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Re: [alsa-devel] Fast Track Ultra latency and new streaming logic.
by Takashi Iwai 09 May '12

09 May '12

At Wed, 9 May 2012 15:05:47 +0200, Felix Homann wrote: > > Am 03.05.2012 16:03 schrieb "Felix Homann" <linuxaudio(a)showlabor.de>: > > > > Is this a > > flaw in Alsa, Jack or the USB stack? > > I'm starting to think that this is a Jack issue. At least I can set rather > small period and buffer sizes in aplay without any xruns. Or a full-duplex issue? You can test the latency via alsa-lib/test/latency program. (Yes, the usage is quite undocumented. Anyone contributing the howto would be great...) Takashi

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[alsa-devel] [PATCH] ASoC: wm8994: Fix sparse warning due to use of 0 as NULL
by Mark Brown 09 May '12

09 May '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b3b2d5f..dfc4e90 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -792,7 +792,7 @@ static void vmid_reference(struct snd_soc_codec *codec) switch (wm8994->vmid_mode) { default: - WARN_ON(0 == "Invalid VMID mode"); + WARN_ON(NULL == "Invalid VMID mode"); case WM8994_VMID_NORMAL: /* Startup bias, VMID ramp & buffer */ snd_soc_update_bits(codec, WM8994_ANTIPOP_2, -- 1.7.10

1 0

[alsa-devel] [PATCH RESEND 3/3] ASoC: cs42l73: Use DAPM routes to hook AIF widgets to DAI's
by Brian Austin 09 May '12

09 May '12

Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 41 ++++++++++++++++++++++++++++------------- 1 file changed, 28 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index a94e1e9..0aa56aa 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -599,17 +599,17 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), - SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("XSPOUTL", NULL, 0, CS42L73_PWRCTL2, 1, 1), - SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("XSPOUTR", NULL, 0, CS42L73_PWRCTL2, 1, 1), - SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0, + SND_SOC_DAPM_AIF_OUT("ASPOUTL", NULL, 0, CS42L73_PWRCTL2, 3, 1), - SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0, + SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL, 0, CS42L73_PWRCTL2, 3, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL, 0, CS42L73_PWRCTL2, 4, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -638,21 +638,21 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINR", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINM", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINL", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINR", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -776,6 +776,14 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"HL Left Mixer", NULL, "VSPIN"}, {"HL Right Mixer", NULL, "VSPIN"}, + {"ASPINL", NULL, "ASP Playback"}, + {"ASPINM", NULL, "ASP Playback"}, + {"ASPINR", NULL, "ASP Playback"}, + {"XSPINL", NULL, "XSP Playback"}, + {"XSPINM", NULL, "XSP Playback"}, + {"XSPINR", NULL, "XSP Playback"}, + {"VSPIN", NULL, "VSP Playback"}, + /* Capture Paths */ {"MIC1", NULL, "MIC1 Bias"}, {"PGA Left Mux", "Mic 1", "MIC1"}, @@ -822,6 +830,13 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"VSPOUTL", NULL, "VSPL Output Mixer"}, {"VSPOUTR", NULL, "VSPR Output Mixer"}, + + {"ASP Capture", NULL, "ASPOUTL"}, + {"ASP Capture", NULL, "ASPOUTR"}, + {"XSP Capture", NULL, "XSPOUTL"}, + {"XSP Capture", NULL, "XSPOUTR"}, + {"VSP Capture", NULL, "VSPOUTL"}, + {"VSP Capture", NULL, "VSPOUTR"}, }; struct cs42l73_mclk_div { -- 1.7.10

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[alsa-devel] [PATCH 1/3] ASoC: cs42l73: Convert to module_i2c_driver()
by Brian Austin 09 May '12

09 May '12

Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 20 +------------------- 1 file changed, 1 insertion(+), 19 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index a94e1e9..60b7f53 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1428,25 +1428,7 @@ static struct i2c_driver cs42l73_i2c_driver = { }; -static int __init cs42l73_modinit(void) -{ - int ret; - ret = i2c_add_driver(&cs42l73_i2c_driver); - if (ret != 0) { - pr_err("Failed to register CS42L73 I2C driver: %d\n", ret); - return ret; - } - return 0; -} - -module_init(cs42l73_modinit); - -static void __exit cs42l73_exit(void) -{ - i2c_del_driver(&cs42l73_i2c_driver); -} - -module_exit(cs42l73_exit); +module_i2c_driver(cs42l73_i2c_driver); MODULE_DESCRIPTION("ASoC CS42L73 driver"); MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com>"); -- 1.7.10

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[alsa-devel] [PATCH 11/11] ASoC: Ux500: Add machine-driver
by Ola Lilja 09 May '12

09 May '12

Add machine-driver for ST-Ericsson U8500 platform, including support for the AB8500-codec. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/ux500/Kconfig | 11 ++ sound/soc/ux500/Makefile | 3 +- sound/soc/ux500/mop500.c | 117 +++++++++++++++++ sound/soc/ux500/mop500_ab8500.c | 270 +++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/mop500_ab8500.h | 21 +++ 5 files changed, 421 insertions(+), 1 deletions(-) create mode 100644 sound/soc/ux500/mop500.c create mode 100644 sound/soc/ux500/mop500_ab8500.c create mode 100644 sound/soc/ux500/mop500_ab8500.h diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 59da6ce..c158e39 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -21,3 +21,14 @@ config SND_SOC_UX500_PLAT_DMA help Say Y if you want to enable the Ux500 platform-driver. ++config SND_SOC_UX500_MACH_MOP500 ++ tristate "Machine - MOP500 (Ux500 + AB8500)" + depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500 + select SND_SOC_AB8500_CODEC + select SND_SOC_UX500_PLAT_MSP_I2S + select SND_SOC_UX500_PLAT_DMA + help + Select this to enable the MOP500 machine-driver. + This will enable platform-drivers for: Ux500 + This will enable codec-drivers for: AB8500 + diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 0c66a44..cce0c11 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -6,4 +6,5 @@ obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o snd-soc-ux500-plat-dma-objs := ux500_pcm.o obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o - +snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o +obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c new file mode 100644 index 0000000..5d632ff --- /dev/null +++ b/sound/soc/ux500/mop500.c @@ -0,0 +1,117 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja (ola.o.lilja(a)stericsson.com) + * for ST-Ericsson. + * + * License terms: + * + * 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 <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/io.h> +#include <linux/spi/spi.h> + +#include <mach/board-mop500-audio.h> + +#include <sound/soc.h> +#include <sound/initval.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" + +#include <mop500_ab8500.h> + +/* Define the whole MOP500 soundcard, linking platform to the codec-drivers */ +struct snd_soc_dai_link mop500_dai_links[] = { + { + .name = "ab8500_0", + .stream_name = "ab8500_0", + .cpu_dai_name = "ux500-msp-i2s.1", + .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = mop500_ab8500_machine_init, + .ops = mop500_ab8500_ops, + }, + { + .name = "ab8500_1", + .stream_name = "ab8500_1", + .cpu_dai_name = "ux500-msp-i2s.3", + .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = NULL, + .ops = mop500_ab8500_ops, + }, +}; + +static struct snd_soc_card mop500_card = { + .name = "MOP500-card", + .probe = NULL, + .dai_link = mop500_dai_links, + .num_links = ARRAY_SIZE(mop500_dai_links), +}; + +static int __devinit mop500_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("%s: Enter.\n", __func__); + + dev_dbg(&pdev->dev, "%s: Enter.\n", __func__); + + mop500_card.dev = &pdev->dev; + + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", + __func__, mop500_card.name); + platform_set_drvdata(pdev, &mop500_card); + + snd_soc_card_set_drvdata(&mop500_card, NULL); + + dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n", + __func__, mop500_card.name, mop500_card.num_links); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n", + __func__, mop500_card.name, mop500_card.dai_link[0].name); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n", + __func__, mop500_card.name, + mop500_card.dai_link[0].stream_name); + + ret = snd_soc_register_card(&mop500_card); + if (ret) + dev_err(&pdev->dev, + "Error: snd_soc_register_card failed (%d)!\n", + ret); + + return ret; +} + +static int __devexit mop500_remove(struct platform_device *pdev) +{ + struct snd_soc_card *mop500_card = platform_get_drvdata(pdev); + + pr_debug("%s: Enter.\n", __func__); + + snd_soc_unregister_card(mop500_card); + platform_set_drvdata(pdev, NULL); + + return 0; +} + +static struct platform_driver snd_soc_mop500_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "snd-soc-mop500", + }, + .probe = mop500_probe, + .remove = __devexit_p(mop500_remove), +}; + +module_platform_driver(snd_soc_mop500_driver); + + diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c new file mode 100644 index 0000000..5b8151d --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.c @@ -0,0 +1,270 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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/device.h> +#include <linux/io.h> + +#include <mach/hardware.h> +#include <mach/board-mop500-audio.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" +#include "../codecs/ab8500-codec.h" + +#define TX_SLOT_MONO 0x0008 +#define TX_SLOT_STEREO 0x000a +#define RX_SLOT_MONO 0x0001 +#define RX_SLOT_STEREO 0x0003 +#define TX_SLOT_8CH 0x00FF +#define RX_SLOT_8CH 0x00FF + +#define DEF_TX_SLOTS TX_SLOT_STEREO +#define DEF_RX_SLOTS RX_SLOT_MONO + +#define DRIVERMODE_NORMAL 0 +#define DRIVERMODE_CODEC_ONLY 1 + +/* Slot configuration */ +static unsigned int tx_slots = DEF_TX_SLOTS; +static unsigned int rx_slots = DEF_RX_SLOTS; + +static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { + SOC_DAPM_PIN_SWITCH("Headset Left"), + SOC_DAPM_PIN_SWITCH("Headset Right"), + SOC_DAPM_PIN_SWITCH("Earpiece"), + SOC_DAPM_PIN_SWITCH("Speaker Left"), + SOC_DAPM_PIN_SWITCH("Speaker Right"), + SOC_DAPM_PIN_SWITCH("LineOut Left"), + SOC_DAPM_PIN_SWITCH("LineOut Right"), + SOC_DAPM_PIN_SWITCH("Vibra 1"), + SOC_DAPM_PIN_SWITCH("Vibra 2"), + SOC_DAPM_PIN_SWITCH("Mic 1"), + SOC_DAPM_PIN_SWITCH("Mic 2"), + SOC_DAPM_PIN_SWITCH("LineIn Left"), + SOC_DAPM_PIN_SWITCH("LineIn Right"), + SOC_DAPM_PIN_SWITCH("DMic 1"), + SOC_DAPM_PIN_SWITCH("DMic 2"), + SOC_DAPM_PIN_SWITCH("DMic 3"), + SOC_DAPM_PIN_SWITCH("DMic 4"), + SOC_DAPM_PIN_SWITCH("DMic 5"), + SOC_DAPM_PIN_SWITCH("DMic 6"), +}; + +/* ASoC */ + +void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + + dev_dbg(dev, "%s: Enter\n", __func__); + + /* Reset slots configuration to default(s) */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tx_slots = DEF_TX_SLOTS; + else + rx_slots = DEF_RX_SLOTS; +} + +int mop500_ab8500_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; + struct device *dev = rtd->card->dev; + unsigned int fmt, fmt_if1; + int channels, ret = 0, driver_mode, slots; + unsigned int sw_codec, sw_cpu; + bool is_playback; + + dev_dbg(dev, "%s: Enter\n", __func__); + + dev_dbg(dev, "%s: substream->pcm->name = %s\n" + "substream->pcm->id = %s.\n" + "substream->name = %s.\n" + "substream->number = %d.\n", + __func__, + substream->pcm->name, + substream->pcm->id, + substream->name, + substream->number); + + channels = params_channels(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + sw_cpu = 32; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + sw_cpu = 16; + break; + + default: + return -EINVAL; + } + + /* Setup codec depending on driver-mode */ + driver_mode = (channels == 8) ? + DRIVERMODE_CODEC_ONLY : DRIVERMODE_NORMAL; + dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__, + (driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY"); + + /* Setup format */ + + if (driver_mode == DRIVERMODE_NORMAL) { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CONT; + } else { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_GATED; + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Setup TDM-slots */ + + is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + switch (channels) { + case 1: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_MONO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_MONO; + break; + case 2: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_STEREO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO; + break; + case 8: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_8CH : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_8CH; + break; + default: + return -EINVAL; + } + + sw_codec = (driver_mode == DRIVERMODE_NORMAL) ? sw_cpu : 20; + + dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots, + sw_cpu); + if (ret) + return ret; + + dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots, + sw_codec); + if (ret) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dev_dbg(dev, "%s: Setup IF1 for FM-radio.\n", __func__); + fmt_if1 = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_I2S; + ret = ab8500_audio_setup_if1(codec_dai->codec, fmt_if1, 16, 1); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_ops mop500_ab8500_ops[] = { + { + .hw_params = mop500_ab8500_hw_params, + .shutdown = mop500_ab8500_shutdown, + } +}; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct device *dev = rtd->card->dev; + struct mop500_audio_platform_data *pdata; + int ret; + + dev_dbg(dev, "%s Enter.\n", __func__); + + /* Add controls */ + ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls, + ARRAY_SIZE(mop500_ab8500_ctrls)); + if (ret < 0) { + pr_err("%s: Failed to add machine-controls (%d)!\n", + __func__, ret); + return ret; + } + + /* Setup AB8500 according to board-settings */ + pdata = (struct mop500_audio_platform_data *)dev_get_platdata(dev); + ret = ab8500_audio_setup_mics(codec, &pdata->amics); + if (ret < 0) { + pr_err("%s: Failed to setup mics (%d)!\n", __func__, ret); + return ret; + } + ret = ab8500_audio_set_ear_cmv(codec, pdata->ear_cmv); + if (ret < 0) { + pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6"); + + return ret; +} + diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h new file mode 100644 index 0000000..642b5dd --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.h @@ -0,0 +1,21 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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. + */ + +#ifndef MOP500_AB8500_H +#define MOP500_AB8500_H + +extern struct snd_soc_ops mop500_ab8500_ops[]; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime); + +#endif -- 1.7.8.3

2 1

[alsa-devel] [PATCH 09/11] ASoC: Ux500: Add platform-driver
by Ola Lilja 09 May '12

09 May '12

Add platform-driver handling all DMA-activities. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/ux500/Kconfig | 7 + sound/soc/ux500/Makefile | 3 + sound/soc/ux500/ux500_pcm.c | 320 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/ux500_pcm.h | 35 +++++ 4 files changed, 365 insertions(+), 0 deletions(-) create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 9b0ae4e..59da6ce 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -14,3 +14,10 @@ config SND_SOC_UX500_PLAT_MSP_I2S help Say Y if you want to enable the Ux500 MSP I2S-driver. +config SND_SOC_UX500_PLAT_DMA + tristate "Platform - DB8500 (DMA)" + depends on SND_SOC_UX500 + select SND_SOC_DMAENGINE_PCM + help + Say Y if you want to enable the Ux500 platform-driver. + diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index d8e9dd2..0c66a44 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -3,4 +3,7 @@ snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o +snd-soc-ux500-plat-dma-objs := ux500_pcm.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o + diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c new file mode 100644 index 0000000..0d944d5 --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.c @@ -0,0 +1,320 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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 <asm/page.h> + +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/slab.h> + +#include <plat/ste_dma40.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "ux500_msp_i2s.h" +#include "ux500_pcm.h" + +static struct snd_pcm_hardware ux500_pcm_hw_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK, + .rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static struct snd_pcm_hardware ux500_pcm_hw_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE, + .rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static void ux500_pcm_dma_hw_free(struct device *dev, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + + if (runtime->dma_area == NULL) + return; + + if (buf != &substream->dma_buffer) { + dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, + buf->addr); + kfree(runtime->dma_buffer_p); + } + + snd_pcm_set_runtime_buffer(substream, NULL); +} + +static int ux500_pcm_open(struct snd_pcm_substream *substream) +{ + int stream_id = substream->pstr->stream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct device *dev = dai->dev; + int ret; + struct ux500_msp_dma_params *dma_params; + u16 per_data_width, mem_data_width; + struct stedma40_chan_cfg *dma_cfg; + + dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + dev_dbg(dev, "%s: Set runtime hwparams.\n", __func__); + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_playback); + else + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_capture); + + /* 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) { + dev_err(dev, "%s: Error: snd_pcm_hw_constraints failed (%d)\n", + __func__, ret); + return ret; + } + + dev_dbg(dev, "%s: Set hw-struct for %s.\n", __func__, + snd_pcm_stream_str(substream)); + runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ? + ux500_pcm_hw_playback : ux500_pcm_hw_capture; + + mem_data_width = STEDMA40_HALFWORD_WIDTH; + + dma_params = snd_soc_dai_get_dma_data(dai, substream); + switch (dma_params->data_size) { + case 32: + per_data_width = STEDMA40_WORD_WIDTH; + break; + case 16: + per_data_width = STEDMA40_HALFWORD_WIDTH; + break; + case 8: + per_data_width = STEDMA40_BYTE_WIDTH; + break; + default: + per_data_width = STEDMA40_WORD_WIDTH; + dev_warn(rtd->platform->dev, + "%s: Unknown data-size (%d)! Assuming 32 bits.\n", + __func__, dma_params->data_size); + } + + dma_cfg = dma_params->dma_cfg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_cfg->src_info.data_width = mem_data_width; + dma_cfg->dst_info.data_width = per_data_width; + } else { + dma_cfg->src_info.data_width = per_data_width; + dma_cfg->dst_info.data_width = mem_data_width; + } + + + ret = snd_dmaengine_pcm_open(substream, stedma40_filter, dma_cfg); + if (ret) { + dev_dbg(dai->dev, + "%s: ERROR: snd_dmaengine_pcm_open failed (%d)!\n", + __func__, ret); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_cfg); + + return 0; +} + +static int ux500_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + + dev_dbg(dai->dev, "%s: Enter\n", __func__); + + snd_dmaengine_pcm_close(substream); + + return 0; +} + +static int ux500_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + int size; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + size = params_buffer_bytes(hw_params); + + if (buf) { + if (buf->bytes >= size) + goto out; + ux500_pcm_dma_hw_free(NULL, substream); + } + + if (substream->dma_buffer.area != NULL && + substream->dma_buffer.bytes >= size) { + buf = &substream->dma_buffer; + } else { + buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); + if (!buf) + goto nomem; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = NULL; + buf->area = dma_alloc_coherent(NULL, size, &buf->addr, + GFP_KERNEL); + buf->bytes = size; + buf->private_data = NULL; + + if (!buf->area) + goto free; + } + snd_pcm_set_runtime_buffer(substream, buf); + ret = 1; + out: + runtime->dma_bytes = size; + return ret; + + free: + kfree(buf); + nomem: + return -ENOMEM; +} + +static int ux500_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + ux500_pcm_dma_hw_free(NULL, substream); + + return 0; +} + +static int ux500_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__); + + return dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); +} + +static struct snd_pcm_ops ux500_pcm_ops = { + .open = ux500_pcm_open, + .close = ux500_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ux500_pcm_hw_params, + .hw_free = ux500_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = ux500_pcm_mmap +}; + +int ux500_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__); + + pcm->info_flags = 0; + strcpy(pcm->name, "UX500_PCM"); + dev_dbg(rtd->platform->dev, "%s: PCM-name: %s\n", __func__, pcm->name); + + return 0; +} + +static struct snd_soc_platform_driver ux500_pcm_soc_drv = { + .ops = &ux500_pcm_ops, + .pcm_new = ux500_pcm_new, +}; + +static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev) +{ + int ret; + + ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv); + if (ret < 0) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to register platform '%s' (%d)!\n", + __func__, pdev->name, ret); + return ret; + } + + return 0; +} + +static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver ux500_pcm_driver = { + .driver = { + .name = "ux500-pcm", + .owner = THIS_MODULE, + }, + + .probe = ux500_pcm_drv_probe, + .remove = __devexit_p(ux500_pcm_drv_remove), +}; +module_platform_driver(ux500_pcm_driver); + +MODULE_LICENSE("GPLv2"); + diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h new file mode 100644 index 0000000..77ed44d --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.h @@ -0,0 +1,35 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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. + */ +#ifndef UX500_PCM_H +#define UX500_PCM_H + +#include <asm/page.h> + +#include <linux/workqueue.h> + +#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000 +#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000 +#define UX500_PLATFORM_MIN_RATE_CAPTURE 8000 +#define UX500_PLATFORM_MAX_RATE_CAPTURE 48000 + +#define UX500_PLATFORM_MIN_CHANNELS 1 +#define UX500_PLATFORM_MAX_CHANNELS 8 + +#define UX500_PLATFORM_PERIODS_BYTES_MIN 128 +#define UX500_PLATFORM_PERIODS_BYTES_MAX (64 * PAGE_SIZE) +#define UX500_PLATFORM_PERIODS_MIN 2 +#define UX500_PLATFORM_PERIODS_MAX 48 +#define UX500_PLATFORM_BUFFER_BYTES_MAX (2048 * PAGE_SIZE) + +#endif -- 1.7.8.3

2 2

[alsa-devel] [PATCH 0/4] omap-abe-twl6040: Device tree support
by Peter Ujfalusi 09 May '12

09 May '12

Hello, The following series adds device tree support for OMAP4+ based boards with twl6040 codec. The first three patch is for preparing the driver to be used with DT based probing. The dts structure used for this machine driver uses some of the features already available in the core (thanks to the Tegra DT support). The DAPM routing information is coming from dts. The DAI link configuration also comes via DTS. In order to handle the different number of PCMs (dai links) the dts file has to tell the number of links (abe-twl6040,number-of-links) and has to contain the same number of dai link phandle configuration in the form of: dai-linkX,codec = <phandle to codec>; dai-linkX,dai = <phandle to CPU dai>; Where X is a number from 1... In this way the driver can enumerate and fix up the dai link structure. As an example of the dts section (SDP4430): sound { /* ASoC */ compatible = "ti,abe-twl6040"; abe-twl6040,model = "SDP4430"; abe-twl6040,jack_detection = <1>; abe-twl6040,mclk_freq = <38400000>; /* Number of DAI link connections */ abe-twl6040,number-of-links = <2>; dai-link1,codec = <&twl6040_codec>; dai-link1,dai = <&mcpdm>; dai-link2,codec = <&dmic_codec>; dai-link2,dai = <&dmic>; /* Audio routing */ abe-twl6040,audio-routing = "Headset Stereophone", "HSOL", "Headset Stereophone", "HSOR", "Earphone Spk", "EP", "Ext Spk", "HFL", "Ext Spk", "HFR", "Line Out", "AUXL", "Line Out", "AUXR", "Vibrator", "VIBRAL", "Vibrator", "VIBRAR", "HSMIC", "Headset Mic", "Headset Mic", "Headset Mic Bias", "MAINMIC", "Main Handset Mic", "Main Handset Mic", "Main Mic Bias", "SUBMIC", "Sub Handset Mic", "Sub Handset Mic", "Main Mic Bias", "AFML", "Line In", "AFMR", "Line In", "DMic", "Digital Mic", "Digital Mic", "Digital Mic1 Bias"; }; Regards, Peter --- Peter Ujfalusi (4): ASoC: omap-abe-twl6040: Keep only one snd_soc_dai_link structure ASoC: omap-abe-twl6040: Move Digital Mic widget into dapm table ASoC: omap-abe-twl6040: Introduce driver data for runtime parameters ASoC: omap-abe-twl6040: Add device tree support .../devicetree/bindings/sound/omap-abe-twl6040.txt | 57 ++++++ sound/soc/omap/omap-abe-twl6040.c | 199 ++++++++++++++----- 2 files changed, 203 insertions(+), 53 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt -- 1.7.8.6

2 10

Re: [alsa-devel] Fast Track Ultra latency and new streaming logic.
by Felix Homann 09 May '12

09 May '12

(For the list..) > > Is this a > flaw in Alsa, Jack or the USB stack? I'm starting to think that this is a Jack issue. At least I can set rather small period and buffer sizes in aplay without any xruns.

