arm/mm/cache-feroceon-l2.c | 1 + arch/arm/mm/cache-v3.S | 137 ----- arch/arm/mm/cache-v4.S | 2 +- arch/arm/mm/mmu.c | 2 + arch/arm/mm/proc-arm740.S | 30 +- arch/arm/mm/proc-arm920.S | 2 +- arch/arm/mm/proc-arm926.S | 2 +- arch/arm/mm/proc-mohawk.S | 2 +- arch/arm/mm/proc-sa1100.S | 2 +- arch/arm/mm/proc-syms.c | 2 + arch/arm/mm/proc-v6.S | 2 +- arch/arm/mm/proc-xsc3.S | 2 +- arch/arm/mm/proc-xscale.S | 2 +- arch/arm/{kernel => mm}/tcm.h | 0 arch/arm/plat-samsung/devs.c | 6 + arch/avr32/include/asm/io.h | 4 + arch/powerpc/kernel/entry_64.S | 4 +- arch/powerpc/kernel/process.c | 2 + arch/powerpc/kernel/signal_32.c | 2 + arch/powerpc/kernel/signal_64.c | 2 + arch/powerpc/kernel/tm.S | 2 + arch/powerpc/kvm/e500.h | 24 +- arch/powerpc/kvm/e500_mmu_host.c | 84 +-- arch/powerpc/kvm/e500mc.c | 7 +- arch/s390/include/asm/io.h | 4 - arch/s390/include/asm/pgtable.h | 4 + arch/sparc/include/asm/Kbuild | 5 + arch/sparc/include/asm/cputime.h | 6 - arch/sparc/include/asm/emergency-restart.h | 6 - arch/sparc/include/asm/mutex.h | 9 - arch/sparc/include/asm/pgtable_64.h | 1 + arch/sparc/include/asm/serial.h | 6 - arch/sparc/include/asm/smp_32.h | 5 - arch/sparc/include/asm/switch_to_64.h | 3 +- arch/sparc/include/asm/tlbflush_64.h | 37 +- arch/sparc/include/uapi/asm/Kbuild | 1 - arch/sparc/include/uapi/asm/types.h | 17 - arch/sparc/kernel/smp_64.c | 41 +- arch/sparc/lib/bitext.c | 6 +- arch/sparc/mm/iommu.c | 2 +- arch/sparc/mm/srmmu.c | 4 +- arch/sparc/mm/tlb.c | 38 +- arch/sparc/mm/tsb.c | 57 +- arch/sparc/mm/ultra.S | 119 +++- arch/x86/Kconfig | 1 + arch/x86/boot/compressed/eboot.c | 47 ++ arch/x86/include/asm/efi.h | 7 + arch/x86/include/uapi/asm/bootparam.h | 1 + arch/x86/kernel/cpu/mshyperv.c | 18 +- arch/x86/kernel/cpu/perf_event_intel.c | 20 +- arch/x86/kernel/microcode_core_early.c | 38 +- arch/x86/kernel/setup.c | 45 +- arch/x86/platform/efi/efi.c | 168 +++++- block/blk-core.c | 1 + drivers/block/rbd.c | 3 +- drivers/char/hpet.c | 14 +- drivers/dma/at_hdmac.c | 9 +- drivers/extcon/extcon-arizona.c | 107 ++-- drivers/firmware/Kconfig | 1 + drivers/firmware/efivars.c | 98 +--- drivers/idle/intel_idle.c | 1 + drivers/input/tablet/wacom_wac.c | 8 +- drivers/irqchip/irq-gic.c | 3 +- drivers/md/dm.c | 1 + drivers/md/raid5.c | 11 +- drivers/mfd/wm5102-tables.c | 10 + drivers/mtd/mtdchar.c | 59 +- drivers/net/bonding/bond_main.c | 90 ++- drivers/net/can/mcp251x.c | 10 +- drivers/net/can/sja1000/sja1000_of_platform.c | 31 +- drivers/net/ethernet/8390/ax88796.c | 2 +- drivers/net/ethernet/broadcom/bnx2x/bnx2x_cmn.c | 7 +- drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c | 4 + drivers/net/ethernet/emulex/benet/be_main.c | 5 +- drivers/net/ethernet/freescale/fec.c | 1 + drivers/net/ethernet/intel/igb/igb.h | 8 - drivers/net/ethernet/intel/igb/igb_main.c | 110 +--- drivers/net/ethernet/intel/ixgbe/ixgbe_sriov.c | 6 + drivers/net/ethernet/marvell/Kconfig | 2 +- drivers/net/ethernet/marvell/mvneta.c | 9 +- .../net/ethernet/qlogic/qlcnic/qlcnic_83xx_hw.c | 15 +- drivers/net/ethernet/qlogic/qlcnic/qlcnic_io.c | 3 +- drivers/net/ethernet/qlogic/qlcnic/qlcnic_sysfs.c | 4 +- drivers/net/ethernet/qlogic/qlge/qlge.h | 2 +- drivers/net/ethernet/qlogic/qlge/qlge_ethtool.c | 2 +- drivers/net/ethernet/qlogic/qlge/qlge_main.c | 36 +- drivers/net/ethernet/stmicro/stmmac/mmc_core.c | 1 + drivers/net/ethernet/ti/cpsw.c | 2 +- drivers/net/tun.c | 2 +- drivers/net/usb/cdc_mbim.c | 2 +- drivers/net/usb/qmi_wwan.c | 104 ++++ .../net/wireless/ath/ath9k/ar9580_1p0_initvals.h | 2 +- .../net/wireless/ath/ath9k/dfs_pattern_detector.c | 4 +- drivers/net/wireless/ath/ath9k/dfs_pri_detector.c | 4 +- drivers/net/wireless/ath/ath9k/htc_drv_init.c | 2 +- drivers/net/wireless/b43/phy_n.c | 3 +- .../net/wireless/brcm80211/brcmfmac/wl_cfg80211.c | 7 +- .../net/wireless/brcm80211/brcmsmac/mac80211_if.c | 264 ++++----- drivers/platform/x86/hp-wmi.c | 4 - drivers/platform/x86/thinkpad_acpi.c | 10 - drivers/sbus/char/bbc_i2c.c | 4 +- drivers/ssb/driver_chipcommon_pmu.c | 29 + drivers/video/fbmem.c | 39 +- drivers/video/mmp/core.c | 2 - fs/binfmt_elf.c | 1 + fs/bio.c | 2 - fs/hfsplus/extents.c | 2 +- fs/hugetlbfs/inode.c | 2 +- fs/proc/array.c | 1 + include/linux/blktrace_api.h | 1 - include/linux/efi.h | 9 +- include/linux/kexec.h | 2 + include/linux/mfd/arizona/core.h | 3 + include/linux/mfd/arizona/registers.h | 42 ++ include/linux/mfd/wm8994/pdata.h | 8 + include/linux/mm.h | 2 + include/linux/netfilter/ipset/ip_set_ahash.h | 30 +- include/linux/sched.h | 5 +- include/linux/ssb/ssb_driver_chipcommon.h | 2 + include/linux/swiotlb.h | 1 + include/linux/ucs2_string.h | 14 + include/net/addrconf.h | 1 + include/net/irda/irlmp.h | 3 +- include/net/scm.h | 4 +- include/sound/dmaengine_pcm.h | 5 + include/sound/soc-dai.h | 8 - include/sound/soc.h | 20 + include/sound/tas5086.h | 7 + include/sound/tegra_wm8903.h | 26 - include/trace/events/block.h | 8 +- include/trace/events/sched.h | 2 +- include/uapi/linux/fuse.h | 436 +++++++-------- kernel/events/core.c | 2 +- kernel/hrtimer.c | 3 +- kernel/kexec.c | 118 +++- kernel/kprobes.c | 19 +- kernel/kthread.c | 52 +- kernel/signal.c | 2 +- kernel/smpboot.c | 14 +- kernel/trace/blktrace.c | 26 +- kernel/user_namespace.c | 22 +- lib/Kconfig | 3 + lib/Makefile | 2 + lib/swiotlb.c | 19 +- lib/ucs2_string.c | 51 ++ mm/hugetlb.c | 12 +- mm/memory.c | 47 ++ mm/vmscan.c | 2 +- net/802/mrp.c | 4 + net/batman-adv/main.c | 5 +- net/batman-adv/main.h | 2 +- net/batman-adv/routing.c | 38 +- net/batman-adv/translation-table.c | 2 +- net/batman-adv/vis.c | 4 +- net/bridge/br_if.c | 3 +- net/bridge/br_private.h | 1 + net/bridge/br_stp_if.c | 1 + net/core/dev.c | 3 + net/ipv4/esp4.c | 6 +- net/ipv4/ip_fragment.c | 14 +- net/ipv4/netfilter/ipt_rpfilter.c | 8 +- net/ipv4/syncookies.c | 4 +- net/ipv4/tcp_input.c | 64 +-- net/ipv4/tcp_output.c | 8 +- net/ipv6/addrconf.c | 24 +- net/ipv6/addrconf_core.c | 19 + net/ipv6/netfilter/ip6t_rpfilter.c | 8 +- net/ipv6/reassembly.c | 12 +- net/irda/iriap.c | 3 +- net/irda/irlmp.c | 10 +- net/mac80211/iface.c | 27 +- net/mac80211/mlme.c | 24 +- net/netfilter/ipset/ip_set_bitmap_ipmac.c | 6 +- net/netfilter/ipset/ip_set_hash_ipportnet.c | 18 + net/netfilter/ipset/ip_set_hash_net.c | 22 +- net/netfilter/ipset/ip_set_hash_netiface.c | 22 +- net/netfilter/ipset/ip_set_hash_netport.c | 18 + net/netfilter/ipset/ip_set_list_set.c | 10 +- net/netfilter/nf_conntrack_sip.c | 6 +- net/netfilter/nf_nat_core.c | 40 +- net/openvswitch/datapath.c | 30 +- net/openvswitch/flow.c | 2 +- net/sched/cls_fw.c | 2 +- scripts/checkpatch.pl | 1 + sound/core/pcm_native.c | 12 +- sound/soc/atmel/atmel_ssc_dai.c | 54 +- sound/soc/au1x/ac97c.c | 9 +- sound/soc/au1x/i2sc.c | 9 +- sound/soc/au1x/psc-ac97.c | 9 +- sound/soc/au1x/psc-i2s.c | 9 +- sound/soc/blackfin/bf5xx-ac97.c | 9 +- sound/soc/blackfin/bf5xx-i2s.c | 9 +- sound/soc/blackfin/bf5xx-tdm.c | 9 +- sound/soc/blackfin/bf6xx-i2s.c | 9 +- sound/soc/cirrus/ep93xx-ac97.c | 9 +- sound/soc/cirrus/ep93xx-i2s.c | 9 +- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/adau1373.c | 5 +- sound/soc/codecs/ak4104.c | 55 +- sound/soc/codecs/ak5386.c | 152 +++++ sound/soc/codecs/arizona.c | 530 +++++++++++++++--- sound/soc/codecs/arizona.h | 34 +- sound/soc/codecs/cs4271.c | 166 +++--- sound/soc/codecs/cs42l73.c | 6 +- sound/soc/codecs/max98088.c | 32 +- sound/soc/codecs/max98090.c | 45 +- sound/soc/codecs/si476x.c | 47 +- sound/soc/codecs/tas5086.c | 591 ++++++++++++++++++++ sound/soc/codecs/wm0010.c | 6 +- sound/soc/codecs/wm2000.c | 4 +- sound/soc/codecs/wm2000.h | 2 + sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5102.c | 129 ++--- sound/soc/codecs/wm5102.h | 6 +- sound/soc/codecs/wm5110.c | 54 +- sound/soc/codecs/wm5110.h | 6 +- sound/soc/codecs/wm8903.c | 4 + sound/soc/codecs/wm8960.c | 10 + sound/soc/codecs/wm8994.c | 68 ++- sound/soc/codecs/wm8994.h | 3 +- sound/soc/codecs/wm_adsp.c | 124 +++- sound/soc/codecs/wm_adsp.h | 5 +- sound/soc/codecs/wm_hubs.c | 9 +- sound/soc/davinci/davinci-i2s.c | 15 +- sound/soc/davinci/davinci-mcasp.c | 110 ++-- sound/soc/davinci/davinci-mcasp.h | 2 +- sound/soc/davinci/davinci-pcm.c | 9 +- sound/soc/davinci/davinci-vcif.c | 11 +- sound/soc/dwc/designware_i2s.c | 9 +- sound/soc/fsl/fsl_ssi.c | 11 +- sound/soc/fsl/fsl_ssi.h | 8 + sound/soc/fsl/imx-audmux.c | 3 +- sound/soc/fsl/imx-pcm-fiq.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/imx-ssi.c | 13 +- sound/soc/fsl/mpc5200_psc_ac97.c | 8 +- sound/soc/fsl/mpc5200_psc_i2s.c | 9 +- sound/soc/jz4740/jz4740-i2s.c | 9 +- sound/soc/kirkwood/kirkwood-i2s.c | 11 +- sound/soc/mid-x86/sst_platform.c | 20 +- sound/soc/mid-x86/sst_platform.h | 2 + sound/soc/mxs/Kconfig | 2 +- sound/soc/mxs/mxs-pcm.c | 139 +---- sound/soc/mxs/mxs-saif.c | 14 +- sound/soc/nuc900/nuc900-ac97.c | 9 +- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-hdmi.c | 9 +- sound/soc/omap/omap-mcbsp.c | 9 +- sound/soc/omap/omap-mcpdm.c | 9 +- sound/soc/pxa/mmp-sspa.c | 9 +- sound/soc/pxa/pxa-ssp.c | 9 +- sound/soc/pxa/pxa2xx-ac97.c | 11 +- sound/soc/pxa/pxa2xx-i2s.c | 9 +- sound/soc/s6000/s6000-i2s.c | 9 +- sound/soc/samsung/Kconfig | 5 +- sound/soc/samsung/ac97.c | 14 +- sound/soc/samsung/goni_wm8994.c | 11 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 16 +- sound/soc/samsung/idma.c | 11 +- sound/soc/samsung/neo1973_wm8753.c | 3 +- sound/soc/samsung/pcm.c | 11 +- .../include/plat => sound/soc/samsung}/regs-ac97.h | 0 .../include/plat => sound/soc/samsung}/regs-iis.h | 0 sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 9 +- sound/soc/samsung/s3c-i2s-v2.h | 7 +- sound/soc/samsung/s3c2412-i2s.c | 12 +- sound/soc/samsung/s3c24xx-i2s.c | 13 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- sound/soc/samsung/spdif.c | 11 +- sound/soc/sh/fsi.c | 14 +- sound/soc/sh/hac.c | 10 +- sound/soc/sh/migor.c | 2 +- sound/soc/sh/siu_dai.c | 11 +- sound/soc/sh/ssi.c | 10 +- sound/soc/soc-compress.c | 27 + sound/soc/soc-core.c | 99 +++- sound/soc/soc-dapm.c | 319 ++++++----- sound/soc/soc-generic-dmaengine-pcm.c | 44 +- sound/soc/soc-utils.c | 25 + sound/soc/spear/spdif_in.c | 9 +- sound/soc/spear/spdif_out.c | 9 +- sound/soc/spear/spear_pcm.c | 4 +- sound/soc/tegra/tegra20_ac97.c | 15 +- sound/soc/tegra/tegra20_i2s.c | 15 +- sound/soc/tegra/tegra20_spdif.c | 15 +- sound/soc/tegra/tegra30_ahub.c | 64 ++- sound/soc/tegra/tegra30_ahub.h | 16 + sound/soc/tegra/tegra30_i2s.c | 15 +- sound/soc/tegra/tegra_alc5632.c | 15 +- sound/soc/tegra/tegra_asoc_utils.c | 34 +- sound/soc/tegra/tegra_asoc_utils.h | 1 + sound/soc/tegra/tegra_wm8753.c | 15 +- sound/soc/tegra/tegra_wm8903.c | 179 +++--- sound/soc/tegra/tegra_wm9712.c | 7 +- sound/soc/tegra/trimslice.c | 56 +- sound/soc/txx9/txx9aclc-ac97.c | 9 +- sound/soc/ux500/ux500_msp_dai.c | 13 +- sound/soc/ux500/ux500_msp_dai.h | 5 - sound/soc/ux500/ux500_msp_i2s.h | 1 + tools/power/x86/turbostat/turbostat.c | 5 +- 326 files changed, 5225 insertions(+), 2620 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak5386.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,tas5086.txt delete mode 100644 arch/arm/mm/cache-v3.S rename arch/arm/{kernel => mm}/tcm.h (100%) delete mode 100644 arch/sparc/include/asm/cputime.h delete mode 100644 arch/sparc/include/asm/emergency-restart.h delete mode 100644 arch/sparc/include/asm/mutex.h delete mode 100644 arch/sparc/include/asm/serial.h delete mode 100644 arch/sparc/include/uapi/asm/types.h create mode 100644 include/linux/ucs2_string.h create mode 100644 include/sound/tas5086.h delete mode 100644 include/sound/tegra_wm8903.h create mode 100644 lib/ucs2_string.c create mode 100644 sound/soc/codecs/ak5386.c create mode 100644 sound/soc/codecs/tas5086.c rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-ac97.h (100%) rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-iis.h (100%)

2 1

Re: [alsa-devel] [PATCH] drm/i915: Add private api for power well usage
by Wang, Xingchao 25 Apr '13

25 Apr '13

Hi Daniel/Paulo, Any comments on this? Add Jesse and Rafael in loop. thanks --xingchao > -----Original Message----- > From: Wang, Xingchao > Sent: Wednesday, April 24, 2013 3:29 PM > To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R; 'Takashi Iwai' > Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; Li, Jocelyn; > intel-gfx(a)lists.freedesktop.org; alsa-devel(a)alsa-project.org; 'Wang Xingchao' > Subject: RE: [PATCH] drm/i915: Add private api for power well usage > > Hi Guys, > > Sorry I should