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

[alsa-devel] [RFC PATCH 0/4] snd-ice1712: Add Philips PSC724 Ultimate Edge
by Ondrej Zary 18 Apr '12

18 Apr '12

Hello, this is first attempt to add Philips PSC724 Ultimate Edge sound card support to snd-ice1712 driver. Unlike other snd-ice1712 subdrivers, the codec code is splitted into separate modules and thus reusable. Working: all analog outputs (front, rear, center+lfe, headphone) and inputs (front mic, rear mic, line, cd, aux), mixer, headphone jack detection Currently untested: SPDIF, real multichannel sounds, power management Buggy: switching AGC mode in mixer causes alsamixer to crash - looks like it does not like notifications about control activating/deactivating

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[alsa-devel] [PATCH forward] ASoC: fsi: update for dmaengine prep_slave_sg fallout.
by kuninori.morimoto.gx＠renesas.com 18 Apr '12

18 Apr '12

From: Paul Mundt <lethal(a)linux-sh.org> Leading up to the ->device_prep_slave_sg change in 185ecb5f4fd43911c35956d4cc7d94a1da30417f 'dmaengine: add context parameter to prep_slave_sg and prep_dma_cyclic' a generic wrapper was added in place to guard against the API change, though the fsi driver wasn't updated in the process (presumably its dmaengine support hadn't been merged yet at the time). This trivially switches over to the new wrapper and gets it building again. Signed-off-by: Paul Mundt <lethal(a)linux-sh.org> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Nothing happened on ALSA ML. So I forward this patch. I added my Acked-by on it. sound/soc/sh/fsi.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 378cc5b..74ed2df 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1001,11 +1001,10 @@ static void fsi_dma_do_tasklet(unsigned long data) sg_dma_address(&sg) = buf; sg_dma_len(&sg) = len; - desc = chan->device->device_prep_slave_sg(chan, &sg, 1, dir, - DMA_PREP_INTERRUPT | - DMA_CTRL_ACK); + desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { - dev_err(dai->dev, "device_prep_slave_sg() fail\n"); + dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); return; }

2 1

[alsa-devel] [PATCH] ALSA: Add definitions for CEA-861 Audio InfoFrames
by Ricardo Neri 18 Apr '12

18 Apr '12

Along with the IEC-60958 channel status word, CEA-861 Audio InfoFrames are used in HDMI and DisplayPort to describe the parameters of the audio stream. Hence, drivers for such devices may use these definitions to, for instance, fill a CEA-861 data structure and pass it to a display driver to configure an IP. Signed-off-by: Ricardo Neri <ricardo.neri(a)ti.com> --- include/sound/asound.h | 14 ++++++++++++++ include/sound/asoundef.h | 41 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 55 insertions(+), 0 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index a2e4ff5..0876a1e 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -70,6 +70,20 @@ struct snd_aes_iec958 { /**************************************************************************** * * + * CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort * + * * + ****************************************************************************/ + +struct snd_cea_861_aud_if { + unsigned char db1_ct_cc; /* coding type and channel count */ + unsigned char db2_sf_ss; /* sample frequency and size */ + unsigned char db3; /* not used, all zeros */ + unsigned char db4_ca; /* channel allocation code */ + unsigned char db5_dminh_lsv; /* downmix inhibit & level-shit values */ +}; + +/**************************************************************************** + * * * Section for driver hardware dependent interface - /dev/snd/hw? * * * ****************************************************************************/ diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h index 20ebf32..bb05c02 100644 --- a/include/sound/asoundef.h +++ b/include/sound/asoundef.h @@ -170,6 +170,47 @@ #define IEC958_AES5_CON_CGMSA_COPYNOMORE (2<<0) /* condition not be used */ #define IEC958_AES5_CON_CGMSA_COPYNEVER (3<<0) /* no copying is permitted */ +/**************************************************************************** + * * + * CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort * + * * + ****************************************************************************/ +#define CEA861_AUDIO_INFOFRAME_DB1CC (7<<0) /* mask - channel count */ +#define CEA861_AUDIO_INFOFRAME_DB1CT (0xf<<4) /* mask - coding type */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM (0<<4) /* refer to stream */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_IEC60958 (1<<4) /* IEC-60958 L-PCM */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_AC3 (2<<4) /* AC-3 */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG1 (3<<4) /* MPEG1 Layers 1 & 2 */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_MP3 (4<<4) /* MPEG1 Layer 3 */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG2_MULTICH (5<<4) /* MPEG2 Multichannel */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_AAC (6<<4) /* AAC */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS (7<<4) /* DTS */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_ATRAC (8<<4) /* ATRAC */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_ONEBIT (9<<4) /* One Bit Audio */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_DOLBY_DIG_PLUS (10<<4) /* Dolby Digital + */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS_HD (11<<4) /* DTS-HD */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_MAT (12<<4) /* MAT (MLP) */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_DST (13<<4) /* DST */ +#define CEA861_AUDIO_INFOFRAME_DB1CT_WMA_PRO (14<<4) /* WMA Pro */ +#define CEA861_AUDIO_INFOFRAME_DB2SF (7<<2) /* mask - sample frequency */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM (0<<2) /* refer to stream */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_32000 (1<<2) /* 32kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_44100 (2<<2) /* 44.1kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_48000 (3<<2) /* 48kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_88200 (4<<2) /* 88.2kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_96000 (5<<2) /* 96kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_176400 (6<<2) /* 176.4kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SF_192000 (7<<2) /* 192kHz */ +#define CEA861_AUDIO_INFOFRAME_DB2SS (3<<0) /* mask - sample size */ +#define CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM (0<<0) /* refer to stream */ +#define CEA861_AUDIO_INFOFRAME_DB2SS_16BIT (1<<0) /* 16 bits */ +#define CEA861_AUDIO_INFOFRAME_DB2SS_20BIT (2<<0) /* 20 bits */ +#define CEA861_AUDIO_INFOFRAME_DB2SS_24BIT (3<<0) /* 24 bits */ +#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH (1<<7) /* mask - inhibit downmixing */ +#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PERMITTED (0<<7) /* stereo downmix permitted */ +#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED (1<<7) /* stereo downmis prohibited */ +#define CEA861_AUDIO_INFOFRAME_DB5_LSV (0xf<<3) /* mask - level-shift values */ + /***************************************************************************** * * * MIDI v1.0 interface * -- 1.7.5.4