1 0

[alsa-devel] ASoC fix for 3.4
by Mark Brown 09 May '12

09 May '12

The following changes since commit 06412088ce98f745405b8f65cfc51ddd6b842bbf: ASoC: s3c2412-i2s: Fix dai registration (2012-04-30 23:45:25 +0100) 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 c8587193ba511b788a9888e5e701a9747e70c0d8: ASoC: sh: fix migor.c compilation (2012-05-09 12:41:05 +0100) ---------------------------------------------------------------- ASoC: Build fix for SH in 3.4 An API update which wasn't sufficiently thorough in updating the tree... ---------------------------------------------------------------- Guennadi Liakhovetski (1): ASoC: sh: fix migor.c compilation sound/soc/sh/migor.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

2 1

[alsa-devel] [PATCH 08/11] ASoC: Ux500: Add MSP I2S-driver
by Ola Lilja 09 May '12

09 May '12

Add driver for running I2S with the MSP-block. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/ux500/Kconfig | 16 + sound/soc/ux500/Makefile | 6 + sound/soc/ux500/ux500_msp_dai.c | 843 +++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/ux500_msp_dai.h | 79 ++++ sound/soc/ux500/ux500_msp_i2s.c | 742 ++++++++++++++++++++++++++++++++++ sound/soc/ux500/ux500_msp_i2s.h | 553 +++++++++++++++++++++++++ 8 files changed, 2241 insertions(+), 0 deletions(-) create mode 100644 sound/soc/ux500/Kconfig create mode 100644 sound/soc/ux500/Makefile create mode 100644 sound/soc/ux500/ux500_msp_dai.c create mode 100644 sound/soc/ux500/ux500_msp_dai.h create mode 100644 sound/soc/ux500/ux500_msp_i2s.c create mode 100644 sound/soc/ux500/ux500_msp_i2s.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 91c9855..359eb0f 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -48,6 +48,7 @@ source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" +source "sound/soc/ux500/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 2feaf37..f5fe621 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -25,3 +25,4 @@ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ +obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig new file mode 100644 index 0000000..9b0ae4e --- /dev/null +++ b/sound/soc/ux500/Kconfig @@ -0,0 +1,16 @@ +# +# Ux500 SoC audio configuration +# +menuconfig SND_SOC_UX500 + tristate "SoC Audio support for Ux500 platform" + depends on SND_SOC + help + Say Y if you want to enable ASoC-support for + any of the Ux500 platforms (e.g. U8500). + +config SND_SOC_UX500_PLAT_MSP_I2S + tristate "Platform - MSP (I2S)" + depends on SND_SOC_UX500 + help + Say Y if you want to enable the Ux500 MSP I2S-driver. + diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile new file mode 100644 index 0000000..d8e9dd2 --- /dev/null +++ b/sound/soc/ux500/Makefile @@ -0,0 +1,6 @@ +# Ux500 Platform Support + +snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o + + diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c new file mode 100644 index 0000000..93c6c40 --- /dev/null +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -0,0 +1,843 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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/slab.h> +#include <linux/bitops.h> +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/regulator/consumer.h> +#include <linux/mfd/dbx500-prcmu.h> + +#include <mach/hardware.h> +#include <mach/board-mop500-msp.h> + +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "ux500_msp_i2s.h" +#include "ux500_msp_dai.h" + +static int setup_pcm_multichan(struct snd_soc_dai *dai, + struct ux500_msp_config *msp_config) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct msp_multichannel_config *multi = + &msp_config->multichannel_config; + + if (drvdata->slots > 1) { + msp_config->multichannel_configured = 1; + + multi->tx_multichannel_enable = true; + multi->rx_multichannel_enable = true; + multi->rx_comparison_enable_mode = MSP_COMPARISON_DISABLED; + + multi->tx_channel_0_enable = drvdata->tx_mask; + multi->tx_channel_1_enable = 0; + multi->tx_channel_2_enable = 0; + multi->tx_channel_3_enable = 0; + + multi->rx_channel_0_enable = drvdata->rx_mask; + multi->rx_channel_1_enable = 0; + multi->rx_channel_2_enable = 0; + multi->rx_channel_3_enable = 0; + + dev_dbg(dai->dev, + "%s: Multichannel enabled. Slots: %d, TX: %u, RX: %u\n", + __func__, drvdata->slots, multi->tx_channel_0_enable, + multi->rx_channel_0_enable); + } + + return 0; +} + +static int setup_frameper(struct snd_soc_dai *dai, unsigned int rate, + struct msp_protdesc *prot_desc) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + switch (drvdata->slots) { + case 1: + switch (rate) { + case 8000: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_8_KHZ; + break; + + case 16000: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_16_KHZ; + break; + + case 44100: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_44_1_KHZ; + break; + + case 48000: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_48_KHZ; + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported sample-rate (freq = %d)!\n", + __func__, rate); + return -EINVAL; + } + break; + + case 2: + prot_desc->frame_period = FRAME_PER_2_SLOTS; + break; + + case 8: + prot_desc->frame_period = FRAME_PER_8_SLOTS; + break; + + case 16: + prot_desc->frame_period = FRAME_PER_16_SLOTS; + break; + default: + dev_err(dai->dev, + "%s: Error: Unsupported slot-count (slots = %d)!\n", + __func__, drvdata->slots); + return -EINVAL; + } + + prot_desc->clocks_per_frame = + prot_desc->frame_period+1; + + dev_dbg(dai->dev, "%s: Clocks per frame: %u\n", + __func__, + prot_desc->clocks_per_frame); + + return 0; +} + +static int setup_pcm_framing(struct snd_soc_dai *dai, unsigned int rate, + struct msp_protdesc *prot_desc) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + u32 frame_length = MSP_FRAME_LEN_1; + prot_desc->frame_width = 0; + + switch (drvdata->slots) { + case 1: + frame_length = MSP_FRAME_LEN_1; + break; + + case 2: + frame_length = MSP_FRAME_LEN_2; + break; + + case 8: + frame_length = MSP_FRAME_LEN_8; + break; + + case 16: + frame_length = MSP_FRAME_LEN_16; + break; + default: + dev_err(dai->dev, + "%s: Error: Unsupported slot-count (slots = %d)!\n", + __func__, drvdata->slots); + return -EINVAL; + } + + prot_desc->tx_frame_len_1 = frame_length; + prot_desc->rx_frame_len_1 = frame_length; + prot_desc->tx_frame_len_2 = frame_length; + prot_desc->rx_frame_len_2 = frame_length; + + prot_desc->tx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->rx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->tx_elem_len_2 = MSP_ELEM_LEN_16; + prot_desc->rx_elem_len_2 = MSP_ELEM_LEN_16; + + return setup_frameper(dai, rate, prot_desc); +} + +static int setup_clocking(struct snd_soc_dai *dai, + unsigned int fmt, + struct ux500_msp_config *msp_config) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + + case SND_SOC_DAIFMT_NB_IF: + msp_config->tx_fsync_pol ^= 1 << TFSPOL_SHIFT; + msp_config->rx_fsync_pol ^= 1 << RFSPOL_SHIFT; + + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsopported inversion (fmt = 0x%x)!\n", + __func__, fmt); + + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + dev_dbg(dai->dev, "%s: Codec is master.\n", __func__); + + msp_config->iodelay = 0x20; + msp_config->rx_fsync_sel = 0; + msp_config->tx_fsync_sel = 1 << TFSSEL_SHIFT; + msp_config->tx_clk_sel = 0; + msp_config->rx_clk_sel = 0; + msp_config->srg_clk_sel = 0x2 << SCKSEL_SHIFT; + + break; + + case SND_SOC_DAIFMT_CBS_CFS: + dev_dbg(dai->dev, "%s: Codec is slave.\n", __func__); + + msp_config->tx_clk_sel = TX_CLK_SEL_SRG; + msp_config->tx_fsync_sel = TX_SYNC_SRG_PROG; + msp_config->rx_clk_sel = RX_CLK_SEL_SRG; + msp_config->rx_fsync_sel = RX_SYNC_SRG; + msp_config->srg_clk_sel = 1 << SCKSEL_SHIFT; + + break; + + default: + dev_err(dai->dev, "%s: Error: Unsopported master (fmt = 0x%x)!\n", + __func__, fmt); + + return -EINVAL; + } + + return 0; +} + +static int setup_pcm_protdesc(struct snd_soc_dai *dai, + unsigned int fmt, + struct msp_protdesc *prot_desc) +{ + prot_desc->rx_phase_mode = MSP_SINGLE_PHASE; + prot_desc->tx_phase_mode = MSP_SINGLE_PHASE; + prot_desc->rx_phase2_start_mode = MSP_PHASE2_START_MODE_IMEDIATE; + prot_desc->tx_phase2_start_mode = MSP_PHASE2_START_MODE_IMEDIATE; + prot_desc->rx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_fsync_pol = MSP_FSYNC_POL(MSP_FSYNC_POL_ACT_HI); + prot_desc->rx_fsync_pol = MSP_FSYNC_POL_ACT_HI << RFSPOL_SHIFT; + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_DSP_A) { + dev_dbg(dai->dev, "%s: DSP_A.\n", __func__); + prot_desc->rx_clk_pol = MSP_RISING_EDGE; + prot_desc->tx_clk_pol = MSP_FALLING_EDGE; + + prot_desc->rx_data_delay = MSP_DELAY_1; + prot_desc->tx_data_delay = MSP_DELAY_1; + } else { + dev_dbg(dai->dev, "%s: DSP_B.\n", __func__); + prot_desc->rx_clk_pol = MSP_FALLING_EDGE; + prot_desc->tx_clk_pol = MSP_RISING_EDGE; + + prot_desc->rx_data_delay = MSP_DELAY_0; + prot_desc->tx_data_delay = MSP_DELAY_0; + } + + prot_desc->rx_half_word_swap = MSP_SWAP_NONE; + prot_desc->tx_half_word_swap = MSP_SWAP_NONE; + prot_desc->compression_mode = MSP_COMPRESS_MODE_LINEAR; + prot_desc->expansion_mode = MSP_EXPAND_MODE_LINEAR; + prot_desc->frame_sync_ignore = MSP_FSYNC_IGNORE; + + return 0; +} + +static int setup_i2s_protdesc(struct msp_protdesc *prot_desc) +{ + prot_desc->rx_phase_mode = MSP_DUAL_PHASE; + prot_desc->tx_phase_mode = MSP_DUAL_PHASE; + prot_desc->rx_phase2_start_mode = MSP_PHASE2_START_MODE_FSYNC; + prot_desc->tx_phase2_start_mode = MSP_PHASE2_START_MODE_FSYNC; + prot_desc->rx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_fsync_pol = MSP_FSYNC_POL(MSP_FSYNC_POL_ACT_LO); + prot_desc->rx_fsync_pol = MSP_FSYNC_POL_ACT_LO << RFSPOL_SHIFT; + + prot_desc->rx_frame_len_1 = MSP_FRAME_LEN_1; + prot_desc->rx_frame_len_2 = MSP_FRAME_LEN_1; + prot_desc->tx_frame_len_1 = MSP_FRAME_LEN_1; + prot_desc->tx_frame_len_2 = MSP_FRAME_LEN_1; + prot_desc->rx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->rx_elem_len_2 = MSP_ELEM_LEN_16; + prot_desc->tx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->tx_elem_len_2 = MSP_ELEM_LEN_16; + + prot_desc->rx_clk_pol = MSP_RISING_EDGE; + prot_desc->tx_clk_pol = MSP_FALLING_EDGE; + + prot_desc->rx_data_delay = MSP_DELAY_0; + prot_desc->tx_data_delay = MSP_DELAY_0; + + prot_desc->tx_half_word_swap = MSP_SWAP_NONE; + prot_desc->rx_half_word_swap = MSP_SWAP_NONE; + prot_desc->compression_mode = MSP_COMPRESS_MODE_LINEAR; + prot_desc->expansion_mode = MSP_EXPAND_MODE_LINEAR; + prot_desc->frame_sync_ignore = MSP_FSYNC_IGNORE; + + return 0; +} + +static int setup_msp_config(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai, + struct ux500_msp_config *msp_config) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct msp_protdesc *prot_desc = &msp_config->protdesc; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int fmt = drvdata->fmt; + int ret; + + memset(msp_config, 0, sizeof(*msp_config)); + + msp_config->f_inputclk = drvdata->master_clk; + + msp_config->tx_fifo_config = TX_FIFO_ENABLE; + msp_config->rx_fifo_config = RX_FIFO_ENABLE; + msp_config->def_elem_len = 1; + msp_config->direction = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + MSP_DIR_TX : MSP_DIR_RX; + msp_config->data_size = MSP_DATA_BITS_32; + msp_config->frame_freq = runtime->rate; + + dev_dbg(dai->dev, "%s: f_inputclk = %u, frame_freq = %u.\n", + __func__, msp_config->f_inputclk, msp_config->frame_freq); + /* To avoid division by zero */ + prot_desc->clocks_per_frame = 1; + + dev_dbg(dai->dev, "%s: rate: %u, channels: %d.\n", __func__, + runtime->rate, runtime->channels); + switch (fmt & + (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + dev_dbg(dai->dev, "%s: SND_SOC_DAIFMT_I2S.\n", __func__); + + msp_config->default_protdesc = 1; + msp_config->protocol = MSP_I2S_PROTOCOL; + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + dev_dbg(dai->dev, "%s: SND_SOC_DAIFMT_I2S.\n", __func__); + + msp_config->data_size = MSP_DATA_BITS_16; + msp_config->protocol = MSP_I2S_PROTOCOL; + + ret = setup_i2s_protdesc(prot_desc); + if (ret < 0) + return ret; + + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM: + dev_dbg(dai->dev, "%s: PCM format.\n", __func__); + + msp_config->data_size = MSP_DATA_BITS_16; + msp_config->protocol = MSP_PCM_PROTOCOL; + + ret = setup_pcm_protdesc(dai, fmt, prot_desc); + if (ret < 0) + return ret; + + ret = setup_pcm_multichan(dai, msp_config); + if (ret < 0) + return ret; + + ret = setup_pcm_framing(dai, runtime->rate, prot_desc); + if (ret < 0) + return ret; + + break; + + default: + dev_err(dai->dev, "%s: Error: Unsopported format (%d)!\n", + __func__, fmt); + return -EINVAL; + } + + return setup_clocking(dai, fmt, msp_config); +} + +static int ux500_msp_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + /* Enable regulator */ + ret = regulator_enable(drvdata->reg_vape); + if (ret != 0) { + dev_err(drvdata->msp->dev, + "%s: Failed to enable regulator!\n", __func__); + return ret; + } + + /* Enable clock */ + dev_dbg(dai->dev, "%s: Enabling MSP-clock.\n", __func__); + clk_enable(drvdata->clk); + + return 0; +} + +static void ux500_msp_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + if (drvdata->vape_opp_constraint == 1) { + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, + "ux500_msp_i2s", 50); + drvdata->vape_opp_constraint = 0; + } + + if (ux500_msp_i2s_close(drvdata->msp, + is_playback ? MSP_DIR_TX : MSP_DIR_RX)) { + dev_err(dai->dev, + "%s: Error: MSP %d (%s): Unable to close i2s.\n", + __func__, dai->id, snd_pcm_stream_str(substream)); + } + + /* Disable clock */ + clk_disable(drvdata->clk); + + /* Disable regulator */ + ret = regulator_disable(drvdata->reg_vape); + if (ret < 0) + dev_err(dai->dev, + "%s: ERROR: Failed to disable regulator (%d)!\n", + __func__, ret); +} + +static int ux500_msp_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct snd_pcm_runtime *runtime = substream->runtime; + struct ux500_msp_config msp_config; + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter (rate = %d).\n", __func__, + dai->id, snd_pcm_stream_str(substream), runtime->rate); + + setup_msp_config(substream, dai, &msp_config); + + ret = ux500_msp_i2s_open(drvdata->msp, &msp_config); + if (ret < 0) { + dev_err(dai->dev, "%s: Error: msp_setup failed (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Set OPP-level */ + if ((drvdata->fmt & SND_SOC_DAIFMT_MASTER_MASK) && + (drvdata->msp->f_bitclk > 19200000)) { + /* If the bit-clock is higher than 19.2MHz, Vape should be + * run in 100% OPP. Only when bit-clock is used (MSP master) */ + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, + "ux500-msp-i2s", 100); + drvdata->vape_opp_constraint = 1; + } else { + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, + "ux500-msp-i2s", 50); + drvdata->vape_opp_constraint = 0; + } + + return ret; +} + +static int ux500_msp_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + unsigned int mask, slots_active; + struct snd_pcm_runtime *runtime = substream->runtime; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", + __func__, dai->id, snd_pcm_stream_str(substream)); + + switch (drvdata->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 1, 2); + break; + + case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_DSP_A: + mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + drvdata->tx_mask : + drvdata->rx_mask; + + slots_active = hweight32(mask); + dev_dbg(dai->dev, "TDM-slots active: %d", slots_active); + + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + slots_active, slots_active); + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported protocol (fmt = 0x%x)!\n", + __func__, drvdata->fmt); + return -EINVAL; + } + + return 0; +} + +static int ux500_msp_dai_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d: Enter.\n", __func__, dai->id); + + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported protocol/master (fmt = 0x%x)!\n", + __func__, drvdata->fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_NB_IF: + case SND_SOC_DAIFMT_IB_IF: + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported inversion (fmt = 0x%x)!\n", + __func__, drvdata->fmt); + return -EINVAL; + } + + drvdata->fmt = fmt; + return 0; +} + +static int ux500_msp_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + unsigned int cap; + + switch (slots) { + case 1: + cap = 0x01; + break; + case 2: + cap = 0x03; + break; + case 8: + cap = 0xFF; + break; + case 16: + cap = 0xFFFF; + break; + default: + dev_err(dai->dev, "%s: Error: Unsupported slot-count (%d)!\n", + __func__, slots); + return -EINVAL; + } + drvdata->slots = slots; + + if (!(slot_width == 16)) { + dev_err(dai->dev, "%s: Error: Unsupported slot-width (%d)!\n", + __func__, slot_width); + return -EINVAL; + } + drvdata->slot_width = slot_width; + + drvdata->tx_mask = tx_mask & cap; + drvdata->rx_mask = rx_mask & cap; + + return 0; +} + +static int ux500_msp_dai_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d: Enter. clk-id: %d, freq: %u.\n", + __func__, dai->id, clk_id, freq); + + switch (clk_id) { + case UX500_MSP_MASTER_CLOCK: + drvdata->master_clk = freq; + break; + + default: + dev_err(dai->dev, "%s: MSP %d: Invalid clk-id (%d)!\n", + __func__, dai->id, clk_id); + return -EINVAL; + } + + return 0; +} + +static int ux500_msp_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int ret = 0; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter (msp->id = %d, cmd = %d).\n", + __func__, dai->id, snd_pcm_stream_str(substream), + (int)drvdata->msp->id, cmd); + + ret = ux500_msp_i2s_trigger(drvdata->msp, cmd, substream->stream); + + return ret; +} + +static int ux500_msp_dai_probe(struct snd_soc_dai *dai) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + drvdata->playback_dma_data.dma_cfg = drvdata->msp->dma_cfg_tx; + drvdata->capture_dma_data.dma_cfg = drvdata->msp->dma_cfg_rx; + + dai->playback_dma_data = &drvdata->playback_dma_data; + dai->capture_dma_data = &drvdata->capture_dma_data; + + drvdata->playback_dma_data.data_size = drvdata->slot_width; + drvdata->capture_dma_data.data_size = drvdata->slot_width; + + return 0; +} + +static struct snd_soc_dai_ops ux500_msp_dai_ops[] = { + { + .set_sysclk = ux500_msp_dai_set_dai_sysclk, + .set_fmt = ux500_msp_dai_set_dai_fmt, + .set_tdm_slot = ux500_msp_dai_set_tdm_slot, + .startup = ux500_msp_dai_startup, + .shutdown = ux500_msp_dai_shutdown, + .prepare = ux500_msp_dai_prepare, + .trigger = ux500_msp_dai_trigger, + .hw_params = ux500_msp_dai_hw_params, + } +}; + +static struct snd_soc_dai_driver ux500_msp_dai_drv[UX500_NBR_OF_DAI] = { + { + .name = "ux500-msp-i2s.0", + .probe = ux500_msp_dai_probe, + .id = 0, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, + { + .name = "ux500-msp-i2s.1", + .probe = ux500_msp_dai_probe, + .id = 1, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, + { + .name = "ux500-msp-i2s.2", + .id = 2, + .probe = ux500_msp_dai_probe, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, + { + .name = "ux500-msp-i2s.3", + .probe = ux500_msp_dai_probe, + .id = 3, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, +}; + +static int __devinit ux500_msp_drv_probe(struct platform_device *pdev) +{ + struct ux500_msp_i2s_drvdata *drvdata; + int ret = 0; + + dev_dbg(&pdev->dev, "%s: Enter (pdev->name = %s).\n", __func__, + pdev->name); + + drvdata = devm_kzalloc(&pdev->dev, + sizeof(struct ux500_msp_i2s_drvdata), + GFP_KERNEL); + drvdata->fmt = 0; + drvdata->slots = 1; + drvdata->tx_mask = 0x01; + drvdata->rx_mask = 0x01; + drvdata->slot_width = 16; + drvdata->master_clk = MSP_INPUT_FREQ_APB; + + drvdata->reg_vape = devm_regulator_get(&pdev->dev, "v-ape"); + if (IS_ERR(drvdata->reg_vape)) { + ret = (int)PTR_ERR(drvdata->reg_vape); + dev_err(&pdev->dev, + "%s: ERROR: Failed to get Vape supply (%d)!\n", + __func__, ret); + return ret; + } + prcmu_qos_add_requirement(PRCMU_QOS_APE_OPP, (char *)pdev->name, 50); + + drvdata->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(drvdata->clk)) { + ret = (int)PTR_ERR(drvdata->clk); + dev_err(&pdev->dev, "%s: ERROR: clk_get failed (%d)!\n", + __func__, ret); + goto err_clk; + } + + ret = ux500_msp_i2s_init_msp(pdev, &drvdata->msp, + pdev->dev.platform_data); + if (!drvdata->msp) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to init MSP-struct (%d)!", + __func__, ret); + goto err_init_msp; + } + dev_set_drvdata(&pdev->dev, drvdata); + + ret = snd_soc_register_dai(&pdev->dev, + &ux500_msp_dai_drv[drvdata->msp->id]); + if (ret < 0) { + dev_err(&pdev->dev, "Error: %s: Failed to register MSP%d!\n", + __func__, drvdata->msp->id); + goto err_init_msp; + } + + return 0; + +err_init_msp: + clk_put(drvdata->clk); + +err_clk: + devm_regulator_put(drvdata->reg_vape); + + return ret; +} + +static int __devexit ux500_msp_drv_remove(struct platform_device *pdev) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(ux500_msp_dai_drv)); + + devm_regulator_put(drvdata->reg_vape); + prcmu_qos_remove_requirement(PRCMU_QOS_APE_OPP, "ux500_msp_i2s"); + + clk_put(drvdata->clk); + + ux500_msp_i2s_cleanup_msp(pdev, drvdata->msp); + + return 0; +} + +static struct platform_driver msp_i2s_driver = { + .driver = { + .name = "ux500-msp-i2s", + .owner = THIS_MODULE, + }, + .probe = ux500_msp_drv_probe, + .remove = ux500_msp_drv_remove, +}; +module_platform_driver(msp_i2s_driver); + +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h new file mode 100644 index 0000000..98202a3 --- /dev/null +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -0,0 +1,79 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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. + */ + +#ifndef UX500_msp_dai_H +#define UX500_msp_dai_H + +#include <linux/types.h> +#include <linux/spinlock.h> + +#include "ux500_msp_i2s.h" + +#define UX500_NBR_OF_DAI 4 + +#define UX500_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define UX500_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +#define FRAME_PER_SINGLE_SLOT_8_KHZ 31 +#define FRAME_PER_SINGLE_SLOT_16_KHZ 124 +#define FRAME_PER_SINGLE_SLOT_44_1_KHZ 63 +#define FRAME_PER_SINGLE_SLOT_48_KHZ 49 +#define FRAME_PER_2_SLOTS 31 +#define FRAME_PER_8_SLOTS 138 +#define FRAME_PER_16_SLOTS 277 + +#ifndef CONFIG_SND_SOC_UX500_AB5500 +#define UX500_MSP_INTERNAL_CLOCK_FREQ 40000000 +#define UX500_MSP1_INTERNAL_CLOCK_FREQ UX500_MSP_INTERNAL_CLOCK_FREQ +#else +#define UX500_MSP_INTERNAL_CLOCK_FREQ 13000000 +#define UX500_MSP1_INTERNAL_CLOCK_FREQ (UX500_MSP_INTERNAL_CLOCK_FREQ * 2) +#endif + +#define UX500_MSP_MIN_CHANNELS 1 +#define UX500_MSP_MAX_CHANNELS 8 + +#define PLAYBACK_CONFIGURED 1 +#define CAPTURE_CONFIGURED 2 + +enum ux500_msp_clock_id { + UX500_MSP_MASTER_CLOCK, +}; + +struct ux500_msp_i2s_drvdata { + struct ux500_msp *msp; + struct regulator *reg_vape; + struct ux500_msp_dma_params playback_dma_data; + struct ux500_msp_dma_params capture_dma_data; + unsigned int fmt; + unsigned int tx_mask; + unsigned int rx_mask; + int slots; + int slot_width; + u8 configured; + int data_delay; + + /* Clocks */ + unsigned int master_clk; + struct clk *clk; + + /* Regulators */ + int vape_opp_constraint; +}; + +int ux500_msp_dai_set_data_delay(struct snd_soc_dai *dai, int delay); + +#endif diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c new file mode 100644 index 0000000..496dec1 --- /dev/null +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -0,0 +1,742 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com>, + * Sandeep Kaushik <sandeep.kaushik(a)st.com> + * for ST-Ericsson. + * + * License terms: + * + * 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/platform_device.h> +#include <linux/delay.h> +#include <linux/slab.h> + +#include <mach/hardware.h> +#include <mach/board-mop500-msp.h> + +#include <sound/soc.h> + +#include "ux500_msp_i2s.h" + + /* Protocol desciptors */ +static const struct msp_protdesc prot_descs[] = { + { /* I2S */ + MSP_SINGLE_PHASE, + MSP_SINGLE_PHASE, + MSP_PHASE2_START_MODE_IMEDIATE, + MSP_PHASE2_START_MODE_IMEDIATE, + MSP_BTF_MS_BIT_FIRST, + MSP_BTF_MS_BIT_FIRST, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_ELEM_LEN_32, + MSP_ELEM_LEN_32, + MSP_ELEM_LEN_32, + MSP_ELEM_LEN_32, + MSP_DELAY_1, + MSP_DELAY_1, + MSP_RISING_EDGE, + MSP_FALLING_EDGE, + MSP_FSYNC_POL_ACT_LO, + MSP_FSYNC_POL_ACT_LO, + MSP_SWAP_NONE, + MSP_SWAP_NONE, + MSP_COMPRESS_MODE_LINEAR, + MSP_EXPAND_MODE_LINEAR, + MSP_FSYNC_IGNORE, + 31, + 15, + 32, + }, { /* PCM */ + MSP_DUAL_PHASE, + MSP_DUAL_PHASE, + MSP_PHASE2_START_MODE_FSYNC, + MSP_PHASE2_START_MODE_FSYNC, + MSP_BTF_MS_BIT_FIRST, + MSP_BTF_MS_BIT_FIRST, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_ELEM_LEN_16, + MSP_ELEM_LEN_16, + MSP_ELEM_LEN_16, + MSP_ELEM_LEN_16, + MSP_DELAY_0, + MSP_DELAY_0, + MSP_RISING_EDGE, + MSP_FALLING_EDGE, + MSP_FSYNC_POL_ACT_HI, + MSP_FSYNC_POL_ACT_HI, + MSP_SWAP_NONE, + MSP_SWAP_NONE, + MSP_COMPRESS_MODE_LINEAR, + MSP_EXPAND_MODE_LINEAR, + MSP_FSYNC_IGNORE, + 255, + 0, + 256, + }, { /* Companded PCM */ + MSP_SINGLE_PHASE, + MSP_SINGLE_PHASE, + MSP_PHASE2_START_MODE_FSYNC, + MSP_PHASE2_START_MODE_FSYNC, + MSP_BTF_MS_BIT_FIRST, + MSP_BTF_MS_BIT_FIRST, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_ELEM_LEN_8, + MSP_ELEM_LEN_8, + MSP_ELEM_LEN_8, + MSP_ELEM_LEN_8, + MSP_DELAY_0, + MSP_DELAY_0, + MSP_RISING_EDGE, + MSP_RISING_EDGE, + MSP_FSYNC_POL_ACT_HI, + MSP_FSYNC_POL_ACT_HI, + MSP_SWAP_NONE, + MSP_SWAP_NONE, + MSP_COMPRESS_MODE_LINEAR, + MSP_EXPAND_MODE_LINEAR, + MSP_FSYNC_IGNORE, + 255, + 0, + 256, + }, +}; + +static void set_prot_desc_tx(struct ux500_msp *msp, + struct msp_protdesc *protdesc, + enum msp_data_size data_size) +{ + u32 temp_reg = 0; + + temp_reg |= MSP_P2_ENABLE_BIT(protdesc->tx_phase_mode); + temp_reg |= MSP_P2_START_MODE_BIT(protdesc->tx_phase2_start_mode); + temp_reg |= MSP_P1_FRAME_LEN_BITS(protdesc->tx_frame_len_1); + temp_reg |= MSP_P2_FRAME_LEN_BITS(protdesc->tx_frame_len_2); + if (msp->def_elem_len) { + temp_reg |= MSP_P1_ELEM_LEN_BITS(protdesc->tx_elem_len_1); + temp_reg |= MSP_P2_ELEM_LEN_BITS(protdesc->tx_elem_len_2); + } else { + temp_reg |= MSP_P1_ELEM_LEN_BITS(data_size); + temp_reg |= MSP_P2_ELEM_LEN_BITS(data_size); + } + temp_reg |= MSP_DATA_DELAY_BITS(protdesc->tx_data_delay); + temp_reg |= MSP_SET_ENDIANNES_BIT(protdesc->tx_byte_order); + temp_reg |= MSP_FSYNC_POL(protdesc->tx_fsync_pol); + temp_reg |= MSP_DATA_WORD_SWAP(protdesc->tx_half_word_swap); + temp_reg |= MSP_SET_COMPANDING_MODE(protdesc->compression_mode); + temp_reg |= MSP_SET_FSYNC_IGNORE(protdesc->frame_sync_ignore); + + writel(temp_reg, msp->registers + MSP_TCF); +} + +static void set_prot_desc_rx(struct ux500_msp *msp, + struct msp_protdesc *protdesc, + enum msp_data_size data_size) +{ + u32 temp_reg = 0; + + temp_reg |= MSP_P2_ENABLE_BIT(protdesc->rx_phase_mode); + temp_reg |= MSP_P2_START_MODE_BIT(protdesc->rx_phase2_start_mode); + temp_reg |= MSP_P1_FRAME_LEN_BITS(protdesc->rx_frame_len_1); + temp_reg |= MSP_P2_FRAME_LEN_BITS(protdesc->rx_frame_len_2); + if (msp->def_elem_len) { + temp_reg |= MSP_P1_ELEM_LEN_BITS(protdesc->rx_elem_len_1); + temp_reg |= MSP_P2_ELEM_LEN_BITS(protdesc->rx_elem_len_2); + } else { + temp_reg |= MSP_P1_ELEM_LEN_BITS(data_size); + temp_reg |= MSP_P2_ELEM_LEN_BITS(data_size); + } + + temp_reg |= MSP_DATA_DELAY_BITS(protdesc->rx_data_delay); + temp_reg |= MSP_SET_ENDIANNES_BIT(protdesc->rx_byte_order); + temp_reg |= MSP_FSYNC_POL(protdesc->rx_fsync_pol); + temp_reg |= MSP_DATA_WORD_SWAP(protdesc->rx_half_word_swap); + temp_reg |= MSP_SET_COMPANDING_MODE(protdesc->expansion_mode); + temp_reg |= MSP_SET_FSYNC_IGNORE(protdesc->frame_sync_ignore); + + writel(temp_reg, msp->registers + MSP_RCF); +} + +static int configure_protocol(struct ux500_msp *msp, + struct ux500_msp_config *config) +{ + struct msp_protdesc *protdesc; + enum msp_data_size data_size; + u32 temp_reg = 0; + + data_size = config->data_size; + msp->def_elem_len = config->def_elem_len; + if (config->default_protdesc == 1) { + if (config->protocol >= MSP_INVALID_PROTOCOL) { + dev_err(msp->dev, "%s: ERROR: Invalid protocol!\n", + __func__); + return -EINVAL; + } + protdesc = + (struct msp_protdesc *)&prot_descs[config->protocol]; + } else { + protdesc = (struct msp_protdesc *)&config->protdesc; + } + + if (data_size < MSP_DATA_BITS_DEFAULT || data_size > MSP_DATA_BITS_32) { + dev_err(msp->dev, + "%s: ERROR: Invalid data-size requested (data_size = %d)!