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

[alsa-devel] [PATCH] alsa: hide HDMI/ELD printks unless in debug kernels
by Fengguang Wu 09 Apr '12

09 Apr '12

These trivial messages will show up repeatedly on hot plug as well as video mode changes, which could be annoying. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_eld.c | 6 +++--- sound/pci/hda/patch_hdmi.c | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) --- linux.orig/sound/pci/hda/hda_eld.c 2011-11-11 19:08:21.000000000 +0800 +++ linux/sound/pci/hda/hda_eld.c 2011-11-11 19:09:35.000000000 +0800 @@ -410,7 +410,7 @@ static void hdmi_show_short_audio_desc(s else buf2[0] = '\0'; - printk(KERN_INFO "HDMI: supports coding type %s:" + snd_printdd("HDMI: supports coding type %s:" " channels = %d, rates =%s%s\n", cea_audio_coding_type_names[a->format], a->channels, @@ -434,14 +434,14 @@ void snd_hdmi_show_eld(struct hdmi_eld * { int i; - printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n", + snd_printdd("HDMI: detected monitor %s at connection type %s\n", e->monitor_name, eld_connection_type_names[e->conn_type]); if (e->spk_alloc) { char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf)); - printk(KERN_INFO "HDMI: available speakers:%s\n", buf); + snd_printdd("HDMI: available speakers:%s\n", buf); } for (i = 0; i < e->sad_count; i++) --- linux.orig/sound/pci/hda/patch_hdmi.c 2011-11-11 19:07:15.000000000 +0800 +++ linux/sound/pci/hda/patch_hdmi.c 2011-11-11 19:08:16.000000000 +0800 @@ -757,7 +757,7 @@ static void hdmi_intrinsic_event(struct int pin_idx; struct hdmi_eld *eld; - printk(KERN_INFO + snd_printdd( "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, pd, eldv); @@ -989,7 +989,7 @@ static void hdmi_present_sense(struct hd else eld->eld_valid = 0; - printk(KERN_INFO + snd_printdd( "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, eld->monitor_present, eld->eld_valid);

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[alsa-devel] [PATCH][Trivial] sound/pci/riptide/riptide.c: remove trailing whitespace
by Jesper Juhl 09 Apr '12

09 Apr '12

There's a bunch of trailing whitespace at the top of the sound/pci/riptide/riptide.c file - this patch gets rid of it. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/pci/riptide/riptide.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 0481d94..a8b07be 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -21,29 +21,29 @@ /* History: - 02/15/2004 first release - + This Driver is based on the OSS Driver version from Linuxant (riptide-0.6lnxtbeta03111100) credits from the original files: - - MODULE NAME: cnxt_rt.h + + MODULE NAME: cnxt_rt.h AUTHOR: K. Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 02/1/2000 KNL - MODULE NAME: int_mdl.c + MODULE NAME: int_mdl.c AUTHOR: Konstantin Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 10/01/99 KNL - - MODULE NAME: riptide.h + + MODULE NAME: riptide.h AUTHOR: O. Druzhinin (Transcribed by OLD) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 10/16/97 OLD - MODULE NAME: Rp_Cmdif.cpp + MODULE NAME: Rp_Cmdif.cpp AUTHOR: O. Druzhinin (Transcribed by OLD) K. Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By @@ -51,7 +51,7 @@ Adopted from NT4 driver 6/22/99 OLD Ported to Linux 9/01/99 KNL - MODULE NAME: rt_hw.c + MODULE NAME: rt_hw.c AUTHOR: O. Druzhinin (Transcribed by OLD) C. Lazarev (Transcribed by CNL) HISTORY: Major Revision Date By @@ -73,13 +73,13 @@ ----------------------------- -------- ----- Created 11/18/97 CNL - MODULE NAME: rt_mdl.c + MODULE NAME: rt_mdl.c AUTHOR: Konstantin Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 10/01/99 KNL - MODULE NAME: mixer.h + MODULE NAME: mixer.h AUTHOR: K. Kenney HISTORY: Major Revision Date By ----------------------------- -------- ----- -- 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] [RFC] Fix internal mic for Lenovo Ideapad U300s
by David Henningsson 07 Apr '12