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[alsa-devel] [PATCH] sound: fix core/vmaster.c kernel-doc warning
by Randy Dunlap 18 Apr '12

18 Apr '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix kernel-doc warning in sound/core/vmaster.c: Warning(sound/core/vmaster.c:429): No description found for parameter 'private_data' Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> --- sound/core/vmaster.c | 1 + 1 file changed, 1 insertion(+) --- lnx-34-rc3.orig/sound/core/vmaster.c +++ lnx-34-rc3/sound/core/vmaster.c @@ -419,6 +419,7 @@ EXPORT_SYMBOL(snd_ctl_make_virtual_maste * snd_ctl_add_vmaster_hook - Add a hook to a vmaster control * @kcontrol: vmaster kctl element * @hook: the hook function + * @private_data: the private_data pointer to be saved * * Adds the given hook to the vmaster control element so that it's called * at each time when the value is changed.

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Re: [alsa-devel] Thinkpad hda regression in 3.3
by Takashi Iwai 18 Apr '12

18 Apr '12

At Tue, 17 Apr 2012 20:23:36 -0400, Josh Boyer wrote: > > On Fri, Apr 13, 2012 at 02:08:14PM -0400, Josh Boyer wrote: > > On Fri, Apr 13, 2012 at 06:07:47PM +0200, Takashi Iwai wrote: > > > At Fri, 13 Apr 2012 15:07:07 +0200, > > > Takashi Iwai wrote: > > > (snip) > > > > If the above seem work (again, make sure that you adjust the mixer > > > > mutes and levels), I'll make kernel patches to add these fixups > > > > statically in the driver so that it works without extra patching. > > > > > > Looking through alsa-info.sh outputs from various ThinkPad models I've > > > gathered, it seems that we can use the same pin setups for all. > > > > We had a user try the alsa-tp-dock patch on an X201 and it didn't seem > > to work for them. See bug > > https://bugzilla.redhat.com/show_bug.cgi?id=806217 > > > > Perhaps the [codec] section isn't set properly for their configuration? > > The alsa-info for their machine is in comment #9 of the bug. > > > > > Below is a test patch for 3.3 kernel. (Note that it'd give a merge > > > conflict for 3.4 kernel.) > > > > OK, I'll do a scratch build with this and get people to test. > > I built your patch in a scratch kernel and have at least one user with a > t510 laptop report success. Christoph reported the mixers showing up, > but it wasn't clear to me if his sound actually worked on the x201. > I've CC'd both users (Kevin t510, Christoph x201 in a dock). Good. Should I wait for a copule of days for the test results before merging the patch, or go for now? thanks, Takashi

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[alsa-devel] [PATCH] ASoC: wm8994: Keep AIF3 tristated when not in use
by Mark Brown 17 Apr '12