\n", + __func__, data_size); + return -EINVAL; + } + + if (config->direction & MSP_DIR_TX) + set_prot_desc_tx(msp, protdesc, data_size); + if (config->direction & MSP_DIR_RX) + set_prot_desc_rx(msp, protdesc, data_size); + + /* The code below should not be separated. */ + temp_reg = readl(msp->registers + MSP_GCR) & ~TX_CLK_POL_RISING; + temp_reg |= MSP_TX_CLKPOL_BIT(~protdesc->tx_clk_pol); + writel(temp_reg, msp->registers + MSP_GCR); + temp_reg = readl(msp->registers + MSP_GCR) & ~RX_CLK_POL_RISING; + temp_reg |= MSP_RX_CLKPOL_BIT(protdesc->rx_clk_pol); + writel(temp_reg, msp->registers + MSP_GCR); + + return 0; +} + +static int setup_bitclk(struct ux500_msp *msp, struct ux500_msp_config *config) +{ + u32 reg_val_GCR; + u32 frame_per = 0; + u32 sck_div = 0; + u32 frame_width = 0; + u32 temp_reg = 0; + struct msp_protdesc *protdesc = NULL; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~SRG_ENABLE, msp->registers + MSP_GCR); + + if (config->default_protdesc) + protdesc = + (struct msp_protdesc *)&prot_descs[config->protocol]; + else + protdesc = (struct msp_protdesc *)&config->protdesc; + + switch (config->protocol) { + case MSP_PCM_PROTOCOL: + case MSP_PCM_COMPAND_PROTOCOL: + frame_width = protdesc->frame_width; + sck_div = config->f_inputclk / (config->frame_freq * + (protdesc->clocks_per_frame)); + frame_per = protdesc->frame_period; + break; + case MSP_I2S_PROTOCOL: + frame_width = protdesc->frame_width; + sck_div = config->f_inputclk / (config->frame_freq * + (protdesc->clocks_per_frame)); + frame_per = protdesc->frame_period; + break; + default: + dev_err(msp->dev, "%s: ERROR: Unknown protocol (%d)!\n", + __func__, + config->protocol); + return -EINVAL; + } + + temp_reg = (sck_div - 1) & SCK_DIV_MASK; + temp_reg |= FRAME_WIDTH_BITS(frame_width); + temp_reg |= FRAME_PERIOD_BITS(frame_per); + writel(temp_reg, msp->registers + MSP_SRG); + + msp->f_bitclk = (config->f_inputclk)/(sck_div + 1); + + /* Enable bit-clock */ + udelay(100); + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | SRG_ENABLE, msp->registers + MSP_GCR); + udelay(100); + + return 0; +} + +static int configure_multichannel(struct ux500_msp *msp, + struct ux500_msp_config *config) +{ + struct msp_protdesc *protdesc; + struct msp_multichannel_config *mcfg; + u32 reg_val_MCR; + + if (config->default_protdesc == 1) { + if (config->protocol >= MSP_INVALID_PROTOCOL) { + dev_err(msp->dev, + "%s: ERROR: Invalid protocol (%d)!\n", + __func__, config->protocol); + return -EINVAL; + } + protdesc = (struct msp_protdesc *) + &prot_descs[config->protocol]; + } else { + protdesc = (struct msp_protdesc *)&config->protdesc; + } + + mcfg = &config->multichannel_config; + if (mcfg->tx_multichannel_enable) { + if (protdesc->tx_phase_mode == MSP_SINGLE_PHASE) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | (mcfg->tx_multichannel_enable ? + 1 << TMCEN_BIT : 0), + msp->registers + MSP_MCR); + writel(mcfg->tx_channel_0_enable, + msp->registers + MSP_TCE0); + writel(mcfg->tx_channel_1_enable, + msp->registers + MSP_TCE1); + writel(mcfg->tx_channel_2_enable, + msp->registers + MSP_TCE2); + writel(mcfg->tx_channel_3_enable, + msp->registers + MSP_TCE3); + } else { + dev_err(msp->dev, + "%s: ERROR: Only single-phase supported (TX-mode: %d)!\n", + __func__, protdesc->tx_phase_mode); + return -EINVAL; + } + } + if (mcfg->rx_multichannel_enable) { + if (protdesc->rx_phase_mode == MSP_SINGLE_PHASE) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | (mcfg->rx_multichannel_enable ? + 1 << RMCEN_BIT : 0), + msp->registers + MSP_MCR); + writel(mcfg->rx_channel_0_enable, + msp->registers + MSP_RCE0); + writel(mcfg->rx_channel_1_enable, + msp->registers + MSP_RCE1); + writel(mcfg->rx_channel_2_enable, + msp->registers + MSP_RCE2); + writel(mcfg->rx_channel_3_enable, + msp->registers + MSP_RCE3); + } else { + dev_err(msp->dev, + "%s: ERROR: Only single-phase supported (RX-mode: %d)!\n", + __func__, protdesc->rx_phase_mode); + return -EINVAL; + } + if (mcfg->rx_comparison_enable_mode) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | + (mcfg->rx_comparison_enable_mode << RCMPM_BIT), + msp->registers + MSP_MCR); + + writel(mcfg->comparison_mask, + msp->registers + MSP_RCM); + writel(mcfg->comparison_value, + msp->registers + MSP_RCV); + + } + } + + return 0; +} + +static int enable_msp(struct ux500_msp *msp, struct ux500_msp_config *config) +{ + int status = 0; + u32 reg_val_DMACR, reg_val_GCR; + + /* Check msp state whether in RUN or CONFIGURED Mode */ + if ((msp->msp_state == MSP_STATE_IDLE) && (msp->plat_init)) { + status = msp->plat_init(); + if (status) { + dev_err(msp->dev, "%s: ERROR: Failed to init MSP (%d)!\n", + __func__, status); + return status; + } + } + + /* Configure msp with protocol dependent settings */ + configure_protocol(msp, config); + setup_bitclk(msp, config); + if (config->multichannel_configured == 1) { + status = configure_multichannel(msp, config); + if (status) + dev_warn(msp->dev, + "%s: WARN: configure_multichannel failed (%d)!\n", + __func__, status); + } + + /* Make sure the correct DMA-directions are configured */ + if ((config->direction & MSP_DIR_RX) && (!msp->dma_cfg_rx)) { + dev_err(msp->dev, "%s: ERROR: MSP RX-mode is not configured!", + __func__); + return -EINVAL; + } + if ((config->direction == MSP_DIR_TX) && (!msp->dma_cfg_tx)) { + dev_err(msp->dev, "%s: ERROR: MSP TX-mode is not configured!", + __func__); + return -EINVAL; + } + + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + if (config->direction & MSP_DIR_RX) + reg_val_DMACR |= RX_DMA_ENABLE; + if (config->direction & MSP_DIR_TX) + reg_val_DMACR |= TX_DMA_ENABLE; + writel(reg_val_DMACR, msp->registers + MSP_DMACR); + + writel(config->iodelay, msp->registers + MSP_IODLY); + + /* Enable frame generation logic */ + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | FRAME_GEN_ENABLE, msp->registers + MSP_GCR); + + return status; +} + +static void flush_fifo_rx(struct ux500_msp *msp) +{ + u32 reg_val_DR, reg_val_GCR, reg_val_FLR; + u32 limit = 32; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | RX_ENABLE, msp->registers + MSP_GCR); + + reg_val_FLR = readl(msp->registers + MSP_FLR); + while (!(reg_val_FLR & RX_FIFO_EMPTY) && limit--) { + reg_val_DR = readl(msp->registers + MSP_DR); + reg_val_FLR = readl(msp->registers + MSP_FLR); + } + + writel(reg_val_GCR, msp->registers + MSP_GCR); +} + +static void flush_fifo_tx(struct ux500_msp *msp) +{ + u32 reg_val_TSTDR, reg_val_GCR, reg_val_FLR; + u32 limit = 32; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | TX_ENABLE, msp->registers + MSP_GCR); + writel(MSP_ITCR_ITEN | MSP_ITCR_TESTFIFO, msp->registers + MSP_ITCR); + + reg_val_FLR = readl(msp->registers + MSP_FLR); + while (!(reg_val_FLR & TX_FIFO_EMPTY) && limit--) { + reg_val_TSTDR = readl(msp->registers + MSP_TSTDR); + reg_val_FLR = readl(msp->registers + MSP_FLR); + } + writel(0x0, msp->registers + MSP_ITCR); + writel(reg_val_GCR, msp->registers + MSP_GCR); +} + +int ux500_msp_i2s_open(struct ux500_msp *msp, + struct ux500_msp_config *config) +{ + u32 old_reg, new_reg, mask; + int res; + unsigned int tx_sel, rx_sel, tx_busy, rx_busy; + + if (in_interrupt()) { + dev_err(msp->dev, + "%s: ERROR: Open called in interrupt context!\n", + __func__); + return -1; + } + + tx_sel = (config->direction & MSP_DIR_TX) > 0; + rx_sel = (config->direction & MSP_DIR_RX) > 0; + if (!tx_sel && !rx_sel) { + dev_err(msp->dev, "%s: Error: No direction selected!\n", + __func__); + return -EINVAL; + } + + tx_busy = (msp->dir_busy & MSP_DIR_TX) > 0; + rx_busy = (msp->dir_busy & MSP_DIR_RX) > 0; + if (tx_busy && tx_sel) { + dev_err(msp->dev, "%s: Error: TX is in use!\n", __func__); + return -EBUSY; + } + if (rx_busy && rx_sel) { + dev_err(msp->dev, "%s: Error: RX is in use!\n", __func__); + return -EBUSY; + } + + msp->dir_busy |= (tx_sel ? MSP_DIR_TX : 0) | (rx_sel ? MSP_DIR_RX : 0); + + /* First do the global config register */ + mask = RX_CLK_SEL_MASK | TX_CLK_SEL_MASK | RX_FSYNC_MASK | + TX_FSYNC_MASK | RX_SYNC_SEL_MASK | TX_SYNC_SEL_MASK | + RX_FIFO_ENABLE_MASK | TX_FIFO_ENABLE_MASK | SRG_CLK_SEL_MASK | + LOOPBACK_MASK | TX_EXTRA_DELAY_MASK; + + new_reg = (config->tx_clk_sel | config->rx_clk_sel | + config->rx_fsync_pol | config->tx_fsync_pol | + config->rx_fsync_sel | config->tx_fsync_sel | + config->rx_fifo_config | config->tx_fifo_config | + config->srg_clk_sel | config->loopback_enable | + config->tx_data_enable); + + old_reg = readl(msp->registers + MSP_GCR); + old_reg &= ~mask; + new_reg |= old_reg; + writel(new_reg, msp->registers + MSP_GCR); + + res = enable_msp(msp, config); + if (res < 0) { + dev_err(msp->dev, "%s: ERROR: enable_msp failed (%d)!\n", + __func__, res); + return -EBUSY; + } + if (config->loopback_enable & 0x80) + msp->loopback_enable = 1; + + /* Flush FIFOs */ + flush_fifo_tx(msp); + flush_fifo_rx(msp); + + msp->msp_state = MSP_STATE_CONFIGURED; + return 0; +} + +static void disable_msp_rx(struct ux500_msp *msp) +{ + u32 reg_val_GCR, reg_val_DMACR, reg_val_IMSC; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~RX_ENABLE, msp->registers + MSP_GCR); + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + writel(reg_val_DMACR & ~RX_DMA_ENABLE, msp->registers + MSP_DMACR); + reg_val_IMSC = readl(msp->registers + MSP_IMSC); + writel(reg_val_IMSC & + ~(RX_SERVICE_INT | RX_OVERRUN_ERROR_INT), + msp->registers + MSP_IMSC); + + msp->dir_busy &= ~MSP_DIR_RX; +} + +static void disable_msp_tx(struct ux500_msp *msp) +{ + u32 reg_val_GCR, reg_val_DMACR, reg_val_IMSC; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~TX_ENABLE, msp->registers + MSP_GCR); + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + writel(reg_val_DMACR & ~TX_DMA_ENABLE, msp->registers + MSP_DMACR); + reg_val_IMSC = readl(msp->registers + MSP_IMSC); + writel(reg_val_IMSC & + ~(TX_SERVICE_INT | TX_UNDERRUN_ERR_INT), + msp->registers + MSP_IMSC); + + msp->dir_busy &= ~MSP_DIR_TX; +} + +static int disable_msp(struct ux500_msp *msp, unsigned int dir) +{ + u32 reg_val_GCR; + int status = 0; + unsigned int disable_tx, disable_rx; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + disable_tx = dir & MSP_DIR_TX; + disable_rx = dir & MSP_DIR_TX; + if (disable_tx && disable_rx) { + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | LOOPBACK_MASK, + msp->registers + MSP_GCR); + + /* Flush TX-FIFO */ + flush_fifo_tx(msp); + + /* Disable TX-channel */ + writel((readl(msp->registers + MSP_GCR) & + (~TX_ENABLE)), msp->registers + MSP_GCR); + + /* Flush RX-FIFO */ + flush_fifo_rx(msp); + + /* Disable Loopback and Receive channel */ + writel((readl(msp->registers + MSP_GCR) & + (~(RX_ENABLE | LOOPBACK_MASK))), + msp->registers + MSP_GCR); + + disable_msp_tx(msp); + disable_msp_rx(msp); + } else if (disable_tx) + disable_msp_tx(msp); + else if (disable_rx) + disable_msp_rx(msp); + + return status; +} + +int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) +{ + u32 reg_val_GCR, enable_bit; + + if (msp->msp_state == MSP_STATE_IDLE) { + dev_err(msp->dev, "%s: ERROR: MSP is not configured!\n", + __func__); + return -EINVAL; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + enable_bit = TX_ENABLE; + else + enable_bit = RX_ENABLE; + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | enable_bit, msp->registers + MSP_GCR); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + disable_msp_tx(msp); + else + disable_msp_rx(msp); + break; + default: + return -EINVAL; + break; + } + + return 0; +} + +int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) +{ + int status = 0; + + dev_dbg(msp->dev, "%s: Enter (dir = 0x%01x).\n", __func__, dir); + + status = disable_msp(msp, dir); + if (msp->dir_busy == 0) { + /* disable sample rate and frame generators */ + msp->msp_state = MSP_STATE_IDLE; + writel((readl(msp->registers + MSP_GCR) & + (~(FRAME_GEN_ENABLE | SRG_ENABLE))), + msp->registers + MSP_GCR); + if (msp->plat_exit) + status = msp->plat_exit(); + if (status) + dev_warn(msp->dev, + "%s: WARN: ux500_msp_i2s_exit failed (%d)!\n", + __func__, status); + writel(0, msp->registers + MSP_GCR); + writel(0, msp->registers + MSP_TCF); + writel(0, msp->registers + MSP_RCF); + writel(0, msp->registers + MSP_DMACR); + writel(0, msp->registers + MSP_SRG); + writel(0, msp->registers + MSP_MCR); + writel(0, msp->registers + MSP_RCM); + writel(0, msp->registers + MSP_RCV); + writel(0, msp->registers + MSP_TCE0); + writel(0, msp->registers + MSP_TCE1); + writel(0, msp->registers + MSP_TCE2); + writel(0, msp->registers + MSP_TCE3); + writel(0, msp->registers + MSP_RCE0); + writel(0, msp->registers + MSP_RCE1); + writel(0, msp->registers + MSP_RCE2); + writel(0, msp->registers + MSP_RCE3); + } + + return status; + +} + +int ux500_msp_i2s_init_msp(struct platform_device *pdev, + struct ux500_msp **msp_p, + struct msp_i2s_platform_data *platform_data) +{ + int ret = 0; + struct resource *res = NULL; + struct i2s_controller *i2s_cont; + struct ux500_msp *msp; + + dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, + pdev->name, platform_data->id); + + *msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL); + msp = *msp_p; + + msp->id = platform_data->id; + msp->dev = &pdev->dev; + msp->plat_init = platform_data->msp_i2s_init; + msp->plat_exit = platform_data->msp_i2s_exit; + msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx; + msp->dma_cfg_tx = platform_data->msp_i2s_dma_tx; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "%s: ERROR: Unable to get resource!\n", + __func__); + ret = -ENOMEM; + goto err_res; + } + + msp->registers = ioremap(res->start, (res->end - res->start + 1)); + if (msp->registers == NULL) { + dev_err(&pdev->dev, "%s: ERROR: ioremap failed!\n", __func__); + ret = -ENOMEM; + goto err_res; + } + + msp->msp_state = MSP_STATE_IDLE; + msp->loopback_enable = 0; + + /* I2S-controller is allocated and added in I2S controller class. */ + i2s_cont = devm_kzalloc(&pdev->dev, sizeof(*i2s_cont), GFP_KERNEL); + if (!i2s_cont) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to allocate I2S-controller!\n", + __func__); + goto err_i2s_cont; + } + i2s_cont->dev.parent = &pdev->dev; + i2s_cont->data = (void *)msp; + i2s_cont->id = (s16)msp->id; + snprintf(i2s_cont->name, sizeof(i2s_cont->name), "ux500-msp-i2s.%04x", + msp->id); + dev_dbg(&pdev->dev, "I2S device-name: '%s'\n", i2s_cont->name); + msp->i2s_cont = i2s_cont; + + return 0; + +err_i2s_cont: + iounmap(msp->registers); + +err_res: + devm_kfree(&pdev->dev, msp); + + return ret; +} + +void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, + struct ux500_msp *msp) +{ + dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id); + + device_unregister(&msp->i2s_cont->dev); + devm_kfree(&pdev->dev, msp->i2s_cont); + + iounmap(msp->registers); + + devm_kfree(&pdev->dev, msp); +} + +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h new file mode 100644 index 0000000..7f71b4a --- /dev/null +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -0,0 +1,553 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * for ST-Ericsson. + * + * License terms: + * + * 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. + */ + + +#ifndef UX500_MSP_I2S_H +#define UX500_MSP_I2S_H + +#include <linux/platform_device.h> + +#include <mach/board-mop500-msp.h> + +#define MSP_INPUT_FREQ_APB 48000000 + +/*** Stereo mode. Used for APB data accesses as 16 bits accesses (mono), + * 32 bits accesses (stereo). + ***/ +enum msp_stereo_mode { + MSP_MONO, + MSP_STEREO +}; + +/* Direction (Transmit/Receive mode) */ +enum msp_direction { + MSP_TX = 1, + MSP_RX = 2 +}; + +/* Transmit and receive configuration register */ +#define MSP_BIG_ENDIAN 0x00000000 +#define MSP_LITTLE_ENDIAN 0x00001000 +#define MSP_UNEXPECTED_FS_ABORT 0x00000000 +#define MSP_UNEXPECTED_FS_IGNORE 0x00008000 +#define MSP_NON_MODE_BIT_MASK 0x00009000 + +/* Global configuration register */ +#define RX_ENABLE 0x00000001 +#define RX_FIFO_ENABLE 0x00000002 +#define RX_SYNC_SRG 0x00000010 +#define RX_CLK_POL_RISING 0x00000020 +#define RX_CLK_SEL_SRG 0x00000040 +#define TX_ENABLE 0x00000100 +#define TX_FIFO_ENABLE 0x00000200 +#define TX_SYNC_SRG_PROG 0x00001800 +#define TX_SYNC_SRG_AUTO 0x00001000 +#define TX_CLK_POL_RISING 0x00002000 +#define TX_CLK_SEL_SRG 0x00004000 +#define TX_EXTRA_DELAY_ENABLE 0x00008000 +#define SRG_ENABLE 0x00010000 +#define FRAME_GEN_ENABLE 0x00100000 +#define SRG_CLK_SEL_APB 0x00000000 +#define RX_FIFO_SYNC_HI 0x00000000 +#define TX_FIFO_SYNC_HI 0x00000000 +#define SPI_CLK_MODE_NORMAL 0x00000000 + +#define MSP_FRAME_SIZE_AUTO -1 + +#define MSP_DR 0x00 +#define MSP_GCR 0x04 +#define MSP_TCF 0x08 +#define MSP_RCF 0x0c +#define MSP_SRG 0x10 +#define MSP_FLR 0x14 +#define MSP_DMACR 0x18 + +#define MSP_IMSC 0x20 +#define MSP_RIS 0x24 +#define MSP_MIS 0x28 +#define MSP_ICR 0x2c +#define MSP_MCR 0x30 +#define MSP_RCV 0x34 +#define MSP_RCM 0x38 + +#define MSP_TCE0 0x40 +#define MSP_TCE1 0x44 +#define MSP_TCE2 0x48 +#define MSP_TCE3 0x4c + +#define MSP_RCE0 0x60 +#define MSP_RCE1 0x64 +#define MSP_RCE2 0x68 +#define MSP_RCE3 0x6c +#define MSP_IODLY 0x70 + +#define MSP_ITCR 0x80 +#define MSP_ITIP 0x84 +#define MSP_ITOP 0x88 +#define MSP_TSTDR 0x8c + +#define MSP_PID0 0xfe0 +#define MSP_PID1 0xfe4 +#define MSP_PID2 0xfe8 +#define MSP_PID3 0xfec + +#define MSP_CID0 0xff0 +#define MSP_CID1 0xff4 +#define MSP_CID2 0xff8 +#define MSP_CID3 0xffc + +/* Protocol dependant parameters list */ +#define RX_ENABLE_MASK BIT(0) +#define RX_FIFO_ENABLE_MASK BIT(1) +#define RX_FSYNC_MASK BIT(2) +#define DIRECT_COMPANDING_MASK BIT(3) +#define RX_SYNC_SEL_MASK BIT(4) +#define RX_CLK_POL_MASK BIT(5) +#define RX_CLK_SEL_MASK BIT(6) +#define LOOPBACK_MASK BIT(7) +#define TX_ENABLE_MASK BIT(8) +#define TX_FIFO_ENABLE_MASK BIT(9) +#define TX_FSYNC_MASK BIT(10) +#define TX_MSP_TDR_TSR BIT(11) +#define TX_SYNC_SEL_MASK (BIT(12) | BIT(11)) +#define TX_CLK_POL_MASK BIT(13) +#define TX_CLK_SEL_MASK BIT(14) +#define TX_EXTRA_DELAY_MASK BIT(15) +#define SRG_ENABLE_MASK BIT(16) +#define SRG_CLK_POL_MASK BIT(17) +#define SRG_CLK_SEL_MASK (BIT(19) | BIT(18)) +#define FRAME_GEN_EN_MASK BIT(20) +#define SPI_CLK_MODE_MASK (BIT(22) | BIT(21)) +#define SPI_BURST_MODE_MASK BIT(23) + +#define RXEN_SHIFT 0 +#define RFFEN_SHIFT 1 +#define RFSPOL_SHIFT 2 +#define DCM_SHIFT 3 +#define RFSSEL_SHIFT 4 +#define RCKPOL_SHIFT 5 +#define RCKSEL_SHIFT 6 +#define LBM_SHIFT 7 +#define TXEN_SHIFT 8 +#define TFFEN_SHIFT 9 +#define TFSPOL_SHIFT 10 +#define TFSSEL_SHIFT 11 +#define TCKPOL_SHIFT 13 +#define TCKSEL_SHIFT 14 +#define TXDDL_SHIFT 15 +#define SGEN_SHIFT 16 +#define SCKPOL_SHIFT 17 +#define SCKSEL_SHIFT 18 +#define FGEN_SHIFT 20 +#define SPICKM_SHIFT 21 +#define TBSWAP_SHIFT 28 + +#define RCKPOL_MASK BIT(0) +#define TCKPOL_MASK BIT(0) +#define SPICKM_MASK (BIT(1) | BIT(0)) +#define MSP_RX_CLKPOL_BIT(n) ((n & RCKPOL_MASK) << RCKPOL_SHIFT) +#define MSP_TX_CLKPOL_BIT(n) ((n & TCKPOL_MASK) << TCKPOL_SHIFT) + +#define P1ELEN_SHIFT 0 +#define P1FLEN_SHIFT 3 +#define DTYP_SHIFT 10 +#define ENDN_SHIFT 12 +#define DDLY_SHIFT 13 +#define FSIG_SHIFT 15 +#define P2ELEN_SHIFT 16 +#define P2FLEN_SHIFT 19 +#define P2SM_SHIFT 26 +#define P2EN_SHIFT 27 +#define FSYNC_SHIFT 15 + +#define P1ELEN_MASK 0x00000007 +#define P2ELEN_MASK 0x00070000 +#define P1FLEN_MASK 0x00000378 +#define P2FLEN_MASK 0x03780000 +#define DDLY_MASK 0x00003000 +#define DTYP_MASK 0x00000600 +#define P2SM_MASK 0x04000000 +#define P2EN_MASK 0x08000000 +#define ENDN_MASK 0x00001000 +#define TFSPOL_MASK 0x00000400 +#define TBSWAP_MASK 0x30000000 +#define COMPANDING_MODE_MASK 0x00000c00 +#define FSYNC_MASK 0x00008000 + +#define MSP_P1_ELEM_LEN_BITS(n) (n & P1ELEN_MASK) +#define MSP_P2_ELEM_LEN_BITS(n) (((n) << P2ELEN_SHIFT) & P2ELEN_MASK) +#define MSP_P1_FRAME_LEN_BITS(n) (((n) << P1FLEN_SHIFT) & P1FLEN_MASK) +#define MSP_P2_FRAME_LEN_BITS(n) (((n) << P2FLEN_SHIFT) & P2FLEN_MASK) +#define MSP_DATA_DELAY_BITS(n) (((n) << DDLY_SHIFT) & DDLY_MASK) +#define MSP_DATA_TYPE_BITS(n) (((n) << DTYP_SHIFT) & DTYP_MASK) +#define MSP_P2_START_MODE_BIT(n) ((n << P2SM_SHIFT) & P2SM_MASK) +#define MSP_P2_ENABLE_BIT(n) ((n << P2EN_SHIFT) & P2EN_MASK) +#define MSP_SET_ENDIANNES_BIT(n) ((n << ENDN_SHIFT) & ENDN_MASK) +#define MSP_FSYNC_POL(n) ((n << TFSPOL_SHIFT) & TFSPOL_MASK) +#define MSP_DATA_WORD_SWAP(n) ((n << TBSWAP_SHIFT) & TBSWAP_MASK) +#define MSP_SET_COMPANDING_MODE(n) ((n << DTYP_SHIFT) & \ + COMPANDING_MODE_MASK) +#define MSP_SET_FSYNC_IGNORE(n) ((n << FSYNC_SHIFT) & FSYNC_MASK) + +/* Flag register */ +#define RX_BUSY BIT(0) +#define RX_FIFO_EMPTY BIT(1) +#define RX_FIFO_FULL BIT(2) +#define TX_BUSY BIT(3) +#define TX_FIFO_EMPTY BIT(4) +#define TX_FIFO_FULL BIT(5) + +#define RBUSY_SHIFT 0 +#define RFE_SHIFT 1 +#define RFU_SHIFT 2 +#define TBUSY_SHIFT 3 +#define TFE_SHIFT 4 +#define TFU_SHIFT 5 + +/* Multichannel control register */ +#define RMCEN_SHIFT 0 +#define RMCSF_SHIFT 1 +#define RCMPM_SHIFT 3 +#define TMCEN_SHIFT 5 +#define TNCSF_SHIFT 6 + +/* Sample rate generator register */ +#define SCKDIV_SHIFT 0 +#define FRWID_SHIFT 10 +#define FRPER_SHIFT 16 + +#define SCK_DIV_MASK 0x0000003FF +#define FRAME_WIDTH_BITS(n) (((n) << FRWID_SHIFT) & 0x0000FC00) +#define FRAME_PERIOD_BITS(n) (((n) << FRPER_SHIFT) & 0x1FFF0000) + +/* DMA controller register */ +#define RX_DMA_ENABLE BIT(0) +#define TX_DMA_ENABLE BIT(1) + +#define RDMAE_SHIFT 0 +#define TDMAE_SHIFT 1 + +/* Interrupt Register */ +#define RX_SERVICE_INT BIT(0) +#define RX_OVERRUN_ERROR_INT BIT(1) +#define RX_FSYNC_ERR_INT BIT(2) +#define RX_FSYNC_INT BIT(3) +#define TX_SERVICE_INT BIT(4) +#define TX_UNDERRUN_ERR_INT BIT(5) +#define TX_FSYNC_ERR_INT BIT(6) +#define TX_FSYNC_INT BIT(7) +#define ALL_INT 0x000000ff + +/* MSP test control register */ +#define MSP_ITCR_ITEN BIT(0) +#define MSP_ITCR_TESTFIFO BIT(1) + +#define RMCEN_BIT 0 +#define RMCSF_BIT 1 +#define RCMPM_BIT 3 +#define TMCEN_BIT 5 +#define TNCSF_BIT 6 + +/* Single or dual phase mode */ +enum msp_phase_mode { + MSP_SINGLE_PHASE, + MSP_DUAL_PHASE +}; + +/* Frame length */ +enum msp_frame_length { + MSP_FRAME_LEN_1 = 0, + MSP_FRAME_LEN_2 = 1, + MSP_FRAME_LEN_4 = 3, + MSP_FRAME_LEN_8 = 7, + MSP_FRAME_LEN_12 = 11, + MSP_FRAME_LEN_16 = 15, + MSP_FRAME_LEN_20 = 19, + MSP_FRAME_LEN_32 = 31, + MSP_FRAME_LEN_48 = 47, + MSP_FRAME_LEN_64 = 63 +}; + +/* Element length */ +enum msp_elem_length { + MSP_ELEM_LEN_8 = 0, + MSP_ELEM_LEN_10 = 1, + MSP_ELEM_LEN_12 = 2, + MSP_ELEM_LEN_14 = 3, + MSP_ELEM_LEN_16 = 4, + MSP_ELEM_LEN_20 = 5, + MSP_ELEM_LEN_24 = 6, + MSP_ELEM_LEN_32 = 7 +}; + +enum msp_data_xfer_width { + MSP_DATA_TRANSFER_WIDTH_BYTE, + MSP_DATA_TRANSFER_WIDTH_HALFWORD, + MSP_DATA_TRANSFER_WIDTH_WORD +}; + +enum msp_frame_sync { + MSP_FSYNC_UNIGNORE = 0, + MSP_FSYNC_IGNORE = 1, +}; + +enum msp_phase2_start_mode { + MSP_PHASE2_START_MODE_IMEDIATE, + MSP_PHASE2_START_MODE_FSYNC +}; + +enum msp_btf { + MSP_BTF_MS_BIT_FIRST = 0, + MSP_BTF_LS_BIT_FIRST = 1 +}; + +enum msp_fsync_pol { + MSP_FSYNC_POL_ACT_HI = 0, + MSP_FSYNC_POL_ACT_LO = 1 +}; + +/* Data delay (in bit clock cycles) */ +enum msp_delay { + MSP_DELAY_0 = 0, + MSP_DELAY_1 = 1, + MSP_DELAY_2 = 2, + MSP_DELAY_3 = 3 +}; + +/* Configurations of clocks (transmit, receive or sample rate generator) */ +enum msp_edge { + MSP_FALLING_EDGE = 0, + MSP_RISING_EDGE = 1, +}; + +enum msp_hws { + MSP_SWAP_NONE = 0, + MSP_SWAP_BYTE_PER_WORD = 1, + MSP_SWAP_BYTE_PER_HALF_WORD = 2, + MSP_SWAP_HALF_WORD_PER_WORD = 3 +}; + +enum msp_compress_mode { + MSP_COMPRESS_MODE_LINEAR = 0, + MSP_COMPRESS_MODE_MU_LAW = 2, + MSP_COMPRESS_MODE_A_LAW = 3 +}; + +enum msp_spi_burst_mode { + MSP_SPI_BURST_MODE_DISABLE = 0, + MSP_SPI_BURST_MODE_ENABLE = 1 +}; + +enum msp_expand_mode { + MSP_EXPAND_MODE_LINEAR = 0, + MSP_EXPAND_MODE_LINEAR_SIGNED = 1, + MSP_EXPAND_MODE_MU_LAW = 2, + MSP_EXPAND_MODE_A_LAW = 3 +}; + +#define MSP_FRAME_PERIOD_IN_MONO_MODE 256 +#define MSP_FRAME_PERIOD_IN_STEREO_MODE 32 +#define MSP_FRAME_WIDTH_IN_STEREO_MODE 16 + +enum msp_protocol { + MSP_I2S_PROTOCOL, + MSP_PCM_PROTOCOL, + MSP_PCM_COMPAND_PROTOCOL, + MSP_INVALID_PROTOCOL +}; + +/* + * No of registers to backup during + * suspend resume + */ +#define MAX_MSP_BACKUP_REGS 36 + +enum enum_i2s_controller { + MSP_0_I2S_CONTROLLER = 0, + MSP_1_I2S_CONTROLLER, + MSP_2_I2S_CONTROLLER, + MSP_3_I2S_CONTROLLER, +}; + +enum i2s_direction_t { + MSP_DIR_TX = 0x01, + MSP_DIR_RX = 0x02, +}; + +enum msp_data_size { + MSP_DATA_BITS_DEFAULT = -1, + MSP_DATA_BITS_8 = 0x00, + MSP_DATA_BITS_10, + MSP_DATA_BITS_12, + MSP_DATA_BITS_14, + MSP_DATA_BITS_16, + MSP_DATA_BITS_20, + MSP_DATA_BITS_24, + MSP_DATA_BITS_32, +}; + +enum msp_state { + MSP_STATE_IDLE = 0, + MSP_STATE_CONFIGURED = 1, + MSP_STATE_RUNNING = 2, +}; + +enum msp_rx_comparison_enable_mode { + MSP_COMPARISON_DISABLED = 0, + MSP_COMPARISON_NONEQUAL_ENABLED = 2, + MSP_COMPARISON_EQUAL_ENABLED = 3 +}; + +struct msp_multichannel_config { + bool rx_multichannel_enable; + bool tx_multichannel_enable; + enum msp_rx_comparison_enable_mode rx_comparison_enable_mode; + u8 padding; + u32 comparison_value; + u32 comparison_mask; + u32 rx_channel_0_enable; + u32 rx_channel_1_enable; + u32 rx_channel_2_enable; + u32 rx_channel_3_enable; + u32 tx_channel_0_enable; + u32 tx_channel_1_enable; + u32 tx_channel_2_enable; + u32 tx_channel_3_enable; +}; + +struct msp_protdesc { + u32 rx_phase_mode; + u32 tx_phase_mode; + u32 rx_phase2_start_mode; + u32 tx_phase2_start_mode; + u32 rx_byte_order; + u32 tx_byte_order; + u32 rx_frame_len_1; + u32 rx_frame_len_2; + u32 tx_frame_len_1; + u32 tx_frame_len_2; + u32 rx_elem_len_1; + u32 rx_elem_len_2; + u32 tx_elem_len_1; + u32 tx_elem_len_2; + u32 rx_data_delay; + u32 tx_data_delay; + u32 rx_clk_pol; + u32 tx_clk_pol; + u32 rx_fsync_pol; + u32 tx_fsync_pol; + u32 rx_half_word_swap; + u32 tx_half_word_swap; + u32 compression_mode; + u32 expansion_mode; + u32 frame_sync_ignore; + u32 frame_period; + u32 frame_width; + u32 clocks_per_frame; +}; + +struct i2s_message { + enum i2s_direction_t i2s_direction; + void *txdata; + void *rxdata; + size_t txbytes; + size_t rxbytes; + int dma_flag; + int tx_offset; + int rx_offset; + bool cyclic_dma; + dma_addr_t buf_addr; + size_t buf_len; + size_t period_len; +}; + +struct i2s_controller { + struct module *owner; + unsigned int id; + unsigned int class; + const struct i2s_algorithm *algo; /* the algorithm to access the bus */ + void *data; + struct mutex bus_lock; + struct device dev; /* the controller device */ + char name[48]; +}; + +struct ux500_msp_config { + unsigned int f_inputclk; + unsigned int rx_clk_sel; + unsigned int tx_clk_sel; + unsigned int srg_clk_sel; + unsigned int rx_fsync_pol; + unsigned int tx_fsync_pol; + unsigned int rx_fsync_sel; + unsigned int tx_fsync_sel; + unsigned int rx_fifo_config; + unsigned int tx_fifo_config; + unsigned int spi_clk_mode; + unsigned int spi_burst_mode; + unsigned int loopback_enable; + unsigned int tx_data_enable; + unsigned int default_protdesc; + struct msp_protdesc protdesc; + int multichannel_configured; + struct msp_multichannel_config multichannel_config; + unsigned int direction; + unsigned int protocol; + unsigned int frame_freq; + unsigned int frame_size; + enum msp_data_size data_size; + unsigned int def_elem_len; + unsigned int iodelay; + void (*handler) (void *data); + void *tx_callback_data; + void *rx_callback_data; +}; + +struct ux500_msp { + enum enum_i2s_controller id; + void __iomem *registers; + struct device *dev; + struct i2s_controller *i2s_cont; + struct stedma40_chan_cfg *dma_cfg_rx; + struct stedma40_chan_cfg *dma_cfg_tx; + struct dma_chan *tx_pipeid; + struct dma_chan *rx_pipeid; + enum msp_state msp_state; + int (*transfer) (struct ux500_msp *msp, struct i2s_message *message); + int (*plat_init) (void); + int (*plat_exit) (void); + struct timer_list notify_timer; + int def_elem_len; + unsigned int dir_busy; + int loopback_enable; + u32 backup_regs[MAX_MSP_BACKUP_REGS]; + unsigned int f_bitclk; +}; + +struct ux500_msp_dma_params { + unsigned int data_size; + struct stedma40_chan_cfg *dma_cfg; +}; + +int ux500_msp_i2s_init_msp(struct platform_device *pdev, + struct ux500_msp **msp_p, + struct msp_i2s_platform_data *platform_data); +void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, + struct ux500_msp *msp); +int ux500_msp_i2s_open(struct ux500_msp *msp, struct ux500_msp_config *config); +int ux500_msp_i2s_close(struct ux500_msp *msp, + unsigned int dir); +int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, + int direction); + +#endif -- 1.7.8.3