07 Apr '12

The internal mic input is phase inverted on one channel. To avoid people in userspace summing the channels together and get zero result, use a separate mixer control for the inverted channel. BugLink: https://bugs.launchpad.net/bugs/903853 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Inverted Internal Mic, take 2. I refined my idea a little. By calling the left channel "Internal Mic" and the right channel "Inverted Internal Mic", we're giving the user a choice whether to activate it or keeping it muted. For PulseAudio, the quick fix is just to make sure the "Inverted Internal Mic" control is muted. If we in the long run want to actually handle this and swap the phase around, we now also have a reasonable indicator if we should do this, by checking if the "Inverted Internal Mic" control exists. For the ALSA init DB (i e, should the "Inverted Internal Mic" be initially muted?) that is no big deal for me, since this will be overwritten by PulseAudio later anyway. I'm also open for better names than "Inverted Internal Mic" if you have any. As with the previous version, this is a draft version of the patch, and could be done slightly more elegant I guess. What do you say? sound/pci/hda/patch_conexant.c | 45 +++++++++++++++++++++++++++++---------- 1 files changed, 33 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e6eafb1..cd3859b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int single_adc_amp:1; + unsigned int fixup_stereo_dmic:1; unsigned int adc_switching:1; @@ -4107,9 +4108,9 @@ static int cx_auto_init(struct hda_codec *codec) static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir, int amp_idx) + hda_nid_t nid, int hda_dir, int amp_idx, int chs) { - static char name[32]; + static char name[44]; static struct snd_kcontrol_new knew[] = { HDA_CODEC_VOLUME(name, 0, 0, 0), HDA_CODEC_MUTE(name, 0, 0, 0), @@ -4119,7 +4120,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx, hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; @@ -4138,7 +4139,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, } #define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ - cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3) #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -4216,14 +4217,24 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int i; for (i = 0; i < spec->num_adc_nids; i++) { + int chs = 3; hda_nid_t adc_nid = spec->adc_nids[i]; int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; if (spec->single_adc_amp) idx = 0; + + if (spec->fixup_stereo_dmic && (strcmp("Internal Mic", label) == 0)) { + /* Make two independent kcontrols */ + int err = cx_auto_add_volume_idx(codec, "Inverted Internal Mic", pfx, + cidx, adc_nid, HDA_INPUT, idx, 2); + if (err < 0) + return err; + chs = 1; + } return cx_auto_add_volume_idx(codec, label, pfx, - cidx, adc_nid, HDA_INPUT, idx); + cidx, adc_nid, HDA_INPUT, idx, chs); } return 0; } @@ -4405,22 +4416,30 @@ static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg) } -static void apply_pin_fixup(struct hda_codec *codec, +enum { + CXT_PINCFG_LENOVO_X200, + CXT_FIXUP_STEREO_DMIC, +}; + +static void apply_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, const struct cxt_pincfg **table) { + struct conexant_spec *spec = codec->spec; + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { + if (quirk && table[quirk->value]) { snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", quirk->name); apply_pincfg(codec, table[quirk->value]); } + if (quirk->value == CXT_FIXUP_STEREO_DMIC) { + snd_printdd(KERN_INFO "hda_codec: applying internal mic workaround for %s\n", + quirk->name); + spec->fixup_stereo_dmic = 1; + } } -enum { - CXT_PINCFG_LENOVO_X200, -}; - static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4431,10 +4450,12 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { static const struct cxt_pincfg *cxt_pincfg_tbl[] = { [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_FIXUP_STEREO_DMIC] = NULL, }; static const struct snd_pci_quirk cxt_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), + SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), {} }; @@ -4477,7 +4498,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); + apply_fixup(codec, cxt_fixups, cxt_pincfg_tbl); /* Show mute-led control only on HP laptops * This is a sort of white-list: on HP laptops, EAPD corresponds -- 1.7.9.1