Add Takashi and alsa maillist in loop. > > I tested this patch against Daniel's "drm-intel-next-queued" Branch, on > Haswell Mobile C3 stepping machines. > > Test steps: > 1. only connect one monitor through HDMI cable 2. echo 1 > > /sys/module/i915/parameters/disable_power_well ==> I think this could be set > as 1 by default if applied my patch. > 3. plug out HDMI cable > 4. cat /proc/asound/card0/codec#0 > > At step 3, i915 will try to disable power well but rejected as display audio is > using power well. > Without the patch, display audio would crash at step 4. > > I did not list suspend/resume test here, as during suspend sequence, i915 will > _ENABLE_ power well, that's a BUG? > > For me above test case is the only one to reproduce the issue, if you have more > ideas and need more test, please let me know. > > I'm afraid there's some risk because the dependency between alsa driver and > drm driver. For example, if i915 module was not loaded, display audio Will call > request_power_well() and meet symbol error. Even if i915 module loaded after > snd-hda-intel module, there's such risk too. Any comment on this? > > Another risk is the pointer "power_well_device", it should be set NULL when > i915 module unloaded, I added the caller but have no idea where to call it, if > anyone has suggestion, that would be fine. > > thanks > --xingchao > > > > -----Original Message----- > > From: Wang Xingchao [mailto:xingchao.wang@linux.intel.com] > > Sent: Thursday, April 24, 2014 10:20 PM > > To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R > > Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; > > Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; Wang, Xingchao; Wang > > Xingchao > > Subject: [PATCH] drm/i915: Add private api for power well usage > > > > Display audio depends on power well in graphic side, it should request > > power well before use it and release power well after use. > > I915 will not shutdown power well if it detects audio is using. > > This patch protects display audio crash for Intel Haswell mobile > > C3 stepping board. > > > > Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> > > --- > > drivers/gpu/drm/i915/intel_pm.c | 73 > > +++++++++++++++++++++++++++++++++++---- > > include/drm/audio_drm.h | 39 +++++++++++++++++++++ > > sound/pci/hda/hda_intel.c | 8 +++++ > > 3 files changed, 113 insertions(+), 7 deletions(-) create mode > > 100644 include/drm/audio_drm.h > > > > diff --git a/drivers/gpu/drm/i915/intel_pm.c > > b/drivers/gpu/drm/i915/intel_pm.c index 2557926..9983f56 100644 > > --- a/drivers/gpu/drm/i915/intel_pm.c > > +++ b/drivers/gpu/drm/i915/intel_pm.c > > @@ -4298,18 +4298,12 @@ bool intel_using_power_well(struct drm_device > > *dev) > > return true; > > } > > > > -void intel_set_power_well(struct drm_device *dev, bool enable) > > +void set_power_well(struct drm_device *dev, bool enable) > > { > > struct drm_i915_private *dev_priv = dev->dev_private; > > bool is_enabled, enable_requested; > > uint32_t tmp; > > > > - if (!HAS_POWER_WELL(dev)) > > - return; > > - > > - if (!i915_disable_power_well && !enable) > > - return; > > - > > tmp = I915_READ(HSW_PWR_WELL_DRIVER); > > is_enabled = tmp & HSW_PWR_WELL_STATE; > > enable_requested = tmp & HSW_PWR_WELL_ENABLE; @@ -4332,6 > > +4326,71 @@ void intel_set_power_well(struct drm_device *dev, bool > > +enable) > > } > > } > > > > +/* Global drm_device for display audio drvier usage */ static struct > > +drm_device *power_well_device; static bool audio_power_well_using, > > +i915_power_well_using; > > +/* Lock protecting power well setting */ > > +DEFINE_SPINLOCK(powerwell_lock); > > + > > +void request_power_well(void) > > +{ > > + DRM_DEBUG_KMS("Display audio requesting power well\n"); > > + spin_lock_irq(&powerwell_lock); > > + if (!power_well_device) > > + goto out; > > + if (i915_power_well_using == false) > > + set_power_well(power_well_device, true); > > + audio_power_well_using = true; > > +out: > > + spin_unlock_irq(&powerwell_lock); > > + return; > > +} > > +EXPORT_SYMBOL_GPL(request_power_well); > > + > > +void release_power_well(void) > > +{ > > + DRM_DEBUG_KMS("Display audio releasing power well\n"); > > + spin_lock_irq(&powerwell_lock); > > + if (!power_well_device) > > + goto out; > > + if (i915_power_well_using == false) > > + set_power_well(power_well_device, false); > > + audio_power_well_using = false; > > +out: > > + spin_unlock_irq(&powerwell_lock); > > + return; > > +} > > +EXPORT_SYMBOL_GPL(release_power_well); > > + > > +/* TODO: Call this when i915 module unload */ void > > +remove_power_well(void) { > > + spin_lock_irq(&powerwell_lock); > > + audio_power_well_using = false; > > + i915_power_well_using = false; > > + power_well_device = NULL; > > + spin_unlock_irq(&powerwell_lock); > > +} > > + > > +void intel_set_power_well(struct drm_device *dev, bool enable) { > > + if (!HAS_POWER_WELL(dev)) > > + return; > > + > > + spin_lock_irq(&powerwell_lock); > > + power_well_device = dev; > > + i915_power_well_using = enable; > > + if (!enable && audio_power_well_using) > > + goto out; > > + > > + if (!i915_disable_power_well && !enable) > > + goto out; > > + set_power_well(dev, enable); > > +out: > > + spin_unlock_irq(&powerwell_lock); > > + return; > > +} > > + > > /* > > * Starting with Haswell, we have a "Power Down Well" that can be turned > off > > * when not needed anymore. We have 4 registers that can request the > > power well diff --git a/include/drm/audio_drm.h > > b/include/drm/audio_drm.h new file mode 100644 index 0000000..4412b36 > > --- /dev/null > > +++ b/include/drm/audio_drm.h > > @@ -0,0 +1,39 @@ > > > +/************************************************************** > > ******** > > +**** > > + * > > + * Copyright 2013 Intel Inc. > > + * All Rights Reserved. > > + * > > + * Permission is hereby granted, free of charge, to any person > > +obtaining a > > + * copy of this software and associated documentation files (the > > + * "Software"), to deal in the Software without restriction, > > +including > > + * without limitation the rights to use, copy, modify, merge, > > +publish, > > + * distribute, sub license, and/or sell copies of the Software, and > > +to > > + * permit persons to whom the Software is furnished to do so, subject > > +to > > + * the following conditions: > > + * > > + * The above copyright notice and this permission notice (including > > +the > > + * next paragraph) shall be included in all copies or substantial > > +portions > > + * of the Software. > > + * > > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY > KIND, > > +EXPRESS OR > > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF > > +MERCHANTABILITY, > > + * FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT. IN NO > > EVENT > > +SHALL > > + * THE COPYRIGHT HOLDERS, AUTHORS AND/OR ITS SUPPLIERS BE LIABLE > > FOR > > +ANY CLAIM, > > + * DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, > > TORT > > +OR > > + * OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE > > SOFTWARE > > +OR THE > > + * USE OR OTHER DEALINGS IN THE SOFTWARE. > > + * > > + * > > + > > > +*************************************************************** > > ******** > > +***/ > > +/* > > + * Authors: > > + * Wang Xingchao <xingchao.wang(a)intel.com> */ > > + > > +#ifndef _AUDIO_DRM_H_ > > +#define _AUDIO_DRM_H_ > > + > > +extern void request_power_well(void); extern void > > +release_power_well(void); > > + > > +#endif /* _AUDIO_DRM_H_ */ > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > > index 418bfc0..59a85de 100644 > > --- a/sound/pci/hda/hda_intel.c > > +++ b/sound/pci/hda/hda_intel.c > > @@ -50,6 +50,7 @@ > > #include <linux/clocksource.h> > > #include <linux/time.h> > > #include <linux/completion.h> > > +#include <drm/audio_drm.h> > > > > #ifdef CONFIG_X86 > > /* for snoop control */ > > @@ -2869,6 +2870,7 @@ static int azx_suspend(struct device *dev) > > pci_disable_device(pci); > > pci_save_state(pci); > > pci_set_power_state(pci, PCI_D3hot); > > + release_power_well(); > > return 0; > > } > > > > @@ -2881,6 +2883,7 @@ static int azx_resume(struct device *dev) > > if (chip->disabled) > > return 0; > > > > + request_power_well(); > > pci_set_power_state(pci, PCI_D0); > > pci_restore_state(pci); > > if (pci_enable_device(pci) < 0) { > > @@ -2913,6 +2916,7 @@ static int azx_runtime_suspend(struct device > > *dev) > > > > azx_stop_chip(chip); > > azx_clear_irq_pending(chip); > > + release_power_well(); > > return 0; > > } > > > > @@ -2921,6 +2925,7 @@ static int azx_runtime_resume(struct device *dev) > > struct snd_card *card = dev_get_drvdata(dev); > > struct azx *chip = card->private_data; > > > > + request_power_well(); > > azx_init_pci(chip); > > azx_init_chip(chip, 1); > > return 0; > > @@ -3671,6 +3676,8 @@ static int azx_probe(struct pci_dev *pci, > > return -ENOENT; > > } > > > > + request_power_well(); > > + > > err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); > > if (err < 0) { > > snd_printk(KERN_ERR "hda-intel: Error creating card!\n"); @@ > > -3806,6 +3813,7 @@ static void azx_remove(struct pci_dev *pci) > > if (card) > > snd_card_free(card); > > pci_set_drvdata(pci, NULL); > > + release_power_well(); > > } > > > > /* PCI IDs */ > > -- > > 1.7.9.5

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[alsa-devel] code improvements in /sound/usb/mixer.c