2 1

[alsa-devel] [PATCH] ASoC: sh: fix migor.c compilation
by Guennadi Liakhovetski 09 May '12

09 May '12

Fix a recent compilation breakage, caused by a change in SH clock API. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- sound/soc/sh/migor.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 9d9ad8d..8526e1e 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -35,7 +35,7 @@ static unsigned long siumckb_recalc(struct clk *clk) return codec_freq; } -static struct clk_ops siumckb_clk_ops = { +static struct sh_clk_ops siumckb_clk_ops = { .recalc = siumckb_recalc, }; -- 1.7.2.5

2 3

[alsa-devel] [PATCH] ALSA: HDA: Lessen CPU usage when waiting for chip to respond
by David Henningsson 09 May '12

09 May '12

When an IRQ for some reason gets lost, we wait up to a second using udelay, which is CPU intensive. This patch improves the situation by waiting about 30 ms in the CPU intensive mode, then stepping down to using msleep(2) instead. In essence, we trade some granularity in exchange for less CPU consumption when the waiting time is a bit longer. As a result, PulseAudio should no longer be killed by the kernel for taking up to much RT-prio CPU time. At least not for *this* reason. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) Hi Arun, Can you check if this patch resolves your problem with PulseAudio getting killed by the kernel? If so, we should apply it to the kernel, perhaps even to stable. // David diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7b6a823..0e7c3f1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -783,11 +783,13 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, { struct azx *chip = bus->private_data; unsigned long timeout; + unsigned long loopcounter; int do_poll = 0; again: timeout = jiffies + msecs_to_jiffies(1000); - for (;;) { + + for (loopcounter = 0;; loopcounter++) { if (chip->polling_mode || do_poll) { spin_lock_irq(&chip->reg_lock); azx_update_rirb(chip); @@ -803,7 +805,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } if (time_after(jiffies, timeout)) break; - if (bus->needs_damn_long_delay) + if (bus->needs_damn_long_delay || loopcounter > 3000) msleep(2); /* temporary workaround */ else { udelay(10); -- 1.7.9.5

3 4

[alsa-devel] [PATCH 03/11] ASoC: core: Add widget SND_SOC_DAPM_CLOCK_SUPPLY
by Ola Lilja 09 May '12