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Re: [alsa-devel] [PATCH 0/6] ALSA: hda - CX20549 cleanup/fixes
by Takashi Iwai 07 Apr '12

07 Apr '12

At Fri, 6 Apr 2012 15:34:14 +0200, Michael Karcher wrote: > > This is the revised patch series for the CX20549 (and related stuff) > > Changes: > Patch 1/6: > - spec->single_adc_amp has been moved to codec->single_adc_amp > to fix printing in /proc. > Patch 2/6: > - "Line In" has been changed to "Line". > - This change has also been performed on the Mixer Labels. > Patch 4/6: > - .num_items has been adjusted, too. > - A comment about physical testing has been added to the commit message. > Patch 5/6: > - pin_amp_workaround is not set for CX20549 in auto-setup, too Thanks, applied all patches now. (BTW, the first patch caused a few obvious warnings from checkpatch.pl, so I fixed manually. At the next time, please run checkpatch.pl once before submission.) Takashi

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Re: [alsa-devel] [PATCH 0/6] sound/pci/hda: CX20549 cleanup/fixes
by Takashi Iwai 07 Apr '12

07 Apr '12

At Fri, 06 Apr 2012 15:01:47 +0200, Michael Karcher wrote: > > On Thu, 2012-04-05 at 16:28 +0200, Takashi Iwai wrote: > > BTW, did you try model=auto option to enable the auto-parser? > > If it doesn't work well for you, let me know. > > I tried the model=auto option, too. It works mostly, but the mixer node > seems to not be utilized correctly/at all. The capture source list does > not contain "mixer" anymore, and there are no playback controls for > Internal Mic/Mic/Line. Right, the analog loopback is dropped in the auto-parser for simplicity. > The Capture setup seems to be built correctly > (except for the missing mixer). The auto setup adds two extra input > devices for jack sense. > > > In anyway, give alsa-info.sh output (run with --no-upload option) for > > better checks. > > alsa-info for the revised patch-set (mailed soon) with and without > model=auto is attached. Information I consider irrelevant (non-sound > modules and dmesg context not relating to sound) has been edited out. Thanks. > BTW: I seem to have encountered an ALSA user-space bug: The codec > provides a "master capture switch", i.e. the "mute" of the adc in amp. > It also provides a "capture source selection enum", i.e. the "Connection > Select" of the adc widget. The enum provides mutually exclusive capture > sources that have no volume control, whilst the master capture switch is > coupled with the capture volume control, and should not be related to > the source enumeration. But if I play around with capture sources in > alsamixer, the "Capture Switch" is turned off as soon as I switch the > source from the first item in the Mux list to something else. This sounds like a bug in alsa-lib mixer abstraction. There is a known shortcomings in the "capture source" enum, e.g. it can't take more than one ADCs. > This happens with both the patched hard-coded "BenQ" configuration and > the auto configuration. OK, I'll check it in the next week. thanks, Takashi

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[alsa-devel] ASoC updates for 3.4
by Mark Brown 07 Apr '12