by Daniel Schürmann 25 Apr '13

25 Apr '13

Hi Takashi, Hi alsa-devel. in my last mail I had supposed scaling issues with mixer.c, but finally it turns out that was just a debugging problem of myself. I was struggled how types are handled in the code. I have some ideas for improvement but I am not sure if it its worth the work since nothing is broken. 1. get rid of convert_bytes_value(cval, value_set): This function return any type in a integer container, which is confusing. 2. use type casts in convert_signed_value(): gcc knows perfecly how to convert types. 3. merge convert_signed_value() and snd_usb_combine_bytes() to get rid of the intermediate integer variant. --- I am currently working on a patch that introduces a get_ctl_value_range_v2. I want to get rid of the three single queries for min max and rate. Also here, nothing broken. Is it worth to do the work? Kind regards Daniel

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[alsa-devel] [PATCH v2] ALSA: snd-usb-audio: fixed mixer range reading during probing, when compiled with CONFIG_PM defined
by Daniel Schürmann 25 Apr '13

25 Apr '13

During probing, snd_usb_autoresume() fails with -ENODEV. This brakes get_ctl_value() in mixer.c when reading the range values. In the new version snd_usb_autoresume() returns 0 in this case which brakes not the following query. The following snd_usb_suspend() fails silently during probing anyway. The patch also makes the error messages more significant. The original error messages where identified as misleading during debug. Signed-off-by: Daniel Schürmann <daschuer(a)mixxx.org> --- sound/usb/card.c | 7 +++++-- sound/usb/mixer.c | 11 +++++++---- 2 files changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 5254b18..2f55d00 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -629,9 +629,12 @@ static void usb_audio_disconnect(struct usb_interface *intf) int snd_usb_autoresume(struct snd_usb_audio *chip) { int err = -ENODEV; - + + if (chip->probing) { + return 0; + } down_read(&chip->shutdown_rwsem); - if (!chip->shutdown && !chip->probing) + if (!chip->shutdown) err = usb_autopm_get_interface(chip->pm_intf); up_read(&chip->shutdown_rwsem); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ca4739c..5d4f3c6 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -333,9 +333,12 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v memset(buf, 0, sizeof(buf)); - ret = snd_usb_autoresume(chip) ? -EIO : 0; - if (ret) + ret = snd_usb_autoresume(chip); + if (ret) { + snd_printk(KERN_ERR "snd_usb_autoresume() failed with %d\n", ret); + ret = -EIO; goto error; + } down_read(&chip->shutdown_rwsem); if (chip->shutdown) @@ -351,8 +354,8 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v if (ret < 0) { error: - snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", - request, validx, idx, cval->val_type); + snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, ret = %d\n", + bRequest, validx, idx, cval->val_type, ret); return ret; } -- 1.7.9.5

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