17 Apr '12

Since AIF3 shares clock signals with other audio interfaces in order to ensure it doesn't drive undesirable clocks we need to tristate it. Rather than forcing the machine driver to do so have the driver do this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 36c338f..35c8a2a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1735,6 +1735,7 @@ SND_SOC_DAPM_MUX("AIF3ADC Mux", SND_SOC_NOPM, 0, 0, &wm8994_aif3adc_mux), }; static const struct snd_soc_dapm_widget wm8958_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF3", WM8994_POWER_MANAGEMENT_6, 5, 1, NULL, 0), SND_SOC_DAPM_MUX("Mono PCM Out Mux", SND_SOC_NOPM, 0, 0, &mono_pcm_out_mux), SND_SOC_DAPM_MUX("AIF2DACL Mux", SND_SOC_NOPM, 0, 0, &aif2dacl_src_mux), SND_SOC_DAPM_MUX("AIF2DACR Mux", SND_SOC_NOPM, 0, 0, &aif2dacr_src_mux), @@ -1975,6 +1976,9 @@ static const struct snd_soc_dapm_route wm8958_intercon[] = { { "AIF2DACR Mux", "AIF2", "AIF2DAC Mux" }, { "AIF2DACR Mux", "AIF3", "AIF3DACDAT" }, + { "AIF3DACDAT", NULL, "AIF3" }, + { "AIF3ADCDAT", NULL, "AIF3" }, + { "Mono PCM Out Mux", "AIF2ADCL", "AIF2ADCL" }, { "Mono PCM Out Mux", "AIF2ADCR", "AIF2ADCR" }, @@ -2845,10 +2849,6 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) reg = WM8994_AIF2_MASTER_SLAVE; mask = WM8994_AIF2_TRI; break; - case 3: - reg = WM8994_POWER_MANAGEMENT_6; - mask = WM8994_AIF3_TRI; - break; default: return -EINVAL; } @@ -2901,7 +2901,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = { static const struct snd_soc_dai_ops wm8994_aif3_dai_ops = { .hw_params = wm8994_aif3_hw_params, - .set_tristate = wm8994_set_tristate, }; static struct snd_soc_dai_driver wm8994_dai[] = { -- 1.7.10

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[alsa-devel] [PATCH v2] ASoC: da7210: Minor update for PLL and SRM
by Ashish Chavan 17 Apr '12

17 Apr '12

This patch converts multiple if conditions in to single if with "&&"s. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: Aligned '(' s for the branches --- sound/soc/codecs/da7210.c | 22 +++++++++------------- 1 files changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index cf4063b..d5fdf17 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -992,19 +992,15 @@ static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Search pll div array for correct divisors */ for (cnt = 0; cnt < ARRAY_SIZE(da7210_pll_div); cnt++) { - /* check fref */ - if (fref == da7210_pll_div[cnt].fref) { - /* check mode */ - if (da7210->master == da7210_pll_div[cnt].mode) { - /* check fout */ - if (fout == da7210_pll_div[cnt].fout) { - /* all match, pick up divisors */ - pll_div1 = da7210_pll_div[cnt].div1; - pll_div2 = da7210_pll_div[cnt].div2; - pll_div3 = da7210_pll_div[cnt].div3; - break; - } - } + /* check fref, mode and fout */ + if ((fref == da7210_pll_div[cnt].fref) && + (da7210->master == da7210_pll_div[cnt].mode) && + (fout == da7210_pll_div[cnt].fout)) { + /* all match, pick up divisors */ + pll_div1 = da7210_pll_div[cnt].div1; + pll_div2 = da7210_pll_div[cnt].div2; + pll_div3 = da7210_pll_div[cnt].div3; + break; } } if (cnt >= ARRAY_SIZE(da7210_pll_div)) -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: core: Fix card RTD count for deferred probe.
by Liam Girdwood 17 Apr '12

17 Apr '12

Currently we increment the number of RTD's per card during the DAI link bind. This can cause an incorrect RTD count when we cannot find a component and defer the probe (and hence perform the DAI link bind for the card again). Fix the count so that it is cleared before every card registration and bind attempt. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5d308e9..20e55531 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3155,6 +3155,7 @@ int snd_soc_register_card(struct snd_soc_card *card) GFP_KERNEL); if (card->rtd == NULL) return -ENOMEM; + card->num_rtd = 0; card->rtd_aux = &card->rtd[card->num_links]; for (i = 0; i < card->num_links; i++) -- 1.7.5.4

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[alsa-devel] [PATCH] [Trivial] Documentation: Add newline at end-of-file to files lacking one
by Jesper Juhl 17 Apr '12