07 Apr '12

The following changes since commit dd775ae2549217d3ae09363e3edb305d0fa19928: Linux 3.4-rc1 (2012-03-31 16:24:09 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.4 for you to fetch changes up to 8abe05c6eb358967f16bce8a02c88d57c82cfbd6: ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS (2012-04-06 12:28:49 +0100) ---------------------------------------------------------------- ASoC: fixes for 3.4 A bunch of driver-specific fixes and one generic fix for the new support for platform DAPM contexts - we were picking the wrong default for the idle_bias_off setting which was meaning we weren't actually achieving any useful runtime PM on platform devices. ---------------------------------------------------------------- Fabio Estevam (2): ASoC: imx-audmux: Fix ssi port numbers in sysfs ASoC: imx-audmux: Check for NULL pointer Jarkko Nikula (1): MAINTAINERS: Add missing ASoC OMAP co-maintainer Kuninori Morimoto (1): ASoC: ak4642: fixup: mute needs +1 step Mark Brown (1): MAINTAINERS: Don't list everyone working on Wolfson drivers Martin Jansa (1): ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macro Stephen Warren (3): ASoC: tegra: ensure clocks are enabled when touching registers ASoC: set idle_bias_off=1 for all platform DAPM contexts ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS Zeng Zhaoming (1): ASoC: sgtl5000: Enable VAG when DAC/ADC up MAINTAINERS | 4 ++-- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/sgtl5000.c | 25 +++++++++++++------------ sound/soc/imx/imx-audmux.c | 5 ++++- sound/soc/pxa/pxa2xx-i2s.c | 1 + sound/soc/soc-core.c | 2 ++ sound/soc/tegra/tegra_i2s.c | 6 +++++- sound/soc/tegra/tegra_spdif.c | 4 ++++ 8 files changed, 32 insertions(+), 17 deletions(-)

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[alsa-devel] sgtl5000 module install fails during instantiation for i.MX53 QSB
by madhukesh s 07 Apr '12

07 Apr '12

Hi, On insmod'ing alsa modules for i.MX53 QSB, I get the following error: root@freescale-qsb:/lib/modules/2.6.38.8+# modprobe snd-soc-sgtl5000 snd_soc_instantiate_card:Entry snd_soc_instantiate_card:Entry soc_probe_dai_link:Entry soc_probe_codec: Probing for card sgtl5000-audio sgtl5000_enable_regulators:Entry _regulator_get: devname = 1-000a sgtl5000 1-000a: regulator_bulk_get [for (null)]: Failed to get supply 'VDDA': -19 sgtl5000_enable_regulators: regulator_bulk_get err!!! print_constraints: 1-000a: 1200 mV normal _regulator_get: devname = 1-000a sgtl5000 1-000a: regulator_bulk_get [for (null)]: Failed to get supply 'VDDA': -19 sgtl5000 1-000a: Failed to request supplies:1 -19 sgtl5000 1-000a: asoc: failed to probe CODEC sgtl5000.1-000a: -19 asoc: failed to probe codec sgtl5000.1-000a asoc: failed to instantiate card sgtl5000-audio: -19 On adding more printks, I realize that "_regulator_get" function is failing(not able to find supply VDDA). The statically built module gives no such errors. Can someone give some info on this? Regards, maas

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[alsa-devel] Alsa support for conexant
by Sql Sql 06 Apr '12

06 Apr '12

I have a Lenovo Thinkpad T400 with a mini dock which has SPDIF output. It is running Debian Squeeze. When I run alsamixer, I can see the soundcard detected is: Conexant CX20561 When I run aplay -l: **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: Conexant Digital [Conexant Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 However, I can't get it to output through SPDIF. I tried: aplay -Dplug:iec958 /usr/share/sounds/alsa/Front_Center.wav = no sound I tried: pcm.!default { type plug slave { pcm "front:CARD=Intel" } } (restarted alsautils), it still doesn't work. I tried adding: options snd-hda-intel model=lenovo-x200 options snd-hda-intel model=lenovo options snd-hda-intel model=laptop options snd-hda-intel model=thinkpad and then alsa force reload None of these work. Is there anyway I can get it to output through SPDIF? My other computer that has SPDIF outputs fine using the custom asound.conf so why isn't it working for the lenovo? BTW, I confirmed the SPDIF is not defective by booting into Windows and testing it.