09 May '12

Adds a supply-widget variant for connection to the clock-framework. This widget-type corresponds to the variant for regulators. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- include/sound/soc-dapm.h | 9 +++++++ sound/soc/soc-dapm.c | 59 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 68 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 1649ef2..0e7ab1e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -248,6 +248,10 @@ struct device; { .id = snd_soc_dapm_regulator_supply, .name = wname, \ .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } +#define SND_SOC_DAPM_CLOCK_SUPPLY(wname, is_sys) \ +{ .id = snd_soc_dapm_clock_supply, .name = wname, \ + .reg = SND_SOC_NOPM, .shift = is_sys , .event = dapm_clock_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } /* dapm kcontrol types */ #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ @@ -329,6 +333,8 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -401,6 +407,8 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); int snd_soc_dapm_get_regulator_status(struct snd_soc_dapm_context *dapm, const char *pin); +int snd_soc_dapm_get_power_status(struct snd_soc_dapm_context *dapm, + const char *pin); /* Mostly internal - should not normally be used */ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); @@ -429,6 +437,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ snd_soc_dapm_regulator_supply, /* external regulator */ + snd_soc_dapm_clock_supply, /* external clock */ snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b6c9df5..8649c1f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -35,6 +35,7 @@ #include <linux/debugfs.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> +#include <linux/clk.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -51,6 +52,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, + [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_dai] = 3, [snd_soc_dapm_aif_in] = 3, @@ -88,6 +90,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, [snd_soc_dapm_dai] = 10, + [snd_soc_dapm_clock_supply] = 11, [snd_soc_dapm_regulator_supply] = 11, [snd_soc_dapm_supply] = 11, [snd_soc_dapm_post] = 12, @@ -365,6 +368,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_vmid: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: @@ -697,6 +701,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -768,6 +773,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -857,6 +863,21 @@ EXPORT_SYMBOL_GPL(dapm_reg_event); /* * Handler for regulator supply widget. */ +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + return clk_enable(w->priv); + else { + clk_disable(w->priv); + return 0; + } +} +EXPORT_SYMBOL_GPL(dapm_clock_event); + +/* + * Handler for clock supply widget. + */ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1355,6 +1376,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, switch (w->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -1457,6 +1479,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_micbias: if (d->target_bias_level < SND_SOC_BIAS_STANDBY) d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1809,6 +1832,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -2050,6 +2074,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_post: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: @@ -2723,6 +2748,16 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, return NULL; } break; + case snd_soc_dapm_clock_supply: + w->priv = (w->shift) ? clk_get_sys(w->name, NULL) : + clk_get(dapm->dev, w->name); + if (IS_ERR(w->priv)) { + ret = PTR_ERR(w->priv); + dev_err(dapm->dev, "Failed to request %s: %d\n", + w->name, ret); + return NULL; + } + break; default: break; } @@ -2773,6 +2808,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; case snd_soc_dapm_dai: @@ -3114,6 +3150,29 @@ int snd_soc_dapm_get_regulator_status(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_regulator_status); + /** + * snd_soc_dapm_get_power_status - get widget power status + * @dapm: DAPM context + * @pin: Widget name + * + * Get widget power status - Enabled or disabled. + * + * Returns -1 for failure. + * Returns 0 if disabled. + * Returns 1 if enabled. + */ +int snd_soc_dapm_get_power_status(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + + if (!w) + return -1; + + return w->power; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_power_status); + /** * snd_soc_dapm_ignore_suspend - ignore suspend status for DAPM endpoint * @dapm: DAPM context -- 1.7.8.3

2 3

[alsa-devel] [PATCH 0/3] ARM: OMAP: Add platform devices for ASoC HDMI drivers
by Ricardo Neri 09 May '12

09 May '12

This set of patches is intended to add the platform devices for the ASoC HDMI drivers when not using device tree. For this, I took an approach similar to DMIC and McPDM by creating a dedicated functions to create the devices. I included the device for the CPU DAI driver, omap-hdmi-audio-dai, and the device for the machine driver, omap-hdmi-audio, in devices.c to reflect the fact that any OMAP4 (and OMAP5 in the future) has HDMI IP. I put the device for the codec in the board file to reflect the fact that not necessarily all the boards with OMAP4/5 have the HDMI output wired. Best regards Ricardo Ricardo Neri (3): ARM: OMAP: devices: Register platform devices for HDMI audio ARM: OMAP4: board-4430sdp: Register platform device for HDMI audio codec ARM: OMAP4: board-omap4panda: Register platform device for HDMI audio codec arch/arm/mach-omap2/board-4430sdp.c | 6 ++++++ arch/arm/mach-omap2/board-omap4panda.c | 6 ++++++ arch/arm/mach-omap2/devices.c | 31 +++++++++++++++++++++++++++++++ 3 files changed, 43 insertions(+), 0 deletions(-) -- 1.7.5.4

2 6

[alsa-devel] [PATCH] alsa-lib conf: Add "Plantronics USB Headset" to S/PDIF blacklist
by David Henningsson 09 May '12

09 May '12

Unsurprisingly, Plantronics does not have S/PDIF connections on their headsets. 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 0a0e374..db18dad 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -43,6 +43,7 @@ USB-Audio.pcm.iec958_device { "Logitech USB Headset" 999 "Logitech Wireless Headset" 999 "Sennheiser USB headset" 999 + "Plantronics USB Headset" 999 "USB Device 0x46d:0x992" 999 } -- 1.7.9.5

1 1

[PATCH] alsa-lib conf: Add "Plantronics USB Headset" to S/PDIF blacklist
by David Henningsson 09 May '12

09 May '12

Unsurprisingly, Plantronics does not have S/PDIF connections on their headsets. 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 0a0e374..732e8f0 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -42,6 +42,7 @@ USB-Audio.pcm.iec958_device { "Logitech Speaker Lapdesk N700" 999 "Logitech USB Headset" 999 "Logitech Wireless Headset" 999 + "Plantronics USB Headset" 999 "Sennheiser USB headset" 999 "USB Device 0x46d:0x992" 999 } -- 1.7.9.5 --------------010900050302030302060002--

1 0

[alsa-devel] [PATCH 1/3] ASoC: alc5623: Convert to module_i2c_driver()
by Axel Lin 08 May '12

08 May '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 20 +------------------- 1 files changed, 1 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 3061d35..1960478 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -1083,25 +1083,7 @@ static struct i2c_driver alc5623_i2c_driver = { .id_table = alc5623_i2c_table, }; -static int __init alc5623_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&alc5623_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "%s: can't add i2c driver", __func__); - return ret; - } - - return ret; -} -module_init(alc5623_modinit); - -static void __exit alc5623_modexit(void) -{ - i2c_del_driver(&alc5623_i2c_driver); -} -module_exit(alc5623_modexit); +module_i2c_driver(alc5623_i2c_driver); MODULE_DESCRIPTION("ASoC alc5621/2/3 driver"); MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); -- 1.7.5.4

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[alsa-devel] Fast Track Ultra latency and new streaming logic.
by Aurélien Leblond 08 May '12

08 May '12

> I can't seem to get below 512frames in flight (2 256sample frames or 4 128 > or 8 64's) without consistant overruns. On my side, I can run 256 frames ; 3 periods, no xruns at all at 48kHz... I haven't tried to push the driver more than that, Aurélien

1 0

[alsa-devel] [PATCH 04/11] ASoC: core: Add signed multi register control
by Ola Lilja 08 May '12

08 May '12

From: Kristoffer KARLSSON <kristoffer.karlsson(a)stericsson.com> Added control type that can span multiple consecutive codec registers forming a single signed value in a MSB/LSB manner. The control dynamically adjusts to the register word size configured in driver. Added convenience macro. SOC_SINGLE_XR_SX Added accessor implementations. snd_soc_info_xr_sx snd_soc_get_xr_sx snd_soc_put_xr_sx Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson(a)stericsson.com> --- include/sound/soc.h | 22 +++++++++- sound/soc/soc-core.c | 118 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 139 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2ebf787..8888adc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -200,6 +200,15 @@ {.base = xbase, .num_regs = xregs, \ .mask = xmask }) } +#define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \ + xmin, xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = snd_soc_info_xr_sx, .get = snd_soc_get_xr_sx, \ + .put = snd_soc_put_xr_sx, \ + .private_value = (unsigned long)&(struct soc_mreg_control) \ + {.regbase = xregbase, .regcount = xregcount, .nbits = xnbits, \ + .invert = xinvert, .min = xmin, .max = xmax} } + /* * Simplified versions of above macros, declaring a struct and calculating * ARRAY_SIZE internally @@ -433,7 +442,12 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); - +int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /** * struct snd_soc_reg_access - Describes whether a given register is @@ -916,6 +930,12 @@ struct soc_bytes { u32 mask; }; +/* multi register control */ +struct soc_mreg_control { + long min, max; + unsigned int regbase, regcount, nbits, invert; +}; + /* enumerated kcontrol */ struct soc_enum { unsigned short reg; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e19c24a..7fc7294b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2848,6 +2848,124 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_bytes_put); /** + * snd_soc_info_xr_sx - signed multi register info callback + * @kcontrol: mreg control + * @uinfo: control element information + * + * Callback to provide information of a control that can + * span multiple codec registers which together + * forms a single signed value in a MSB/LSB manner. + * + * Returns 0 for success. + */ +int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = mc->min; + uinfo->value.integer.max = mc->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_xr_sx); + +/** + * snd_soc_get_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to get the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = codec->driver->reg_word_size * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long min = mc->min; + long max = mc->max; + long val = 0; + unsigned long regval; + unsigned int i; + + for (i = 0; i < regcount; i++) { + regval = snd_soc_read(codec, regbase+i) & regwmask; + val |= regval << (regwshift*(regcount-i-1)); + } + val &= mask; + if (min < 0 && val > max) + val |= ~mask; + if (invert) + val = max - val; + ucontrol->value.integer.value[0] = val; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_xr_sx); + +/** + * snd_soc_put_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to set the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = codec->driver->reg_word_size * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long min = mc->min; + long max = mc->max; + long val = ucontrol->value.integer.value[0]; + unsigned int i, regval, regmask; + int err; + + if (invert) + val = max - val; + val &= mask; + for (i = 0; i < regcount; i++) { + regval = (val >> (regwshift*(regcount-i-1))) & regwmask; + regmask = (mask >> (regwshift*(regcount-i-1))) & regwmask; + err = snd_soc_update_bits_locked(codec, regbase+i, + regmask, regval); + if (err < 0) + return err; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); + +/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID -- 1.7.8.3

2 1

[alsa-devel] [PATCH 02/11] ASoC: core: Add method snd_soc_dapm_get_regulator_status
by Ola Lilja 08 May '12

08 May '12

Add method to get the status of a regulator supply widget. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- include/sound/soc-dapm.h | 3 +++ sound/soc/soc-dapm.c | 23 +++++++++++++++++++++++ 2 files changed, 26 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8da3c24..1649ef2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -399,6 +399,9 @@ int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); +int snd_soc_dapm_get_regulator_status(struct snd_soc_dapm_context *dapm, + const char *pin); + /* Mostly internal - should not normally be used */ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5cbd2d7..b6c9df5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3092,6 +3092,29 @@ int snd_soc_dapm_get_pin_status(struct snd_soc_dapm_context *dapm, EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); /** + * snd_soc_dapm_get_regulator_status - get regulator supply status + * @dapm: DAPM context + * @pin: Regulator supply name + * + * Get regulator supply status - Enabled or disabled. + * + * Returns -1 for failure. + * Returns 0 if disabled. + * Returns 1 if enabled. + */ +int snd_soc_dapm_get_regulator_status(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + + if (!w) + return -1; + + return regulator_is_enabled((struct regulator *)w->priv); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_regulator_status); + +/** * snd_soc_dapm_ignore_suspend - ignore suspend status for DAPM endpoint * @dapm: DAPM context * @pin: audio signal pin endpoint (or start point) -- 1.7.8.3

2 1

[alsa-devel] Follow Up: "Why my linux OS has only 8 sound cards ? How can I modify it ?"
by Maciej Paszta 08 May '12

08 May '12

Hi! This issue has already been brought up by Chris Shen on Nov 8 2011. Quick reminder: I have 10 sound cards which need to have oss emulation enabled. However, I only see 8 devices (dsp-dsp7). Takashi suggested a patch to workaround this problem: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-November/045764.h… I've applied it to kernel 3.0.0 and the only result I get is message appearing in kernel logs: "unable to register OSS PCM device 8:0". I've done some debugging and the registration fails on these lines (sound/core/sound_oss.c): register1 = register_sound_special_device(f_ops, minor, carddev); if (register1 != minor) { snd_printk(KERN_ERR "OSS: 4, register1: %i, minor %i\n", register1, minor); goto __end; } the debug lines seem to be constant about the value of "register1" == -2. I would really appreciate any help with this problem and I will be glad to test any patches, that may solve the issue. -- Maciej Paszta Mobile Systems Research Labs, Poznan University of Technology

3 4

[alsa-devel] Strange Problem with snd-hda-intel
by Frank Maik Kittelmann 08 May '12

08 May '12

We have some strange behavior in our audio application using the Realtek ALC 888 hardware on a Jetway JNC96FL-525-LF mainboard (Intel Atom D525+NM10) in ubuntu 10.10 (2.6.35-32-generic-pae) with the hda-intel driver (ALSA version 1.0.23). Sound output is done by mpd (0.17 alpha) via hw:0,1 (SPDIF). Basically the main problem seems to be a buffer or memory management issue. Once a playback is stopped the very last sample is repeated forever, independently from sampling rate or sound file format. This sample leads to a constant DC voltage after the DA converters. If we mute the SPDIF Output using the ALSAMIXER, there is no S/Pdif signal, hence no DC. By demuting again, the same sample appears again. The annoying thing is, that this sample always stays for a while if we play another sound file. If the sample has been at a high value, this leads to a loud noise in the speaker. We already tried the following steps to solve the problem: 1.) we changed a line in hda_intel.c to reset the fifo --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1797,7 +1797,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); - azx_sd_writel(azx_dev, SD_CTL, 0); + azx_stream_reset(chip, azx_dev); azx_dev->bufsize = 0; azx_dev->period_bytes = 0; azx_dev->format_val = 0; 2.) we set 2 parameters in the .config file before kernel compiling CONFIG_SND_HDA_POWER_SAVE=y CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0 Both solutions didn´t work for us - still same problem. Has anyone another idea how to solve this? Would be great if we could fix this. Frank

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5631: Convert to module_i2c_driver()
by Mark Brown 08 May '12

08 May '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/rt5631.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 95147e4..a64f2a7 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1751,17 +1751,7 @@ static struct i2c_driver rt5631_i2c_driver = { .id_table = rt5631_i2c_id, }; -static int __init rt5631_modinit(void) -{ - return i2c_add_driver(&rt5631_i2c_driver); -} -module_init(rt5631_modinit); - -static void __exit rt5631_modexit(void) -{ - i2c_del_driver(&rt5631_i2c_driver); -} -module_exit(rt5631_modexit); +module_i2c_driver(rt5631_i2c_driver); MODULE_DESCRIPTION("ASoC RT5631 driver"); MODULE_AUTHOR("flove <flove(a)realtek.com>"); -- 1.7.10

1 1

Re: [alsa-devel] [ALSA hda 3.4-rc] suspend2ram regression in snd_hda_intel
by Takashi Iwai 08 May '12

08 May '12

At Tue, 08 May 2012 11:13:23 +0400, Konstantin Khlebnikov wrote: > > Takashi Iwai wrote: > > At Sun, 06 May 2012 12:32:14 +0400, > > Konstantin Khlebnikov wrote: > >> > >> Konstantin Khlebnikov wrote: > >>> Mainline commit 785f857d1cb0856b612b46a0545b74aa2596e44a > >>> ("ALSA: hda - Set codec to D3 forcibly even if not used") > >>> breaks sound for my notebook (thinkpad x220), > >>> looks like codec cannot wake from power-save state after resume. > >>> After reverting looks like all works fine. > >>> > >>> how to reproduce: > >>> * unplug the power cord (to enable hda power safe mode) > >>> * play something and stop > >>> * wait until the background noise disappears (codec went into power-save mode) > >>> * suspend > >>> * resume > >>> * play something again and hear the silence > >>> -- > >>> To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > >>> the body of a message to majordomo(a)vger.kernel.org > >>> More majordomo info at http://vger.kernel.org/majordomo-info.html > >>> Please read the FAQ at http://www.tux.org/lkml/ > >> > >> Bump > > > > It's a weekend, so don't expect quick responces from any normal person > > :) > > Yeah, sorry. In this country a mess with holidays and weekends. > > > > > Which 3.4-rc did you try? There was a known problem with this codec > > regarding the power-up sequence, but this should have been fixed in > > the very recent version. > > Bug still in latest v3.4-rc6. > This is serious degradation since v3.3, so I hope it will be fixed before release. > > > > > If the latest Linus tree still shows the problem, try sound git tree > > master or for-next branch. > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git > > I don't see there anything related to this problem. Bug is still there. > It's 100% reproductible and disappears after reverting that commit. > Looks like driver missed one power-up cycle after system-resume, > after second power-save down/up cycle all work fine. OK. Could you try to pass model=auto and see whether it changes the behavior? I've checked only the auto-parser mode recently. It might be just the static quirk that got broken. Note that the mixer elements will be renamed by this, so make sure that you adjust the mixer before testing S3. Takashi

2 6

[alsa-devel] [PATCH 01/11] sound: core: Add debug-print helper function
by Ola Lilja 08 May '12

08 May '12

Adds a function getting the stream-name as a string for a specific stream. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- include/sound/pcm.h | 2 ++ sound/core/pcm.c | 10 ++++++++++ 2 files changed, 12 insertions(+), 0 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0d11128..8a153e6 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1073,4 +1073,6 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) const char *snd_pcm_format_name(snd_pcm_format_t format); +const char *snd_pcm_stream_str(struct snd_pcm_substream *substream); + #endif /* __SOUND_PCM_H */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 1a3070b..9e83f00 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1187,6 +1187,16 @@ static void snd_pcm_proc_done(void) #define snd_pcm_proc_done() #endif /* CONFIG_PROC_FS */ +/** + * Get a string naming the direction of a stream + */ +const char *snd_pcm_stream_str(struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} /* * ENTRY functions -- 1.7.8.3

2 1

[alsa-devel] [PATCH 07/11] arm: ux500: Add audio-related platform-data
by Ola Lilja 08 May '12

08 May '12

This patch adds audio-related platform-data like hardwired settings for microphones and common-mode voltage for the earpiece. There is also some refactoring of the msp/audio-files. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- arch/arm/mach-ux500/Makefile | 3 +- arch/arm/mach-ux500/board-mop500-audio.c | 58 +++++++++++++++ arch/arm/mach-ux500/board-mop500-msp.c | 76 ++++++------------- arch/arm/mach-ux500/board-mop500.c | 5 +- arch/arm/mach-ux500/board-mop500.h | 2 + .../mach-ux500/include/mach/board-mop500-audio.h | 52 +++++++++++++ .../arm/mach-ux500/include/mach/board-mop500-msp.h | 32 ++++++++ 7 files changed, 174 insertions(+), 54 deletions(-) create mode 100644 arch/arm/mach-ux500/board-mop500-audio.c create mode 100644 arch/arm/mach-ux500/include/mach/board-mop500-audio.h create mode 100644 arch/arm/mach-ux500/include/mach/board-mop500-msp.h diff --git a/arch/arm/mach-ux500/Makefile b/arch/arm/mach-ux500/Makefile index 11729bc..f97cee8 100644 --- a/arch/arm/mach-ux500/Makefile +++ b/arch/arm/mach-ux500/Makefile @@ -12,7 +12,8 @@ obj-$(CONFIG_MACH_MOP500) += board-mop500.o board-mop500-sdi.o \ board-mop500-uib.o board-mop500-stuib.o \ board-mop500-u8500uib.o \ board-mop500-pins.o \ - board-mop500-msp.o + board-mop500-msp.o \ + board-mop500-audio.o obj-$(CONFIG_MACH_U5500) += board-u5500.o board-u5500-sdi.o obj-$(CONFIG_SMP) += platsmp.o headsmp.o obj-$(CONFIG_HOTPLUG_CPU) += hotplug.o diff --git a/arch/arm/mach-ux500/board-mop500-audio.c b/arch/arm/mach-ux500/board-mop500-audio.c new file mode 100644 index 0000000..e50b6a9 --- /dev/null +++ b/arch/arm/mach-ux500/board-mop500-audio.c @@ -0,0 +1,58 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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/platform_device.h> + +#include <mach/board-mop500-audio.h> + +static struct mop500_audio_platform_data mop500_audio_pdata = { + .amics = { + .mic1_type = AMIC_TYPE_DIFFERENTIAL, + .mic2_type = AMIC_TYPE_DIFFERENTIAL, + .mic1a_micbias = AMIC_MICBIAS_VAMIC1, + .mic1b_micbias = AMIC_MICBIAS_VAMIC1, + .mic2_micbias = AMIC_MICBIAS_VAMIC2 + }, + .ear_cmv = EAR_CMV_0_95V +}; + +/* Platform device for MOP500 audio-parts */ +static struct platform_device mop500_audio = { + .name = "snd-soc-mop500", + .id = 0, + .dev = { + .platform_data = &mop500_audio_pdata, + }, +}; + +/* Platform device for Ux500-PCM */ +static struct platform_device ux500_pcm = { + .name = "ux500-pcm", + .id = 0, + .dev = { + .platform_data = NULL, + }, +}; + +int mop500_audio_init(struct device *parent) +{ + int ret; + + pr_info("%s: Register platform-device 'snd-soc-mop500'.\n", __func__); + ret = platform_device_register(&mop500_audio); + + pr_info("%s: Register platform-device 'ux500-pcm'\n", __func__); + ret |= platform_device_register(&ux500_pcm); + + return ret; +}; diff --git a/arch/arm/mach-ux500/board-mop500-msp.c b/arch/arm/mach-ux500/board-mop500-msp.c index c8f6300..2354bd0 100644 --- a/arch/arm/mach-ux500/board-mop500-msp.c +++ b/arch/arm/mach-ux500/board-mop500-msp.c @@ -5,21 +5,16 @@ */ #include <linux/platform_device.h> -#include <linux/init.h> #include <linux/gpio.h> -#include <plat/gpio-nomadik.h> +#include <plat/gpio-nomadik.h> #include <plat/pincfg.h> #include <plat/ste_dma40.h> -#include <mach/devices.h> #include <ste-dma40-db8500.h> -#include <mach/hardware.h> -#include <mach/irqs.h> -#include <mach/msp.h> -#include "board-mop500.h" -#include "devices-db8500.h" +#include <mach/board-mop500-msp.h> + #include "pins-db8500.h" /* MSP1/3 Tx/Rx usage protection */ @@ -100,7 +95,7 @@ static struct stedma40_chan_cfg msp0_dma_tx = { /* data_width is set during configuration */ }; -static struct msp_i2s_platform_data msp0_platform_data = { +static struct msp_i2s_platform_data msp0_pdata = { .id = MSP_I2S_0, .msp_i2s_dma_rx = &msp0_dma_rx, .msp_i2s_dma_tx = &msp0_dma_tx, @@ -132,7 +127,7 @@ static struct stedma40_chan_cfg msp1_dma_tx = { /* data_width is set during configuration */ }; -static struct msp_i2s_platform_data msp1_platform_data = { +static struct msp_i2s_platform_data msp1_pdata = { .id = MSP_I2S_1, .msp_i2s_dma_rx = NULL, .msp_i2s_dma_tx = &msp1_dma_tx, @@ -170,7 +165,7 @@ static struct stedma40_chan_cfg msp2_dma_tx = { /* data_width is set during configuration */ }; -static int db8500_add_msp_i2s(struct device *parent, int id, +static int mop500_msp_register_device(struct device *parent, int id, resource_size_t base, int irq, struct msp_i2s_platform_data *pdata) { @@ -180,46 +175,28 @@ static int db8500_add_msp_i2s(struct device *parent, int id, DEFINE_RES_IRQ(irq), }; - pr_info("Register platform-device 'ux500-msp-i2s', id %d, irq %d\n", - id, irq); + pr_info("%s: Register platform-device 'ux500-msp-i2s', id %d, irq %d\n", + __func__, id, irq); pdev = platform_device_register_resndata(parent, "ux500-msp-i2s", id, res, ARRAY_SIZE(res), pdata, sizeof(*pdata)); if (!pdev) { - pr_err("Failed to register platform-device 'ux500-msp-i2s.%d'!\n", - id); + pr_err("%s: Failed to register platform-device 'ux500-msp-i2s.%d'!\n", + __func__, id); return -EIO; } return 0; } -/* Platform device for ASoC U8500 machine */ -static struct platform_device snd_soc_u8500 = { - .name = "snd-soc-u8500", - .id = 0, - .dev = { - .platform_data = NULL, - }, -}; - -/* Platform device for Ux500-PCM */ -static struct platform_device ux500_pcm = { - .name = "ux500-pcm", - .id = 0, - .dev = { - .platform_data = NULL, - }, -}; - -static struct msp_i2s_platform_data msp2_platform_data = { +static struct msp_i2s_platform_data msp2_pdata = { .id = MSP_I2S_2, .msp_i2s_dma_rx = &msp2_dma_rx, .msp_i2s_dma_tx = &msp2_dma_tx, }; -static struct msp_i2s_platform_data msp3_platform_data = { - .id = MSP_I2S_3, +static struct msp_i2s_platform_data msp3_pdata = { + .id = MSP_I2S_3, .msp_i2s_dma_rx = &msp1_dma_rx, .msp_i2s_dma_tx = NULL, .msp_i2s_init = msp13_i2s_init, @@ -230,21 +207,16 @@ int mop500_msp_init(struct device *parent) { int ret; - pr_info("%s: Register platform-device 'snd-soc-u8500'.\n", __func__); - platform_device_register(&snd_soc_u8500); - - pr_info("Initialize MSP I2S-devices.\n"); - ret = db8500_add_msp_i2s(parent, 0, U8500_MSP0_BASE, IRQ_DB8500_MSP0, - &msp0_platform_data); - ret |= db8500_add_msp_i2s(parent, 1, U8500_MSP1_BASE, IRQ_DB8500_MSP1, - &msp1_platform_data); - ret |= db8500_add_msp_i2s(parent, 2, U8500_MSP2_BASE, IRQ_DB8500_MSP2, - &msp2_platform_data); - ret |= db8500_add_msp_i2s(parent, 3, U8500_MSP3_BASE, IRQ_DB8500_MSP1, - &msp3_platform_data); - - pr_info("%s: Register platform-device 'ux500-pcm'\n", __func__); - platform_device_register(&ux500_pcm); + pr_info("%s: Initialize MSP I2S-devices.\n", __func__); + ret = mop500_msp_register_device(parent, 0, U8500_MSP0_BASE, + IRQ_DB8500_MSP0, &msp0_pdata); + ret |= mop500_msp_register_device(parent, 1, U8500_MSP1_BASE, + IRQ_DB8500_MSP1, &msp1_pdata); + ret |= mop500_msp_register_device(parent, 2, U8500_MSP2_BASE, + IRQ_DB8500_MSP2, &msp2_pdata); + ret |= mop500_msp_register_device(parent, 3, U8500_MSP3_BASE, + IRQ_DB8500_MSP1, &msp3_pdata); return ret; -} +}; + diff --git a/arch/arm/mach-ux500/board-mop500.c b/arch/arm/mach-ux500/board-mop500.c index 4cbed08..3d975df 100644 --- a/arch/arm/mach-ux500/board-mop500.c +++ b/arch/arm/mach-ux500/board-mop500.c @@ -53,7 +53,7 @@ #include "devices-db8500.h" #include "board-mop500.h" #include "board-mop500-regulators.h" -#include "board-mop500-msp.h" + static struct gpio_led snowball_led_array[] = { { @@ -633,6 +633,7 @@ static void __init mop500_init_machine(void) mop500_sdi_init(parent); mop500_spi_init(parent); mop500_msp_init(parent); + mop500_audio_init(parent); mop500_uart_init(parent); i2c0_devs = ARRAY_SIZE(mop500_i2c0_devices); @@ -665,6 +666,7 @@ static void __init snowball_init_machine(void) snowball_sdi_init(parent); mop500_spi_init(parent); mop500_msp_init(parent); + mop500_audio_init(parent); mop500_uart_init(parent); i2c0_devs = ARRAY_SIZE(mop500_i2c0_devices); @@ -703,6 +705,7 @@ static void __init hrefv60_init_machine(void) hrefv60_sdi_init(parent); mop500_spi_init(parent); mop500_msp_init(parent); + mop500_audio_init(parent); mop500_uart_init(parent); i2c0_devs = ARRAY_SIZE(mop500_i2c0_devices); diff --git a/arch/arm/mach-ux500/board-mop500.h b/arch/arm/mach-ux500/board-mop500.h index fdcfa87..a5e1ccb 100644 --- a/arch/arm/mach-ux500/board-mop500.h +++ b/arch/arm/mach-ux500/board-mop500.h @@ -84,6 +84,8 @@ void __init mop500_stuib_init(void); void __init mop500_pins_init(void); void __init hrefv60_pins_init(void); void __init snowball_pins_init(void); +int mop500_msp_init(struct device *parent); +int mop500_audio_init(struct device *parent); void mop500_uib_i2c_add(int busnum, struct i2c_board_info *info, unsigned n); diff --git a/arch/arm/mach-ux500/include/mach/board-mop500-audio.h b/arch/arm/mach-ux500/include/mach/board-mop500-audio.h new file mode 100644 index 0000000..15ffea6 --- /dev/null +++ b/arch/arm/mach-ux500/include/mach/board-mop500-audio.h @@ -0,0 +1,52 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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. + */ + +#ifndef __BOARD_MOP500_AUDIO_H_ +#define __BOARD_MOP500_AUDIO_H_ + +/* Mic-types */ +enum amic_type { + AMIC_TYPE_SINGLE_ENDED, + AMIC_TYPE_DIFFERENTIAL +}; + +/* Mic-biases */ +enum amic_micbias { + AMIC_MICBIAS_VAMIC1, + AMIC_MICBIAS_VAMIC2 +}; + +/* Bias-voltage */ +enum ear_cm_voltage { + EAR_CMV_0_95V, + EAR_CMV_1_10V, + EAR_CMV_1_27V, + EAR_CMV_1_58V +}; + +/* Analog microphone settings */ +struct amic_settings { + enum amic_type mic1_type; + enum amic_type mic2_type; + enum amic_micbias mic1a_micbias; + enum amic_micbias mic1b_micbias; + enum amic_micbias mic2_micbias; +}; + +/* Platform data structure for the audio-parts of the MOP500 */ +struct mop500_audio_platform_data { + struct amic_settings amics; + enum ear_cm_voltage ear_cmv; +}; + +#endif diff --git a/arch/arm/mach-ux500/include/mach/board-mop500-msp.h b/arch/arm/mach-ux500/include/mach/board-mop500-msp.h new file mode 100644 index 0000000..3c096b2 --- /dev/null +++ b/arch/arm/mach-ux500/include/mach/board-mop500-msp.h @@ -0,0 +1,32 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * Author: Rabin Vincent <rabin.vincent(a)stericsson.com> for ST-Ericsson + * License terms: GNU General Public License (GPL), version 2. + */ + + +#ifndef __BOARD_MOP500_MSP_H_ +#define __BOARD_MOP500_MSP_H_ + +#include <plat/ste_dma40.h> + +enum msp_i2s_id { + MSP_I2S_0 = 0, + MSP_I2S_1, + MSP_I2S_2, + MSP_I2S_3, +}; + +/* Platform data structure for a MSP I2S-device */ +struct msp_i2s_platform_data { + enum msp_i2s_id id; + struct stedma40_chan_cfg *msp_i2s_dma_rx; + struct stedma40_chan_cfg *msp_i2s_dma_tx; + int (*msp_i2s_init) (void); + int (*msp_i2s_exit) (void); +}; + + + +#endif -- 1.7.8.3