17 Apr '12

This patch simply adds a newline character at end-of-file to those files in Documentation/ that currently lack one. This is done for a few different reasons: A) It's rather annoying when you do "cat some_file.txt" that your prompt/cursor ends up at the end of the last line of output rather than on a new line. B) Some tools that process files line-by-line may get confused by the lack of a newline on the last line. C) The "\ No newline at end of file" line in diffs annoys me for some reason. So, let's just add the missing newline once and for all. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- Documentation/ABI/stable/vdso | 2 +- Documentation/ABI/testing/sysfs-block-zram | 2 +- Documentation/ABI/testing/sysfs-bus-usb | 2 +- Documentation/ABI/testing/sysfs-bus-usb-devices-usbsevseg | 2 +- Documentation/ABI/testing/sysfs-class-backlight-driver-adp8870 | 2 +- Documentation/arm/Samsung-S3C24XX/H1940.txt | 2 +- Documentation/arm/Samsung-S3C24XX/SMDK2440.txt | 2 +- Documentation/dvb/opera-firmware.txt | 2 +- Documentation/sound/alsa/hdspm.txt | 2 +- Documentation/video4linux/cpia2_overview.txt | 2 +- Documentation/video4linux/stv680.txt | 2 +- 11 files changed, 11 insertions(+), 11 deletions(-) diff --git a/Documentation/ABI/stable/vdso b/Documentation/ABI/stable/vdso index 8a1cbb5..7cdfc28 100644 --- a/Documentation/ABI/stable/vdso +++ b/Documentation/ABI/stable/vdso @@ -24,4 +24,4 @@ though. (As of this writing, this ABI documentation as been confirmed for x86_64. The maintainers of the other vDSO-using architectures should confirm - that it is correct for their architecture.) \ No newline at end of file + that it is correct for their architecture.) diff --git a/Documentation/ABI/testing/sysfs-block-zram b/Documentation/ABI/testing/sysfs-block-zram index c8b3b48..ec93fe3 100644 --- a/Documentation/ABI/testing/sysfs-block-zram +++ b/Documentation/ABI/testing/sysfs-block-zram @@ -96,4 +96,4 @@ Description: overhead, allocated for this disk. So, allocator space efficiency can be calculated using compr_data_size and this statistic. - Unit: bytes \ No newline at end of file + Unit: bytes diff --git a/Documentation/ABI/testing/sysfs-bus-usb b/Documentation/ABI/testing/sysfs-bus-usb index 7c22a53..b521754 100644 --- a/Documentation/ABI/testing/sysfs-bus-usb +++ b/Documentation/ABI/testing/sysfs-bus-usb @@ -192,4 +192,4 @@ Description: combination of hub decriptor bits and platform-specific data such as ACPI. This file will read either "removable" or "fixed" if the information is available, and "unknown" - otherwise. \ No newline at end of file + otherwise. diff --git a/Documentation/ABI/testing/sysfs-bus-usb-devices-usbsevseg b/Documentation/ABI/testing/sysfs-bus-usb-devices-usbsevseg index cb830df..70d00df 100644 --- a/Documentation/ABI/testing/sysfs-bus-usb-devices-usbsevseg +++ b/Documentation/ABI/testing/sysfs-bus-usb-devices-usbsevseg @@ -40,4 +40,4 @@ Description: Controls the decimal places on the device. the value of 10 ** n. Assume this field has the value k and has 1 or more decimal places set, to set the mth place (where m is not already set), - change this fields value to k + 10 ** m. \ No newline at end of file + change this fields value to k + 10 ** m. diff --git a/Documentation/ABI/testing/sysfs-class-backlight-driver-adp8870 b/Documentation/ABI/testing/sysfs-class-backlight-driver-adp8870 index 4a9c545..33e6488 100644 --- a/Documentation/ABI/testing/sysfs-class-backlight-driver-adp8870 +++ b/Documentation/ABI/testing/sysfs-class-backlight-driver-adp8870 @@ -53,4 +53,4 @@ Description: Documentation/ABI/stable/sysfs-class-backlight. It can be enabled by writing the value stored in /sys/class/backlight/<backlight>/max_brightness to - /sys/class/backlight/<backlight>/brightness. \ No newline at end of file + /sys/class/backlight/<backlight>/brightness. diff --git a/Documentation/arm/Samsung-S3C24XX/H1940.txt b/Documentation/arm/Samsung-S3C24XX/H1940.txt index f4a7b22..b738859 100644 --- a/Documentation/arm/Samsung-S3C24XX/H1940.txt +++ b/Documentation/arm/Samsung-S3C24XX/H1940.txt @@ -37,4 +37,4 @@ Maintainers Thanks to the many others who have also provided support. -(c) 2005 Ben Dooks \ No newline at end of file +(c) 2005 Ben Dooks diff --git a/Documentation/arm/Samsung-S3C24XX/SMDK2440.txt b/Documentation/arm/Samsung-S3C24XX/SMDK2440.txt index 32e1eae..429390b 100644 --- a/Documentation/arm/Samsung-S3C24XX/SMDK2440.txt +++ b/Documentation/arm/Samsung-S3C24XX/SMDK2440.txt @@ -53,4 +53,4 @@ Maintainers and to Simtec Electronics for allowing me time to work on this. -(c) 2004 Ben Dooks \ No newline at end of file +(c) 2004 Ben Dooks diff --git a/Documentation/dvb/opera-firmware.txt b/Documentation/dvb/opera-firmware.txt index 93e784c..ac5d86f 100644 --- a/Documentation/dvb/opera-firmware.txt +++ b/Documentation/dvb/opera-firmware.txt @@ -24,4 +24,4 @@ After that the driver can load the firmware in kernel config and have hotplug running). -Marco Gittler <g.marco(a)freenet.de> \ No newline at end of file +Marco Gittler <g.marco(a)freenet.de> diff --git a/Documentation/sound/alsa/hdspm.txt b/Documentation/sound/alsa/hdspm.txt index 7a67ff7..7ba3194 100644 --- a/Documentation/sound/alsa/hdspm.txt +++ b/Documentation/sound/alsa/hdspm.txt @@ -359,4 +359,4 @@ Calling Parameter: enable_monitor int array (min = 1, max = 8), "Enable Analog Out on Channel 63/64 by default." - note: here the analog output is enabled (but not routed). \ No newline at end of file + note: here the analog output is enabled (but not routed). diff --git a/Documentation/video4linux/cpia2_overview.txt b/Documentation/video4linux/cpia2_overview.txt index a6e5366..ad6adbe 100644 --- a/Documentation/video4linux/cpia2_overview.txt +++ b/Documentation/video4linux/cpia2_overview.txt @@ -35,4 +35,4 @@ the camera. There are three modes for this. Block mode requests a number of contiguous registers. Random mode reads or writes random registers with a tuple structure containing address/value pairs. The repeat mode is only used by VP4 to load a firmware patch. It contains a starting address and -a sequence of bytes to be written into a gpio port. \ No newline at end of file +a sequence of bytes to be written into a gpio port. diff --git a/Documentation/video4linux/stv680.txt b/Documentation/video4linux/stv680.txt index 4f8946f..e3de336 100644 --- a/Documentation/video4linux/stv680.txt +++ b/Documentation/video4linux/stv680.txt @@ -50,4 +50,4 @@ The latest info on this driver can be found at: http://personal.clt.bellsouth.net/~kjsisson or at http://stv0680-usb.sourceforge.net -Any questions to me can be send to: kjsisson(a)bellsouth.net \ No newline at end of file +Any questions to me can be send to: kjsisson(a)bellsouth.net -- 1.7.10 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] [PATCH] ASoC: da7210: Minor update for PLL and SRM
by Ashish Chavan 17 Apr '12