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[alsa-devel] [PATCH V2] ASoC: tegra: utils: add support for Tegra30 devices
by Stephen Warren 06 Apr '12

06 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> Tegra30 has some additional clocks that need to be manipulated, names some clocks differently, runs PLLs at different base rates, etc. The utility code needs to handle this. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Determine Tegra version internally to the utility code, rather than requiring each ASoC machine driver be updated to pass in this information. sound/soc/tegra/tegra_asoc_utils.c | 29 ++++++++++++++++++++++++----- sound/soc/tegra/tegra_asoc_utils.h | 9 +++++++-- 2 files changed, 31 insertions(+), 7 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index f8428e4..f59e8b5 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -2,7 +2,7 @@ * tegra_asoc_utils.c - Harmony machine ASoC driver * * Author: Stephen Warren <swarren(a)nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -25,6 +25,7 @@ #include <linux/err.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/of.h> #include "tegra_asoc_utils.h" @@ -40,7 +41,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 22050: case 44100: case 88200: - new_baseclock = 56448000; + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + new_baseclock = 56448000; + else + new_baseclock = 564480000; break; case 8000: case 16000: @@ -48,7 +52,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 48000: case 64000: case 96000: - new_baseclock = 73728000; + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + new_baseclock = 73728000; + else + new_baseclock = 552960000; break; default: return -EINVAL; @@ -78,7 +85,7 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, return err; } - /* Don't set cdev1 rate; its locked to pll_a_out0 */ + /* Don't set cdev1/extern1 rate; it's locked to pll_a_out0 */ err = clk_enable(data->clk_pll_a); if (err) { @@ -112,6 +119,15 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->dev = dev; + if (!of_have_populated_dt()) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; + else if (of_machine_is_compatible("nvidia,tegra20")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; + else if (of_machine_is_compatible("nvidia,tegra30")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; + else + return -EINVAL; + data->clk_pll_a = clk_get_sys(NULL, "pll_a"); if (IS_ERR(data->clk_pll_a)) { dev_err(data->dev, "Can't retrieve clk pll_a\n"); @@ -126,7 +142,10 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, goto err_put_pll_a; } - data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); + 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); if (IS_ERR(data->clk_cdev1)) { dev_err(data->dev, "Can't retrieve clk cdev1\n"); ret = PTR_ERR(data->clk_cdev1); diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 4818195..44db1db 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -2,7 +2,7 @@ * tegra_asoc_utils.h - Definitions for Tegra DAS driver * * Author: Stephen Warren <swarren(a)nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -26,8 +26,14 @@ struct clk; struct device; +enum tegra_asoc_utils_soc { + TEGRA_ASOC_UTILS_SOC_TEGRA20, + TEGRA_ASOC_UTILS_SOC_TEGRA30, +}; + struct tegra_asoc_utils_data { struct device *dev; + enum tegra_asoc_utils_soc soc; struct clk *clk_pll_a; struct clk *clk_pll_a_out0; struct clk *clk_cdev1; @@ -42,4 +48,3 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data); #endif - -- 1.7.0.4

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[alsa-devel] [PATCH V2] ASoC: tegra: set a sensible initial clock rate
by Stephen Warren 06 Apr '12

06 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> Initialize the audio clock tree appropriately for some reasonable rate. This makes sure the PLLs etc. are actually programmed to something reasonable when the audio driver is loaded. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Use 44100 as the default rate, not 48000, since it's more common, and consistent with the clock tables in board-dt-tegra*.c. This patch is ordered after the patch I just sent to add Tegra30 support to the utils file. However, the dependency is just for context; this should be independent from that patch. sound/soc/tegra/tegra_asoc_utils.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index f59e8b5..266189d 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -152,8 +152,14 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, goto err_put_pll_a_out0; } + ret = tegra_asoc_utils_set_rate(data, 44100, 256 * 44100); + if (ret) + goto err_put_cdev1; + return 0; +err_put_cdev1: + clk_put(data->clk_cdev1); err_put_pll_a_out0: clk_put(data->clk_pll_a_out0); err_put_pll_a: -- 1.7.0.4

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