1 0

[alsa-devel] [PATCH 06/11] arm: ux500: Add support for MSP I2S-devices
by Ola Lilja 08 May '12

08 May '12

Create devices for the MSP-blocks (MSP0, MSP1, MSP2 and MSP3) and associate it with the correct clocks in the clock-framework. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- arch/arm/mach-ux500/Makefile | 3 +- arch/arm/mach-ux500/board-mop500-msp.c | 250 ++++++++++++++++++++++++++++++++ arch/arm/mach-ux500/board-mop500-msp.h | 14 ++ arch/arm/mach-ux500/board-mop500.c | 4 + arch/arm/mach-ux500/clock.c | 8 +- arch/arm/mach-ux500/devices-db8500.h | 10 -- arch/arm/mach-ux500/include/mach/msp.h | 29 ++++ 7 files changed, 304 insertions(+), 14 deletions(-) create mode 100644 arch/arm/mach-ux500/board-mop500-msp.c create mode 100644 arch/arm/mach-ux500/board-mop500-msp.h create mode 100644 arch/arm/mach-ux500/include/mach/msp.h diff --git a/arch/arm/mach-ux500/Makefile b/arch/arm/mach-ux500/Makefile index 465b9ec..11729bc 100644 --- a/arch/arm/mach-ux500/Makefile +++ b/arch/arm/mach-ux500/Makefile @@ -11,7 +11,8 @@ obj-$(CONFIG_MACH_MOP500) += board-mop500.o board-mop500-sdi.o \ board-mop500-regulators.o \ board-mop500-uib.o board-mop500-stuib.o \ board-mop500-u8500uib.o \ - board-mop500-pins.o + board-mop500-pins.o \ + board-mop500-msp.o obj-$(CONFIG_MACH_U5500) += board-u5500.o board-u5500-sdi.o obj-$(CONFIG_SMP) += platsmp.o headsmp.o obj-$(CONFIG_HOTPLUG_CPU) += hotplug.o diff --git a/arch/arm/mach-ux500/board-mop500-msp.c b/arch/arm/mach-ux500/board-mop500-msp.c new file mode 100644 index 0000000..c8f6300 --- /dev/null +++ b/arch/arm/mach-ux500/board-mop500-msp.c @@ -0,0 +1,250 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * License terms: GNU General Public License (GPL), version 2 + */ + +#include <linux/platform_device.h> +#include <linux/init.h> +#include <linux/gpio.h> +#include <plat/gpio-nomadik.h> + +#include <plat/pincfg.h> +#include <plat/ste_dma40.h> + +#include <mach/devices.h> +#include <ste-dma40-db8500.h> +#include <mach/hardware.h> +#include <mach/irqs.h> +#include <mach/msp.h> + +#include "board-mop500.h" +#include "devices-db8500.h" +#include "pins-db8500.h" + +/* MSP1/3 Tx/Rx usage protection */ +static DEFINE_SPINLOCK(msp_rxtx_lock); + +/* Reference Count */ +static int msp_rxtx_ref; + +static pin_cfg_t mop500_msp1_pins_init[] = { + GPIO33_MSP1_TXD | PIN_OUTPUT_LOW | PIN_SLPM_WAKEUP_DISABLE, + GPIO34_MSP1_TFS | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_DISABLE, + GPIO35_MSP1_TCK | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_DISABLE, + GPIO36_MSP1_RXD | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_DISABLE, +}; + +static pin_cfg_t mop500_msp1_pins_exit[] = { + GPIO33_MSP1_TXD | PIN_OUTPUT_LOW | PIN_SLPM_WAKEUP_ENABLE, + GPIO34_MSP1_TFS | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_ENABLE, + GPIO35_MSP1_TCK | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_ENABLE, + GPIO36_MSP1_RXD | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_ENABLE, +}; + +int msp13_i2s_init(void) +{ + int retval = 0; + unsigned long flags; + + spin_lock_irqsave(&msp_rxtx_lock, flags); + if (msp_rxtx_ref == 0) + retval = nmk_config_pins( + ARRAY_AND_SIZE(mop500_msp1_pins_init)); + if (!retval) + msp_rxtx_ref++; + spin_unlock_irqrestore(&msp_rxtx_lock, flags); + + return retval; +} + +int msp13_i2s_exit(void) +{ + int retval = 0; + unsigned long flags; + + spin_lock_irqsave(&msp_rxtx_lock, flags); + WARN_ON(!msp_rxtx_ref); + msp_rxtx_ref--; + if (msp_rxtx_ref == 0) + retval = nmk_config_pins_sleep( + ARRAY_AND_SIZE(mop500_msp1_pins_exit)); + spin_unlock_irqrestore(&msp_rxtx_lock, flags); + + return retval; +} + +static struct stedma40_chan_cfg msp0_dma_rx = { + .high_priority = true, + .dir = STEDMA40_PERIPH_TO_MEM, + + .src_dev_type = DB8500_DMA_DEV31_MSP0_RX_SLIM0_CH0_RX, + .dst_dev_type = STEDMA40_DEV_DST_MEMORY, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct stedma40_chan_cfg msp0_dma_tx = { + .high_priority = true, + .dir = STEDMA40_MEM_TO_PERIPH, + + .src_dev_type = STEDMA40_DEV_DST_MEMORY, + .dst_dev_type = DB8500_DMA_DEV31_MSP0_TX_SLIM0_CH0_TX, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct msp_i2s_platform_data msp0_platform_data = { + .id = MSP_I2S_0, + .msp_i2s_dma_rx = &msp0_dma_rx, + .msp_i2s_dma_tx = &msp0_dma_tx, +}; + +static struct stedma40_chan_cfg msp1_dma_rx = { + .high_priority = true, + .dir = STEDMA40_PERIPH_TO_MEM, + + .src_dev_type = DB8500_DMA_DEV30_MSP3_RX, + .dst_dev_type = STEDMA40_DEV_DST_MEMORY, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct stedma40_chan_cfg msp1_dma_tx = { + .high_priority = true, + .dir = STEDMA40_MEM_TO_PERIPH, + + .src_dev_type = STEDMA40_DEV_DST_MEMORY, + .dst_dev_type = DB8500_DMA_DEV30_MSP1_TX, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct msp_i2s_platform_data msp1_platform_data = { + .id = MSP_I2S_1, + .msp_i2s_dma_rx = NULL, + .msp_i2s_dma_tx = &msp1_dma_tx, + .msp_i2s_init = msp13_i2s_init, + .msp_i2s_exit = msp13_i2s_exit, +}; + +static struct stedma40_chan_cfg msp2_dma_rx = { + .high_priority = true, + .dir = STEDMA40_PERIPH_TO_MEM, + + .src_dev_type = DB8500_DMA_DEV14_MSP2_RX, + .dst_dev_type = STEDMA40_DEV_DST_MEMORY, + + /* MSP2 DMA doesn't work with PSIZE == 4 on DB8500v2 */ + .src_info.psize = STEDMA40_PSIZE_LOG_1, + .dst_info.psize = STEDMA40_PSIZE_LOG_1, + + /* data_width is set during configuration */ +}; + +static struct stedma40_chan_cfg msp2_dma_tx = { + .high_priority = true, + .dir = STEDMA40_MEM_TO_PERIPH, + + .src_dev_type = STEDMA40_DEV_DST_MEMORY, + .dst_dev_type = DB8500_DMA_DEV14_MSP2_TX, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + .use_fixed_channel = true, + .phy_channel = 1, + + /* data_width is set during configuration */ +}; + +static int db8500_add_msp_i2s(struct device *parent, int id, + resource_size_t base, int irq, + struct msp_i2s_platform_data *pdata) +{ + struct platform_device *pdev; + struct resource res[] = { + DEFINE_RES_MEM(base, SZ_4K), + DEFINE_RES_IRQ(irq), + }; + + pr_info("Register platform-device 'ux500-msp-i2s', id %d, irq %d\n", + id, irq); + pdev = platform_device_register_resndata(parent, "ux500-msp-i2s", id, + res, ARRAY_SIZE(res), + pdata, sizeof(*pdata)); + if (!pdev) { + pr_err("Failed to register platform-device 'ux500-msp-i2s.%d'!\n", + id); + return -EIO; + } + + return 0; +} + +/* Platform device for ASoC U8500 machine */ +static struct platform_device snd_soc_u8500 = { + .name = "snd-soc-u8500", + .id = 0, + .dev = { + .platform_data = NULL, + }, +}; + +/* Platform device for Ux500-PCM */ +static struct platform_device ux500_pcm = { + .name = "ux500-pcm", + .id = 0, + .dev = { + .platform_data = NULL, + }, +}; + +static struct msp_i2s_platform_data msp2_platform_data = { + .id = MSP_I2S_2, + .msp_i2s_dma_rx = &msp2_dma_rx, + .msp_i2s_dma_tx = &msp2_dma_tx, +}; + +static struct msp_i2s_platform_data msp3_platform_data = { + .id = MSP_I2S_3, + .msp_i2s_dma_rx = &msp1_dma_rx, + .msp_i2s_dma_tx = NULL, + .msp_i2s_init = msp13_i2s_init, + .msp_i2s_exit = msp13_i2s_exit, +}; + +int mop500_msp_init(struct device *parent) +{ + int ret; + + pr_info("%s: Register platform-device 'snd-soc-u8500'.\n", __func__); + platform_device_register(&snd_soc_u8500); + + pr_info("Initialize MSP I2S-devices.\n"); + ret = db8500_add_msp_i2s(parent, 0, U8500_MSP0_BASE, IRQ_DB8500_MSP0, + &msp0_platform_data); + ret |= db8500_add_msp_i2s(parent, 1, U8500_MSP1_BASE, IRQ_DB8500_MSP1, + &msp1_platform_data); + ret |= db8500_add_msp_i2s(parent, 2, U8500_MSP2_BASE, IRQ_DB8500_MSP2, + &msp2_platform_data); + ret |= db8500_add_msp_i2s(parent, 3, U8500_MSP3_BASE, IRQ_DB8500_MSP1, + &msp3_platform_data); + + pr_info("%s: Register platform-device 'ux500-pcm'\n", __func__); + platform_device_register(&ux500_pcm); + + return ret; +} diff --git a/arch/arm/mach-ux500/board-mop500-msp.h b/arch/arm/mach-ux500/board-mop500-msp.h new file mode 100644 index 0000000..6fcfb5e --- /dev/null +++ b/arch/arm/mach-ux500/board-mop500-msp.h @@ -0,0 +1,14 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * for ST-Ericsson. + * + * License terms: + * + * 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. + */ + +void mop500_msp_init(struct device *parent); diff --git a/arch/arm/mach-ux500/board-mop500.c b/arch/arm/mach-ux500/board-mop500.c index 9dfaab3..4cbed08 100644 --- a/arch/arm/mach-ux500/board-mop500.c +++ b/arch/arm/mach-ux500/board-mop500.c @@ -53,6 +53,7 @@ #include "devices-db8500.h" #include "board-mop500.h" #include "board-mop500-regulators.h" +#include "board-mop500-msp.h" static struct gpio_led snowball_led_array[] = { { @@ -631,6 +632,7 @@ static void __init mop500_init_machine(void) mop500_i2c_init(parent); mop500_sdi_init(parent); mop500_spi_init(parent); + mop500_msp_init(parent); mop500_uart_init(parent); i2c0_devs = ARRAY_SIZE(mop500_i2c0_devices); @@ -662,6 +664,7 @@ static void __init snowball_init_machine(void) mop500_i2c_init(parent); snowball_sdi_init(parent); mop500_spi_init(parent); + mop500_msp_init(parent); mop500_uart_init(parent); i2c0_devs = ARRAY_SIZE(mop500_i2c0_devices); @@ -699,6 +702,7 @@ static void __init hrefv60_init_machine(void) mop500_i2c_init(parent); hrefv60_sdi_init(parent); mop500_spi_init(parent); + mop500_msp_init(parent); mop500_uart_init(parent); i2c0_devs = ARRAY_SIZE(mop500_i2c0_devices); diff --git a/arch/arm/mach-ux500/clock.c b/arch/arm/mach-ux500/clock.c index cc87f77..9c2ffcdf 100644 --- a/arch/arm/mach-ux500/clock.c +++ b/arch/arm/mach-ux500/clock.c @@ -336,6 +336,7 @@ static DEFINE_PRCMU_CLK(uiccclk, 0x4, 1, UICCCLK); /* v1 */ */ /* Peripheral Cluster #1 */ +static DEFINE_PRCC_CLK(1, msp3, 11, 10, &clk_msp1clk); static DEFINE_PRCC_CLK(1, i2c4, 10, 9, &clk_i2cclk); static DEFINE_PRCC_CLK(1, gpio0, 9, -1, NULL); static DEFINE_PRCC_CLK(1, slimbus0, 8, 8, &clk_slimclk); @@ -405,7 +406,7 @@ static struct clk_lookup u8500_clks[] = { CLK(slimbus0, "slimbus0", NULL), CLK(i2c2, "nmk-i2c.2", NULL), CLK(sdi0, "sdi0", NULL), - CLK(msp0, "msp0", NULL), + CLK(msp0, "ux500-msp-i2s.0", NULL), CLK(i2c1, "nmk-i2c.1", NULL), CLK(uart1, "uart1", NULL), CLK(uart0, "uart0", NULL), @@ -455,7 +456,8 @@ static struct clk_lookup u8500_clks[] = { /* Peripheral Cluster #1 */ CLK(i2c4, "nmk-i2c.4", NULL), CLK(spi3, "spi3", NULL), - CLK(msp1, "msp1", NULL), + CLK(msp1, "ux500-msp-i2s.1", NULL), + CLK(msp3, "ux500-msp-i2s.3", NULL), /* Peripheral Cluster #2 */ CLK(gpio1, "gpio.6", NULL), @@ -465,7 +467,7 @@ static struct clk_lookup u8500_clks[] = { CLK(spi0, "spi0", NULL), CLK(sdi3, "sdi3", NULL), CLK(sdi1, "sdi1", NULL), - CLK(msp2, "msp2", NULL), + CLK(msp2, "ux500-msp-i2s.2", NULL), CLK(sdi4, "sdi4", NULL), CLK(pwl, "pwl", NULL), CLK(spi1, "spi1", NULL), diff --git a/arch/arm/mach-ux500/devices-db8500.h b/arch/arm/mach-ux500/devices-db8500.h index 9fd93e9..e22b786 100644 --- a/arch/arm/mach-ux500/devices-db8500.h +++ b/arch/arm/mach-ux500/devices-db8500.h @@ -34,7 +34,6 @@ db8500_add_ssp(struct device *parent, const char *name, resource_size_t base, return dbx500_add_amba_device(parent, name, base, irq, pdata, 0); } - #define db8500_add_i2c0(parent, pdata) \ dbx500_add_i2c(parent, 0, U8500_I2C0_BASE, IRQ_DB8500_I2C0, pdata) #define db8500_add_i2c1(parent, pdata) \ @@ -46,15 +45,6 @@ db8500_add_ssp(struct device *parent, const char *name, resource_size_t base, #define db8500_add_i2c4(parent, pdata) \ dbx500_add_i2c(parent, 4, U8500_I2C4_BASE, IRQ_DB8500_I2C4, pdata) -#define db8500_add_msp0_i2s(parent, pdata) \ - dbx500_add_msp_i2s(parent, 0, U8500_MSP0_BASE, IRQ_DB8500_MSP0, pdata) -#define db8500_add_msp1_i2s(parent, pdata) \ - dbx500_add_msp_i2s(parent, 1, U8500_MSP1_BASE, IRQ_DB8500_MSP1, pdata) -#define db8500_add_msp2_i2s(parent, pdata) \ - dbx500_add_msp_i2s(parent, 2, U8500_MSP2_BASE, IRQ_DB8500_MSP2, pdata) -#define db8500_add_msp3_i2s(parent, pdata) \ - dbx500_add_msp_i2s(parent, 3, U8500_MSP3_BASE, IRQ_DB8500_MSP1, pdata) - #define db8500_add_msp0_spi(parent, pdata) \ dbx500_add_msp_spi(parent, "msp0", U8500_MSP0_BASE, \ IRQ_DB8500_MSP0, pdata) diff --git a/arch/arm/mach-ux500/include/mach/msp.h b/arch/arm/mach-ux500/include/mach/msp.h new file mode 100644 index 0000000..798be19 --- /dev/null +++ b/arch/arm/mach-ux500/include/mach/msp.h @@ -0,0 +1,29 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * Author: Rabin Vincent <rabin.vincent(a)stericsson.com> for ST-Ericsson + * License terms: GNU General Public License (GPL), version 2. + */ + +#ifndef __MSP_H +#define __MSP_H + +#include <plat/ste_dma40.h> + +enum msp_i2s_id { + MSP_I2S_0 = 0, + MSP_I2S_1, + MSP_I2S_2, + MSP_I2S_3, +}; + +/* Platform data structure for a MSP I2S-device */ +struct msp_i2s_platform_data { + enum msp_i2s_id id; + struct stedma40_chan_cfg *msp_i2s_dma_rx; + struct stedma40_chan_cfg *msp_i2s_dma_tx; + int (*msp_i2s_init) (void); + int (*msp_i2s_exit) (void); +}; + +#endif -- 1.7.8.3

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[alsa-devel] [PATCH 05/11] ASoC: core: Add strobe control
by Ola Lilja 08 May '12

08 May '12

From: Kristoffer KARLSSON <kristoffer.karlsson(a)stericsson.com> Added support for a control that strobes a bit in a register to high then back to low (or the inverse). This is typically useful for hardware that requires strobing a singe bit to trigger some functionality and where exposing the bit in a normal single control would require the user to first manually set then again unset the bit again for the strobe to trigger. Added convenience macro. SOC_SINGLE_STROBE Added accessor implementations. snd_soc_get_strobe snd_soc_put_strobe Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson(a)stericsson.com> --- include/sound/soc.h | 8 ++++++ sound/soc/soc-core.c | 63 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 71 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8888adc..bda0cd2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -209,6 +209,10 @@ {.regbase = xregbase, .regcount = xregcount, .nbits = xnbits, \ .invert = xinvert, .min = xmin, .max = xmax} } +#define SOC_SINGLE_STROBE(xname, xreg, xshift, xinvert) \ + SOC_SINGLE_EXT(xname, xreg, xshift, 1, xinvert, \ + snd_soc_get_strobe, snd_soc_put_strobe) + /* * Simplified versions of above macros, declaring a struct and calculating * ARRAY_SIZE internally @@ -448,6 +452,10 @@ int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /** * struct snd_soc_reg_access - Describes whether a given register is diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7fc7294b..e9c5850 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2966,6 +2966,69 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); /** + * snd_soc_get_strobe - strobe get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback get the value of a strobe mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int val = snd_soc_read(codec, reg) & mask; + + if (shift != 0 && val != 0) + val = val >> shift; + ucontrol->value.enumerated.item[0] = val ^ invert; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_strobe); + +/** + * snd_soc_put_strobe - strobe put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback strobe a register bit to high then low (or the inverse) + * in one pass of a single mixer enum control. + * + * Returns 1 for success. + */ +int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int strobe = ucontrol->value.enumerated.item[0] != 0; + unsigned int val1 = (strobe ^ invert) ? mask : 0; + unsigned int val2 = (strobe ^ invert) ? 0 : mask; + int err; + + err = snd_soc_update_bits_locked(codec, reg, mask, val1); + if (err < 0) + return err; + + err = snd_soc_update_bits_locked(codec, reg, mask, val2); + return err; +} +EXPORT_SYMBOL_GPL(snd_soc_put_strobe); + +/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID -- 1.7.8.3