17 Apr '12

This patch converts multiple if conditions in to single if with "&&"s. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 22 +++++++++------------- 1 files changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index cf4063b..d3ca28b 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -992,19 +992,15 @@ static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Search pll div array for correct divisors */ for (cnt = 0; cnt < ARRAY_SIZE(da7210_pll_div); cnt++) { - /* check fref */ - if (fref == da7210_pll_div[cnt].fref) { - /* check mode */ - if (da7210->master == da7210_pll_div[cnt].mode) { - /* check fout */ - if (fout == da7210_pll_div[cnt].fout) { - /* all match, pick up divisors */ - pll_div1 = da7210_pll_div[cnt].div1; - pll_div2 = da7210_pll_div[cnt].div2; - pll_div3 = da7210_pll_div[cnt].div3; - break; - } - } + /* check fref, mode and fout */ + if ((fref == da7210_pll_div[cnt].fref) && + (da7210->master == da7210_pll_div[cnt].mode) && + (fout == da7210_pll_div[cnt].fout)) { + /* all match, pick up divisors */ + pll_div1 = da7210_pll_div[cnt].div1; + pll_div2 = da7210_pll_div[cnt].div2; + pll_div3 = da7210_pll_div[cnt].div3; + break; } } if (cnt >= ARRAY_SIZE(da7210_pll_div)) -- 1.7.1

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