1 0

[alsa-devel] [PATCH 00/11] Fourth edition of Ux500 ASoC-driver
by Ola Lilja 08 May '12

08 May '12

This is the fourth (and hopefully last {-: ) compilation of patches for the ST-Ericsson Ux500 ASoC-driver. Since patch-set three there has been the following changes: *) Introduction of a clock supply-widget in the core *) Usage of regulator supply-widget *) Usage of clock supply-widget *) Board-specific settings moved to arch/arm/mach-ux500/ *) Bug-fixes for bidirectional audio *) More usage of ASoC-core functions *) Refactoring to get rid of non-ASoC functions etc... Kristoffer KARLSSON (2): ASoC: core: Add signed multi register control ASoC: core: Add strobe control Ola Lilja (9): sound: core: Add debug-print helper function ASoC: core: Add method snd_soc_dapm_get_regulator_status ASoC: core: Add widget SND_SOC_DAPM_CLOCK_SUPPLY arm: ux500: Add support for MSP I2S-devices arm: ux500: Add audio-related platform-data ASoC: Ux500: Add MSP I2S-driver ASoC: Ux500: Add platform-driver ASoC: codecs: Add AB8500 codec-driver ASoC: Ux500: Add machine-driver arch/arm/mach-ux500/Makefile | 4 +- arch/arm/mach-ux500/board-mop500-audio.c | 58 + arch/arm/mach-ux500/board-mop500-msp.c | 222 ++ arch/arm/mach-ux500/board-mop500-msp.h | 14 + arch/arm/mach-ux500/board-mop500.c | 7 + arch/arm/mach-ux500/board-mop500.h | 2 + arch/arm/mach-ux500/clock.c | 8 +- arch/arm/mach-ux500/devices-db8500.h | 10 - .../mach-ux500/include/mach/board-mop500-audio.h | 52 + .../arm/mach-ux500/include/mach/board-mop500-msp.h | 32 + arch/arm/mach-ux500/include/mach/msp.h | 29 + include/sound/pcm.h | 2 + include/sound/soc-dapm.h | 12 + include/sound/soc.h | 30 +- sound/core/pcm.c | 10 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/ab8500-codec.c | 2852 ++++++++++++++++++++ sound/soc/codecs/ab8500-codec.h | 608 +++++ sound/soc/soc-core.c | 181 ++ sound/soc/soc-dapm.c | 82 + sound/soc/ux500/Kconfig | 34 + sound/soc/ux500/Makefile | 10 + sound/soc/ux500/mop500.c | 117 + sound/soc/ux500/mop500_ab8500.c | 270 ++ sound/soc/ux500/mop500_ab8500.h | 21 + sound/soc/ux500/ux500_msp_dai.c | 843 ++++++ sound/soc/ux500/ux500_msp_dai.h | 79 + sound/soc/ux500/ux500_msp_i2s.c | 742 +++++ sound/soc/ux500/ux500_msp_i2s.h | 553 ++++ sound/soc/ux500/ux500_pcm.c | 320 +++ sound/soc/ux500/ux500_pcm.h | 35 + 34 files changed, 7233 insertions(+), 15 deletions(-) create mode 100644 arch/arm/mach-ux500/board-mop500-audio.c create mode 100644 arch/arm/mach-ux500/board-mop500-msp.c create mode 100644 arch/arm/mach-ux500/board-mop500-msp.h create mode 100644 arch/arm/mach-ux500/include/mach/board-mop500-audio.h create mode 100644 arch/arm/mach-ux500/include/mach/board-mop500-msp.h create mode 100644 arch/arm/mach-ux500/include/mach/msp.h create mode 100644 sound/soc/codecs/ab8500-codec.c create mode 100644 sound/soc/codecs/ab8500-codec.h create mode 100644 sound/soc/ux500/Kconfig create mode 100644 sound/soc/ux500/Makefile create mode 100644 sound/soc/ux500/mop500.c create mode 100644 sound/soc/ux500/mop500_ab8500.c create mode 100644 sound/soc/ux500/mop500_ab8500.h create mode 100644 sound/soc/ux500/ux500_msp_dai.c create mode 100644 sound/soc/ux500/ux500_msp_dai.h create mode 100644 sound/soc/ux500/ux500_msp_i2s.c create mode 100644 sound/soc/ux500/ux500_msp_i2s.h create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h -- 1.7.8.3

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[alsa-devel] [PATCH] ASoC: twl6040: Remove HS/HF gain ramp feature
by Peter Ujfalusi 08 May '12

08 May '12

None of the machines uses the gain ramp possibility for HS/HF. This code path is mostly unused and it does not reduces the pop noise on the output (it alters it to sound a bit different). The preferred method to reduce pop noise is to use ABE. Remove the gain ramp, and related features form the driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 441 ++------------------------------------------ 1 files changed, 12 insertions(+), 429 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index dc7509b..0747260 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -46,17 +46,6 @@ #define TWL6040_OUTHF_0dB 0x03 #define TWL6040_OUTHF_M52dB 0x1D -#define TWL6040_RAMP_NONE 0 -#define TWL6040_RAMP_UP 1 -#define TWL6040_RAMP_DOWN 2 - -#define TWL6040_HSL_VOL_MASK 0x0F -#define TWL6040_HSL_VOL_SHIFT 0 -#define TWL6040_HSR_VOL_MASK 0xF0 -#define TWL6040_HSR_VOL_SHIFT 4 -#define TWL6040_HF_VOL_MASK 0x1F -#define TWL6040_HF_VOL_SHIFT 0 - /* Shadow register used by the driver */ #define TWL6040_REG_SW_SHADOW 0x2F #define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1) @@ -64,18 +53,6 @@ /* TWL6040_REG_SW_SHADOW (0x2F) fields */ #define TWL6040_EAR_PATH_ENABLE 0x01 -struct twl6040_output { - u16 active; - u16 left_vol; - u16 right_vol; - u16 left_step; - u16 right_step; - unsigned int step_delay; - u16 ramp; - struct delayed_work work; - struct completion ramp_done; -}; - struct twl6040_jack_data { struct snd_soc_jack *jack; struct delayed_work work; @@ -100,8 +77,6 @@ struct twl6040_data { struct snd_soc_codec *codec; struct workqueue_struct *workqueue; struct mutex mutex; - struct twl6040_output headset; - struct twl6040_output handsfree; }; /* @@ -311,318 +286,6 @@ static void twl6040_restore_regs(struct snd_soc_codec *codec) } } -/* - * Ramp HS PGA volume to minimise pops at stream startup and shutdown. - */ -static inline int twl6040_hs_ramp_step(struct snd_soc_codec *codec, - unsigned int left_step, unsigned int right_step) -{ - - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *headset = &priv->headset; - int left_complete = 0, right_complete = 0; - u8 reg, val; - - /* left channel */ - left_step = (left_step > 0xF) ? 0xF : left_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); - val = (~reg & TWL6040_HSL_VOL_MASK); - - if (headset->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < headset->left_vol) { - if (val + left_step > headset->left_vol) - val = headset->left_vol; - else - val += left_step; - - reg &= ~TWL6040_HSL_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - (reg | (~val & TWL6040_HSL_VOL_MASK))); - } else { - left_complete = 1; - } - } else if (headset->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0x0) { - if ((int)val - (int)left_step < 0) - val = 0; - else - val -= left_step; - - reg &= ~TWL6040_HSL_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, reg | - (~val & TWL6040_HSL_VOL_MASK)); - } else { - left_complete = 1; - } - } - - /* right channel */ - right_step = (right_step > 0xF) ? 0xF : right_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); - val = (~reg & TWL6040_HSR_VOL_MASK) >> TWL6040_HSR_VOL_SHIFT; - - if (headset->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < headset->right_vol) { - if (val + right_step > headset->right_vol) - val = headset->right_vol; - else - val += right_step; - - reg &= ~TWL6040_HSR_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - (reg | (~val << TWL6040_HSR_VOL_SHIFT))); - } else { - right_complete = 1; - } - } else if (headset->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0x0) { - if ((int)val - (int)right_step < 0) - val = 0; - else - val -= right_step; - - reg &= ~TWL6040_HSR_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - reg | (~val << TWL6040_HSR_VOL_SHIFT)); - } else { - right_complete = 1; - } - } - - return left_complete & right_complete; -} - -/* - * Ramp HF PGA volume to minimise pops at stream startup and shutdown. - */ -static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec, - unsigned int left_step, unsigned int right_step) -{ - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *handsfree = &priv->handsfree; - int left_complete = 0, right_complete = 0; - u16 reg, val; - - /* left channel */ - left_step = (left_step > 0x1D) ? 0x1D : left_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFLGAIN); - reg = 0x1D - reg; - val = (reg & TWL6040_HF_VOL_MASK); - if (handsfree->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < handsfree->left_vol) { - if (val + left_step > handsfree->left_vol) - val = handsfree->left_vol; - else - val += left_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFLGAIN, - reg | (0x1D - val)); - } else { - left_complete = 1; - } - } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0) { - if ((int)val - (int)left_step < 0) - val = 0; - else - val -= left_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFLGAIN, - reg | (0x1D - val)); - } else { - left_complete = 1; - } - } - - /* right channel */ - right_step = (right_step > 0x1D) ? 0x1D : right_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFRGAIN); - reg = 0x1D - reg; - val = (reg & TWL6040_HF_VOL_MASK); - if (handsfree->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < handsfree->right_vol) { - if (val + right_step > handsfree->right_vol) - val = handsfree->right_vol; - else - val += right_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFRGAIN, - reg | (0x1D - val)); - } else { - right_complete = 1; - } - } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0) { - if ((int)val - (int)right_step < 0) - val = 0; - else - val -= right_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFRGAIN, - reg | (0x1D - val)); - } - } - - return left_complete & right_complete; -} - -/* - * This work ramps both output PGAs at stream start/stop time to - * minimise pop associated with DAPM power switching. - */ -static void twl6040_pga_hs_work(struct work_struct *work) -{ - struct twl6040_data *priv = - container_of(work, struct twl6040_data, headset.work.work); - struct snd_soc_codec *codec = priv->codec; - struct twl6040_output *headset = &priv->headset; - int i, headset_complete; - - /* do we need to ramp at all ? */ - if (headset->ramp == TWL6040_RAMP_NONE) - return; - - /* HS PGA gain range: 0x0 - 0xf (0 - 15) */ - for (i = 0; i < 16; i++) { - headset_complete = twl6040_hs_ramp_step(codec, - headset->left_step, - headset->right_step); - - /* ramp finished ? */ - if (headset_complete) - break; - - schedule_timeout_interruptible( - msecs_to_jiffies(headset->step_delay)); - } - - if (headset->ramp == TWL6040_RAMP_DOWN) { - headset->active = 0; - complete(&headset->ramp_done); - } else { - headset->active = 1; - } - headset->ramp = TWL6040_RAMP_NONE; -} - -static void twl6040_pga_hf_work(struct work_struct *work) -{ - struct twl6040_data *priv = - container_of(work, struct twl6040_data, handsfree.work.work); - struct snd_soc_codec *codec = priv->codec; - struct twl6040_output *handsfree = &priv->handsfree; - int i, handsfree_complete; - - /* do we need to ramp at all ? */ - if (handsfree->ramp == TWL6040_RAMP_NONE) - return; - - /* - * HF PGA gain range: 0x00 - 0x1d (0 - 29) */ - for (i = 0; i < 30; i++) { - handsfree_complete = twl6040_hf_ramp_step(codec, - handsfree->left_step, - handsfree->right_step); - - /* ramp finished ? */ - if (handsfree_complete) - break; - - schedule_timeout_interruptible( - msecs_to_jiffies(handsfree->step_delay)); - } - - - if (handsfree->ramp == TWL6040_RAMP_DOWN) { - handsfree->active = 0; - complete(&handsfree->ramp_done); - } else - handsfree->active = 1; - handsfree->ramp = TWL6040_RAMP_NONE; -} - -static int out_drv_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out; - struct delayed_work *work; - - switch (w->shift) { - case 2: /* Headset output driver */ - out = &priv->headset; - work = &out->work; - /* - * Make sure, that we do not mess up variables for already - * executing work. - */ - cancel_delayed_work_sync(work); - - out->left_step = priv->hs_left_step; - out->right_step = priv->hs_right_step; - out->step_delay = 5; /* 5 ms between volume ramp steps */ - break; - case 4: /* Handsfree output driver */ - out = &priv->handsfree; - work = &out->work; - /* - * Make sure, that we do not mess up variables for already - * executing work. - */ - cancel_delayed_work_sync(work); - - out->left_step = priv->hf_left_step; - out->right_step = priv->hf_right_step; - out->step_delay = 5; /* 5 ms between volume ramp steps */ - break; - default: - return -1; - } - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - if (out->active) - break; - - /* don't use volume ramp for power-up */ - out->ramp = TWL6040_RAMP_UP; - out->left_step = out->left_vol; - out->right_step = out->right_vol; - - queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); - break; - - case SND_SOC_DAPM_PRE_PMD: - if (!out->active) - break; - - /* use volume ramp for power-down */ - out->ramp = TWL6040_RAMP_DOWN; - INIT_COMPLETION(out->ramp_done); - - queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); - - wait_for_completion_timeout(&out->ramp_done, - msecs_to_jiffies(2000)); - break; - } - - return 0; -} - /* set headset dac and driver power mode */ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) { @@ -747,71 +410,6 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) return IRQ_HANDLED; } -static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = NULL; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int ret; - - /* For HS and HF we shadow the values and only actually write - * them out when active in order to ensure the amplifier comes on - * as quietly as possible. */ - switch (mc->reg) { - case TWL6040_REG_HSGAIN: - out = &twl6040_priv->headset; - break; - case TWL6040_REG_HFLGAIN: - out = &twl6040_priv->handsfree; - break; - default: - dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", - __func__, mc->reg); - return -EINVAL; - } - - out->left_vol = ucontrol->value.integer.value[0]; - out->right_vol = ucontrol->value.integer.value[1]; - if (!out->active) - return 1; - - ret = snd_soc_put_volsw(kcontrol, ucontrol); - if (ret < 0) - return ret; - - return 1; -} - -static int twl6040_get_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = &twl6040_priv->headset; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - - switch (mc->reg) { - case TWL6040_REG_HSGAIN: - out = &twl6040_priv->headset; - break; - case TWL6040_REG_HFLGAIN: - out = &twl6040_priv->handsfree; - break; - default: - dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", - __func__, mc->reg); - return -EINVAL; - } - - ucontrol->value.integer.value[0] = out->left_vol; - ucontrol->value.integer.value[1] = out->right_vol; - return 0; -} - static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1076,12 +674,10 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv), /* Playback gains */ - SOC_DOUBLE_EXT_TLV("Headset Playback Volume", - TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, twl6040_get_volsw, - twl6040_put_volsw, hs_tlv), - SOC_DOUBLE_R_EXT_TLV("Handsfree Playback Volume", - TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, - twl6040_get_volsw, twl6040_put_volsw, hf_tlv), + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), SOC_SINGLE_TLV("Earphone Playback Volume", TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv), @@ -1180,22 +776,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { &auxr_switch_control), /* Analog playback drivers */ - SND_SOC_DAPM_OUT_DRV_E("HF Left Driver", - TWL6040_REG_HFLCTL, 4, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HF Right Driver", - TWL6040_REG_HFRCTL, 4, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HS Left Driver", - TWL6040_REG_HSLCTL, 2, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HS Right Driver", - TWL6040_REG_HSRCTL, 2, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUT_DRV("HF Left Driver", + TWL6040_REG_HFLCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HF Right Driver", + TWL6040_REG_HFRCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Left Driver", + TWL6040_REG_HSLCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Right Driver", + TWL6040_REG_HSRCTL, 2, 0, NULL, 0), SND_SOC_DAPM_OUT_DRV_E("Earphone Driver", TWL6040_REG_EARCTL, 0, 0, NULL, 0, twl6040_ep_drv_event, @@ -1570,14 +1158,9 @@ static int twl6040_probe(struct snd_soc_codec *codec) } INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work); - INIT_DELAYED_WORK(&priv->headset.work, twl6040_pga_hs_work); - INIT_DELAYED_WORK(&priv->handsfree.work, twl6040_pga_hf_work); mutex_init(&priv->mutex); - init_completion(&priv->headset.ramp_done); - init_completion(&priv->handsfree.ramp_done); - ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler, 0, "twl6040_irq_plug", codec); if (ret) { -- 1.7.8.6

2 2

[alsa-devel] [PATCH] ASoC: wm9081: Hook DAC up via DAPM rather than stream
by Mark Brown 08 May '12

08 May '12

More current API usage. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 076c126..9328270 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -774,7 +774,7 @@ static const struct snd_soc_dapm_widget wm9081_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1"), SND_SOC_DAPM_INPUT("IN2"), -SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM9081_POWER_MANAGEMENT, 0, 0), +SND_SOC_DAPM_DAC("DAC", NULL, WM9081_POWER_MANAGEMENT, 0, 0), SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, mixer, ARRAY_SIZE(mixer)), @@ -799,6 +799,7 @@ SND_SOC_DAPM_SUPPLY("TSENSE", WM9081_POWER_MANAGEMENT, 7, 0, NULL, 0), static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "DAC", NULL, "CLK_SYS" }, { "DAC", NULL, "CLK_DSP" }, + { "DAC", NULL, "AIF" }, { "Mixer", "IN1 Switch", "IN1" }, { "Mixer", "IN2 Switch", "IN2" }, @@ -1252,7 +1253,7 @@ static const struct snd_soc_dai_ops wm9081_dai_ops = { static struct snd_soc_dai_driver wm9081_dai = { .name = "wm9081-hifi", .playback = { - .stream_name = "HiFi Playback", + .stream_name = "AIF", .channels_min = 1, .channels_max = 2, .rates = WM9081_RATES, -- 1.7.10

1 0

[alsa-devel] [PATCH] ASoC: pcm: Fix DPCM for aux_devs
by Mark Brown 08 May '12

08 May '12

When we instantiate an aux_dev we use a fake rtd as part of the process which doesn't have a dai_link associated with it. Fix the dpcm startup code to cope with this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a94512c..a3ea8a7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2332,6 +2332,9 @@ static const struct file_operations dpcm_state_fops = { int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) { + if (!rtd->dai_link) + return 0; + rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name, rtd->card->debugfs_card_root); if (!rtd->debugfs_dpcm_root) { -- 1.7.10

2 1

[alsa-devel] snd-hda-intel+dmix: Disabling power save doesn't work
by Adam Nielsen 08 May '12

08 May '12

Hi all, I'm trying to hunt down a bug in either dmix or the snd-hda-intel module and I'm hoping someone can give me some pointers. When I am playing audio and I pause playback, the codec goes into power save and stops generating a SPDIF signal. My amplifier then displays a 'signal lost' message. This is quite annoying as it takes a few seconds to sync when the signal returns (during which time I can't hear the audio being played), so I set "powersave=0" to disable this behaviour. This works as expected when stopping playback, but if I only *pause* playback then the signal drops almost exactly one second later, no matter what "powersave" is set to. So of course then resuming playback means I lose a couple of seconds of audio while the amplifier re-syncs to the SPDIF signal. Setting powersave to other values only affects the time before going into powersave mode after playback has stopped. It has no effect on whether power saving is activated while paused. The problem happens with any audio program (e.g. mplayer and xmms2.) Setting "powersave=0" means the SPDIF signal keeps going when stopping playback, but it still drops after one second when pausing playback. It doesn't happen if I use hw:0,1 as the output device (the SPDIF signal keeps going, even when paused) but as soon as I change the output device to dmix then the signal drops whenever the audio is paused. Is there any way to stop this behaviour and make dmix honour the module setting? Thanks, Adam.

2 3

[alsa-devel] [PATCH] ASoC: lowland: Support digital link for WM9081
by Mark Brown 08 May '12

08 May '12

The WM9081 on Lowland is connected to AIF3 on the WM5100. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/lowland.c | 42 +++++++++++++++++++++++++++--------------- 1 file changed, 27 insertions(+), 15 deletions(-) diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 6cac1e9..6abf341 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -74,6 +74,25 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int lowland_wm9081_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + + snd_soc_dapm_nc_pin(&codec->dapm, "LINEOUT"); + + /* At any time the WM9081 is active it will have this clock */ + return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0, + CLKOUT_RATE, 0); +} + +static const struct snd_soc_pcm_stream sub_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, + .channels_max = 2, +}; + static struct snd_soc_dai_link lowland_dai[] = { { .name = "CPU", @@ -96,21 +115,16 @@ static struct snd_soc_dai_link lowland_dai[] = { SND_SOC_DAIFMT_CBM_CFM, .ignore_suspend = 1, }, -}; - -static int lowland_wm9081_init(struct snd_soc_dapm_context *dapm) -{ - snd_soc_dapm_nc_pin(dapm, "LINEOUT"); - - /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, - CLKOUT_RATE, 0); -} - -static struct snd_soc_aux_dev lowland_aux_dev[] = { { - .name = "wm9081", + .name = "Sub Speaker", + .stream_name = "Sub Speaker", + .cpu_dai_name = "wm5100-aif3", + .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &sub_params, .init = lowland_wm9081_init, }, }; @@ -155,8 +169,6 @@ static struct snd_soc_card lowland = { .owner = THIS_MODULE, .dai_link = lowland_dai, .num_links = ARRAY_SIZE(lowland_dai), - .aux_dev = lowland_aux_dev, - .num_aux_devs = ARRAY_SIZE(lowland_aux_dev), .codec_conf = lowland_codec_conf, .num_configs = ARRAY_SIZE(lowland_codec_conf), -- 1.7.10

1 0

[alsa-devel] [PATCH] ASoC: lowland: Convert to dai_fmt
by Mark Brown 08 May '12

08 May '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/lowland.c | 33 ++++----------------------------- 1 file changed, 4 insertions(+), 29 deletions(-) diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 4adff93..6cac1e9 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -21,33 +21,6 @@ #define MCLK1_RATE (44100 * 512) #define CLKOUT_RATE (44100 * 256) -static int lowland_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 *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops lowland_ops = { - .hw_params = lowland_hw_params, -}; - static struct snd_soc_jack lowland_headset; /* Headset jack detection DAPM pins */ @@ -109,7 +82,8 @@ static struct snd_soc_dai_link lowland_dai[] = { .codec_dai_name = "wm5100-aif1", .platform_name = "samsung-audio", .codec_name = "wm5100.1-001a", - .ops = &lowland_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = lowland_wm5100_init, }, { @@ -118,7 +92,8 @@ static struct snd_soc_dai_link lowland_dai[] = { .cpu_dai_name = "wm5100-aif2", .codec_dai_name = "wm1250-ev1", .codec_name = "wm1250-ev1.1-0027", - .ops = &lowland_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ignore_suspend = 1, }, }; -- 1.7.10

1 0

[alsa-devel] Strange Problem with snd-hda-intel
by Frank Maik Kittelmann 08 May '12

08 May '12

We have some strange behavior in our audio application using the ALC 888 hardware in ubuntu 10.10 with the hda-intel driver . Basically the main problem seems to be a buffer or memory management issue. Once a playback is stopped the very last sample is repeated forever, independently from sampling rate or sound file format. This sample leads to a constant DC voltage after the DA converters. We already tried the following steps: 1.) we changed a line in hda_intel.c to reset the fifo --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1797,7 +1797,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); - azx_sd_writel(azx_dev, SD_CTL, 0); + azx_stream_reset(chip, azx_dev); azx_dev->bufsize = 0; azx_dev->period_bytes = 0; azx_dev->format_val = 0; 2.) we set 2 parameters in the .config file before kernel compiling CONFIG_SND_HDA_POWER_SAVE=y CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0 Both solutions didn´t work for us - still same problem. Has anyone another idea how to solve this? Would be great if we could fix this. Frank

1 0

[alsa-devel] [PATCH 0/1] hdsp - Provide ioctl_compat
by Andre Schramm 08 May '12

08 May '12

Hi, short patch to provide ioctl_compat for HDSP cards as well. Quick test with SNDRV_HDSPM_IOCTL_GET_PEAK_RMS worked. Please review and apply. Thanks! Cheers, Andre

3 4

[alsa-devel] Issue with RME HDSPe AIO latency
by jamesm 07 May '12

07 May '12

Hello all, I'm using the RME hdspm kernel module with a Hammerfall HDSPe AIO card. I'm on RHEL 5.5, with alsa 1.0.17, kernel 2.6.18-194.11.4. I have backported the latest alsa hdspm code to this kernel, as there were a number of bugs in the one that ships with the 2.6.18-194 kernel. I've also implemented support for copy() and silence() functions so that the hdspm driver can be used with applications via the RDWR interface instead of just MMAP (ie it just works with aplay without using plughw first.) The latest hdspm driver code from alsa and kernel.org indicates that the HDSPe AIO card has a fixed buffer size of 16384. I've chosen a period size of 1024, which gives me 16 periods. The issue I have is that whenever I play audio through the hdspm driver, the delay between the hw pointer and sw pointer is always close to the maximum buffer size (approx 16000). This leaves me with a large delay between writing data into the PCM and it coming out of the card. I'd like this to be between 1024 and 2048, I could probably crank the period down further to 512. My application sets the start_threshold to 0 and avail_min to one period (1024). It then runs a thread, constantly writing data to the PCM (silence or real data). Does anyone have any suggestions as to why the hdspm driver hw pointer is not following the application pointer more closely? Is it due to the number of periods? Unfortunately I can't change that as the AIO buffer size is fixed. Here are my pcm hwparams, swparams and status from /proc/asound when my app is running: $ cat /proc/asound/card0/pcm0p/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: RME AIO name: RME AIO subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 $ cat /proc/asound/card0/pcm0p/sub0/hw_params access: RW_NONINTERLEAVED format: S32_LE subformat: STD channels: 16 rate: 48000 (48000/1) period_size: 1024 buffer_size: 16384 tick_time: 1000 $ cat /proc/asound/card0/pcm0p/sub0/sw_params tstamp_mode: NONE period_step: 1 sleep_min: 0 avail_min: 1024 xfer_align: 1 start_threshold: 0 stop_threshold: 18446744073709551615 silence_threshold: 0 silence_size: 0 boundary: 4611686018427387904 $ cat /proc/asound/card0/pcm0p/sub0/status state: RUNNING trigger_time: 1335799965.972782000 tstamp : 1335801526.687332000 delay : 15664 avail : 720 avail_max : 1056 ----- hw_ptr : 74911440 appl_ptr : 74927104 Regards James Milne FilmLight Ltd.

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[alsa-devel] [PATCH] ALSA: jack: Update documention to reflect other userspace interfaces
by Mark Brown 07 May '12

07 May '12

Since this is a generic API which should support any userspace interface for reporting jacks update the documentation a little to make that a bit clearer. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/core/jack.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/jack.c b/sound/core/jack.c index 471e1e3..a06b165 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -155,7 +155,7 @@ EXPORT_SYMBOL(snd_jack_new); * @jack: The jack to configure * @parent: The device to set as parent for the jack. * - * Set the parent for the jack input device in the device tree. This + * Set the parent for the jack devices in the device tree. This * function is only valid prior to registration of the jack. If no * parent is configured then the parent device will be the sound card. */ @@ -179,6 +179,9 @@ EXPORT_SYMBOL(snd_jack_set_parent); * mapping is provided but keys are enabled in the jack type then * BTN_n numeric buttons will be reported. * + * If jacks are not reporting via the input API this call will have no + * effect. + * * Note that this is intended to be use by simple devices with small * numbers of keys that can be reported. It is also possible to * access the input device directly - devices with complex input -- 1.7.10

2 1

[alsa-devel] [PATCH RESEND 0/9] Enable pinctrl support for mach-mxs
by Shawn Guo 07 May '12

07 May '12

[Resend to have subsystem lists Cc-ed] With pinctrl-mxs driver (DT only) applied on pinctrl tree, the mxs device tree conversion can start basing on that support. This series adopts pinctrl support for mxs device drivers with a dummy pinctrl state provided for non-DT boot, so that the pinctrl call in the device drivers will be bypassed for non-DT probe while it starts working for DT probe. To ease the merge process, I would like to ask Arnd and Olof to pull pinctrl tree as a dependency in arm-soc and have this series go through arm-soc. Regards, Shawn Shawn Guo (9): ARM: mxs: enable pinctrl dummy states serial: amba-pl011: adopt pinctrl support serial: mxs-auart: adopt pinctrl support mmc: mxs-mmc: adopt pinctrl support mtd: nand: gpmi: adopt pinctrl support i2c: mxs: adopt pinctrl support ASoC: mxs-saif: adopt pinctrl support video: mxsfb: adopt pinctrl support ARM: mxs: enable pinctrl support arch/arm/Kconfig | 1 + arch/arm/mach-mxs/Kconfig | 2 ++ arch/arm/mach-mxs/include/mach/common.h | 2 ++ arch/arm/mach-mxs/mach-apx4devkit.c | 2 ++ arch/arm/mach-mxs/mach-m28evk.c | 2 ++ arch/arm/mach-mxs/mach-mx23evk.c | 2 ++ arch/arm/mach-mxs/mach-mx28evk.c | 2 ++ arch/arm/mach-mxs/mach-stmp378x_devb.c | 2 ++ arch/arm/mach-mxs/mach-tx28.c | 2 ++ arch/arm/mach-mxs/mm.c | 11 +++++++++++ drivers/i2c/busses/i2c-mxs.c | 6 ++++++ drivers/mmc/host/mxs-mmc.c | 8 ++++++++ drivers/mtd/nand/gpmi-nand/gpmi-nand.c | 9 +++++++++ drivers/tty/serial/amba-pl011.c | 8 ++++++++ drivers/tty/serial/mxs-auart.c | 8 ++++++++ drivers/video/mxsfb.c | 9 +++++++++ sound/soc/mxs/mxs-saif.c | 8 ++++++++ 17 files changed, 84 insertions(+), 0 deletions(-) -- 1.7.5.4

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Re: [alsa-devel] [BUG] Internal CD input missing since linux-3.4
by Takashi Iwai 07 May '12

07 May '12

At Tue, 1 May 2012 12:23:00 +0200, Philipp Matthias Hahn wrote: > > Hello, > > I upgraded my desktop to 3.3.4 last weekend and noticed that the > internal CD connector is no longer available for recording. Instead I > see some new controls about "headphone volume", "rear microphone" and > "auto-mute". But I need the CD input since it's connected to my Analog > TV card, which MythTV needs for recording. > > Using "git bisect" showed the following patch to be doing the switch: > > 31ef22579302ac42054bebecb528710f46580925 is the first bad commit > > commit 31ef22579302ac42054bebecb528710f46580925 > > Author: Takashi Iwai <tiwai(a)suse.de> > > Date: Thu Dec 1 17:41:36 2011 +0100 > > > > ALSA: hda - Integrate input-jack stuff into kctl-jack This looks pretty irrelevant, maybe the bisection went wrong. Basically the problem is that your BIOS doesn't expose the pin for the CD input as an available pin. In alsa-info.sh output, you can see the pin 0x1c is shown as N/A. We can override a pin config value manually in such a case. See HD-Audio.txt in Documentation/sound/alsa. In anyway, I'll try to prepare a test patch in the next week. thanks, Takashi

1 1

[alsa-devel] [PATCH] pci/echoaudio: Remove incorrect part of assertion
by Mark Hills 06 May '12

06 May '12

This assertion seems to imply that chip->dsp_code_to_load is a pointer. It's actually an integer handle on the actual firmware, and 0 has no special meaning. The assertion prevents initialisation of a Darla20 card, but would also affect other models. It seems it was introduced in commit dd7b254d. ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting... ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000 ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware... ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20 ------------[ cut here ]------------ WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]() Hardware name: Dell DM051 BUG? (!chip->dsp_code_to_load || !chip->comm_page) Signed-off-by: Mark Hills <mark(a)pogo.org.uk> --- sound/pci/echoaudio/echoaudio_dsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 64417a7..d8c670c 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -475,7 +475,7 @@ static int load_firmware(struct echoaudio *chip) const struct firmware *fw; int box_type, err; - if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page)) + if (snd_BUG_ON(!chip->comm_page)) return -EPERM; /* See if the ASIC is present and working - only if the DSP is already loaded */ -- 1.7.4.4

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Re: [alsa-devel] [ALSA hda 3.4-rc] suspend2ram regression in snd_hda_intel
by Konstantin Khlebnikov 06 May '12

06 May '12

Konstantin Khlebnikov wrote: > Mainline commit 785f857d1cb0856b612b46a0545b74aa2596e44a > ("ALSA: hda - Set codec to D3 forcibly even if not used") > breaks sound for my notebook (thinkpad x220), > looks like codec cannot wake from power-save state after resume. > After reverting looks like all works fine. > > how to reproduce: > * unplug the power cord (to enable hda power safe mode) > * play something and stop > * wait until the background noise disappears (codec went into power-save mode) > * suspend > * resume > * play something again and hear the silence > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ Bump BTW, I don't see any locks around hda power-save code, for example hda_power_work() reads codec->power_on and codec->power_count without any locks. This seems racy.

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[alsa-devel] Call for propsals: Audio micro-conf at LPC 2012 San Diego
by Takashi Iwai 05 May '12

05 May '12

Hi, we're planning to hold an Audio micro-conference again this year at Linux Plumbers conference (LPC). It'll be held at San Diego, CA, Aug. 29-31. The Wiki page is found at http://wiki.linuxplumbersconf.org/2012:audio The items listed there are rough ideas popping up during a mail thread. If you have interest and want to give a talk or discussions, don't hesitate to submit your proposal. Please mail to submissions(a)linuxplumbersconf.org, as described in the page below: http://www.linuxplumbersconf.org/2012/2012-lpc-call-for-proposals-take-2/ The deadline for submission is 11:59 PDT on May 1, 2012. So, it's pretty little time left. Sorry for the late announce... thanks, Takashi

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Re: [alsa-devel] [GIT PULL] ASoC fix for 3.4
by Takashi Iwai 05 May '12

05 May '12

At Fri, 04 May 2012 12:24:58 +0100, Liam Girdwood wrote: > > Hi Takashi, > > For your final sound 3.4 fixes pull request. > > Thanks > > Liam > > --- > > The following changes since commit 1a38336b8611a04f0a624330c1f815421f4bf5f4: > > ASoC: wm8994: Improve sequencing of AIF channel enables (2012-04-23 12:55:52 +0100) > > are available in the git repository at: > > git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc.git for-3.4 > > for you to fetch changes up to fad9365bcc2a69ae16adc092e8ac192354980665: > > ASoC: omap-pcm: Free dma buffers in case of error. (2012-05-04 12:09:28 +0100) Thanks, pulled. I'm going to send a pull request to Linus later today. Takashi > ---------------------------------------------------------------- > Oleg Matcovschi (1): > ASoC: omap-pcm: Free dma buffers in case of error. > > sound/soc/omap/omap-pcm.c | 4 ++++ > 1 file changed, 4 insertions(+) > >

1 0

[alsa-devel] ASoC updates for 3.4
by Mark Brown 05 May '12

05 May '12

The following changes since commit 1a38336b8611a04f0a624330c1f815421f4bf5f4: ASoC: wm8994: Improve sequencing of AIF channel enables (2012-04-23 12:55:52 +0100) 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 06412088ce98f745405b8f65cfc51ddd6b842bbf: ASoC: s3c2412-i2s: Fix dai registration (2012-04-30 23:45:25 +0100) ---------------------------------------------------------------- ASoC: Updates for 3.4 Nothing terribly exciting here, a bunch of small and simple fixes scattered around the place. ---------------------------------------------------------------- Eric Bénard (1): ASoC: tlv312aic23: unbreak resume Fabio Estevam (1): ASoC: dt: sgtl5000.txt: Add description for 'reg' field Heiko Stübner (1): ASoC: s3c2412-i2s: Fix dai registration Lars-Peter Clausen (1): ASoC: bf5xx-ssm2602: Set DAI format Mark Brown (2): ASoC: wm_hubs: Make sure we don't disable differential line outputs ASoC: wm8350: Don't use locally allocated codec struct Richard Zhao (1): ASoC: core: check of_property_count_strings failure Documentation/devicetree/bindings/sound/sgtl5000.txt | 2 ++ sound/soc/blackfin/bf5xx-ssm2602.c | 2 ++ sound/soc/codecs/tlv320aic23.c | 4 ++-- sound/soc/codecs/wm8350.c | 11 ++++++----- sound/soc/codecs/wm_hubs.c | 15 +++++++++------ sound/soc/samsung/s3c2412-i2s.c | 2 +- sound/soc/soc-core.c | 6 +++--- 7 files changed, 25 insertions(+), 17 deletions(-)

2 1

[alsa-devel] Diagnosing why DAPM config is not correctly applied
by Daniel Drake 04 May '12

04 May '12

Hi, I'm working with Linux 3.3 and the soc/codecs/rt5631.c driver in an OLPC XO-1.75 laptop. We are only getting silence when playing back sound. I have added in some debug messages in soc-io.c and enabled DEBUG and VERBOSE_DEBUG in soc-dapm.c. Here are boot messages: http://dev.laptop.org/~dsd/20120504/boot.log At time X in the above log, I start "speaker-test -c 2 -t wav" and leave it running for the remainder of this test. Here is the "amixer" output at this point: http://dev.laptop.org/~dsd/20120504/amixer-initial.txt speaker-test seems happy but no audio can be heard :( However, I have found that the following tweak to the sound controls will fix the left channel: # amixer sset 'Left SPKVOL Mux' 'Vmid' # amixer sset 'Left SPKVOL Mux' 'SPKMIXL' Note that the original value was SPKMIXL (see initial amixer output) so this should have had no effect. However, the result is that sound is now heard through the left channel. (A very similar sound control change will fix the right channel too) While running these commands, the following kernel messages appear: (amixer sset 'Left SPKVOL Mux' 'Vmid') read c = 0 read c = 0 read 10 = 606 read 10 = 606 read c = 0 read c = 0 read a = 0 read a = 0 read 6 = a080 read 1e = 88c0 read 1e = 88c0 read 1e = 88c0 read 1e = 88c0 read 6 = a080 read 20 = 88c0 read 20 = 88c0 read 20 = 88c0 read 20 = 88c0 read 24 = 3000 read 24 = 3000 read 24 = 3000 read 24 = 3000 read 4 = c0c0 read 4 = c0c0 read 4 = c0c0 read 4 = c0c0 read 4 = c0c0 read 4 = c0c0 read 2c = 0 read 2c = 0 read 4 = c0c0 read 6 = a080 read 2 = 4040 read 22 = 7700 read 22 = 7700 read e = 0 read 22 = 7700 read 22 = 7700 read e = 0 read 2c = 0 read 6 = a080 read 26 = 8404 read 26 = 8404 read 26 = 8404 read 2a = 6c00 read 2a = 6c00 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 2 = 4040 read 2 = 4040 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 4 = c0c0 read 6 = a080 read 2 = 4040 read 40 = 4e00 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 2c = 0 read 2a = 6c00 read 2a = 6c00 read 2c = 0 read 2a = 6c00 read 2a = 6c00 read 2 = 4040 read 2 = 4040 read 2 = 4040 read 2 = 4040 read 2 = 4040 read 2 = 4040 read 2 = 4040 read 2 = 4040 rt5631 0-001a: Marking Vmid dirty due to mux connection rt5631 0-001a: Marking Left SPKVOL Mux dirty due to mux change read 2 = 4040 write 2 = 40 (cached) read 2 = 40 (amixer sset 'Left SPKVOL Mux' 'SPKMIXL') read c = 0 read c = 0 read 10 = 606 read 10 = 606 read c = 0 read c = 0 read a = 0 read a = 0 read 6 = a080 read 1e = 88c0 read 1e = 88c0 read 1e = 88c0 read 1e = 88c0 read 6 = a080 read 20 = 88c0 read 20 = 88c0 read 20 = 88c0 read 20 = 88c0 read 24 = 3000 read 24 = 3000 read 24 = 3000 read 24 = 3000 read 4 = c0c0 read 4 = c0c0 read 4 = c0c0 read 4 = c0c0 read 4 = c0c0 read 4 = c0c0 read 2c = 0 read 2c = 0 read 4 = c0c0 read 6 = a080 read 2 = 40 read 22 = 7700 read 22 = 7700 read e = 0 read 22 = 7700 read 22 = 7700 read e = 0 read 2c = 0 read 6 = a080 read 26 = 8404 read 26 = 8404 read 26 = 8404 read 2a = 6c00 read 2a = 6c00 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1a = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 1c = dfc0 read 2 = 40 read 2 = 40 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 14 = b0b0 read 4 = c0c0 read 6 = a080 read 2 = 40 read 40 = 4e00 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 28 = d8d8 read 2c = 0 read 2a = 6c00 read 2a = 6c00 read 2c = 0 read 2a = 6c00 read 2a = 6c00 read 2 = 40 read 2 = 40 read 2 = 40 read 2 = 40 read 2 = 40 read 2 = 40 read 2 = 40 read 2 = 40 rt5631 0-001a: Marking Vmid dirty due to mux disconnection rt5631 0-001a: Marking SPKMIXL Mixer dirty due to mux connection rt5631 0-001a: Marking Left SPKVOL Mux dirty due to mux change rt5631 0-001a: Marking Left DAC dirty due to peer state change rt5631 0-001a: Marking Left DAC To Mixer dirty due to peer state change rt5631 0-001a: Marking DAC REF dirty due to peer state change rt5631 0-001a: Marking I2S dirty due to peer state change rt5631 0-001a: Marking PLL1 dirty due to peer state change rt5631 0-001a: Marking Left DAC_HP dirty due to peer state change rt5631 0-001a: Marking OUTMIXL Mixer dirty due to peer state change read 1a = dfc0 read 28 = d8d8 read 42 = 0 read 42 = 0 read 42 = 0 read 42 = 0 read 2 = 40 write 2 = 4040 (cached) read 3a = 1000 write 3a = 90c0 (cached) read 3a = 90c0 write 3a = 92c0 (cached) read 3b = c write 3b = 200c (cached) read 2 = 4040 So, changing these settings in a way that should have no effect seems to kick DAPM into applying quite a different config from the one initially applied. In particular, the register value register 0x3a (RT5631_PWR_MANAG_ADD1) is quite different from the one applied earlier, and I think this is the key to having playback working. I've spent some time trying to get my head around the dapm code and see whats happening here, without much luck. If someone more experienced here could take a quick look at these messages and the dapm routes in rt5631.c for anything obvious, it would be much appreciated. Further debugging tips welcome too. Thanks, Daniel

1 0

[alsa-devel] ALSA driver issue with the ssm2602.c
by Bishop, Mark 04 May '12

04 May '12

When I enable the 'Output Mixer HiFi' using amixer it also powers up the Mic input on this codec. I believe this is incorrect. I would like to modify the driver to fix it but am having issues with the ALSA structures in the driver. Would I be off if I attempt to make my changes in the snd_soc_dapm_route structures? I am unsure how everything is hooked together in all those structs{}.

2 2

[alsa-devel] [PATCH v2] Provide interface for audio support in DSS
by Ricardo Neri 04 May '12

04 May '12

Hi, This patch is proposed as a result of a previous discussion related to include audio support in the DSS device driver (http://www.spinics.net/lists/linux-omap/msg64748.html) At the moment, this is intended to cover display technologies with audio based on CEA-861 and IEC-60958, such as HDMI and DisplayPort. However, provisions are made to make it extensible (see below). These are the changes with respect to v1: * The audio_enable and audio_start functions were split into enable/disable and start/stop respectively to improve readability. * Comments were added on whether the DSS audio functions may sleep. * In order to make the audio configuration parameters extensible, a struct omap_dss_audio is defined. This helps to easily add more config parameters in the future if/when required (e.g., speaker order configuration). BR, Ricardo Ricardo Neri (1): OMAPDSS: Provide an interface for audio support in DSS Documentation/arm/OMAP/DSS | 38 ++++++++++++++++++++++++++++++++++++++ include/video/omapdss.h | 25 +++++++++++++++++++++++++ 2 files changed, 63 insertions(+), 0 deletions(-) -- 1.7.5.4

1 1

[alsa-devel] [PATCH] ASoC: da7210: Minor improvements and a bugfix
by Ashish Chavan 03 May '12

03 May '12

This patch improves playback quality for few sample rates like 8000 and 11025 Hz. This also fixes an issue observed during testing of pll slave mode. Due to the issue, on some rare occasions there was no sound output for first time playback after system boot, though all subsequent playbacks were fine. It was mainly because of the sequence in which SRM bit was enabled. 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 | 28 +++++++++++++++------------- 1 files changed, 15 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 10f52ff..af5db70 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -850,8 +850,19 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, if (da7210->mclk_rate && (da7210->mclk_rate != sysclk)) { /* PLL mode, disable PLL bypass */ snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0); + + if (!da7210->master) { + /* PLL slave mode, also enable SRM */ + snd_soc_update_bits(codec, DA7210_PLL, + (DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN), + (DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN)); + } } else { - /* PLL bypass mode, enable PLL bypass */ + /* PLL bypass mode, enable PLL bypass and Auto Detection */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_MCLK_DET_EN, + DA7210_MCLK_DET_EN); snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, DA7210_PLL_BYP); } @@ -1014,18 +1025,9 @@ static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_DIV_L_MASK, pll_div3); - if (da7210->master) { - /* In master mode, no need to enable SRM */ - snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, - DA7210_PLL_EN); - } else { - /* In slave mode, enable SRM and PLL */ - snd_soc_update_bits(codec, DA7210_PLL, - (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | - DA7210_MCLK_DET_EN), - (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | - DA7210_MCLK_DET_EN)); - } + /* Enable PLL */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); + /* Enable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, DA7210_SC_MST_EN); -- 1.7.1

2 1

Re: [alsa-devel] Fast Track Ultra latency and new streaming logic.
by Clemens Ladisch 03 May '12

03 May '12

Felix Homann wrote: > 2012/5/3 Clemens Ladisch <clemens(a)ladisch.de>: >> USB playback also adds further latency by queueing the packets; the >> number of packets can be seen under "URBs" in /proc/asound/card?/stream0. >> (There are 1000/8000 packets per second at full/high speed.) > > I'm getting "URBs = 2 [ 5 6 ]" for playback and "URBs = 8 [ 8 8 8 8 > 8 8 8 8 ]" for capture with the "old" streaming logic. What does it > mean? The queue length is 5+6 = 11 packets; at high speed, that's 11 / 8 kHz = 1.4 ms. Regards, Clemens

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[alsa-devel] Fast Track Ultra latency and new streaming logic.
by Grant Diffey 03 May '12

03 May '12

Daniel, In continuing my personal habit of looking gift horses in the mouth... I can't seem to get below 512frames in flight (2 256sample frames or 4 128 or 8 64's) without consistant overruns. I have a feeling that the pcm <-> urb layer is buffering until a urb is max urbsize from a quick look at the code does this seem like the right track? This leads to rather high latencies (22ms+) or is this conclusion bogus? Grant.

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Re: [alsa-devel] HDA driver without interrupt
by Clemens Ladisch 03 May '12

03 May '12

Fanchao Liu wrote: > It can transfer data via DMA,the problem is I don't know when to > update hw_ptr and other pointers without interrupt. Does the hardware have some other means to at least read the current position? > I set a timer and call snd_pcm_period_elapsed() every 50ms but get > underrun error. With a call to snd_pcm_period_elapsed(), the driver guarantees that the current position is at or after the next period boundary. Regards, Clemens

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[alsa-devel] HDA driver without interrupt
by liufc 03 May '12

03 May '12

Hi guys,I'm trying to implement an alsa driver based on the Intel HDA driver.However due to some hardware defects,I'm not able to use interrupt.So I wonder whether there is a way to do the job without the help of interrupt.Can anyone give me a suggestion?

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[alsa-devel] [PATCH 00/11] enable imx6q_sabrelite sgtl5000 audio support
by Richard Zhao 02 May '12

02 May '12

It's based on Dong Aisheng's early pinctrl patch and Sascha's v2 convert to common clk patch. You can also get at: https://github.com/riczhao/kernel-imx/tree/topics/audio Richard Zhao (11): dma: imx-sdma: make channel0 operations atomic ASoC: imx-sgtl5000: add of_node_put when probe fail. ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000 i2c: imx: add pinctrl support ARM: imx6q: move imx6q_sabrelite specific code to a dedicated function ARM: dts: imx6q-sabrelite: add ssi device ARM: dts: imx6q-sabrelite: add audmux device ASoC: imx-audmux: add pinctrl support ARM: imx6q: add ssi1 clk_lookup ARM: imx6q_sabrelite: clkdev_add cko1 for sgtl5000 ARM: dts: imx6q-sabrelite: add sound device imx6q-sabrelite-sgtl5000 arch/arm/boot/dts/imx6q-sabrelite.dts | 30 ++++++++++++++++ arch/arm/boot/dts/imx6q.dtsi | 46 +++++++++++++++++++++++-- arch/arm/mach-imx/clk-imx6q.c | 4 ++ arch/arm/mach-imx/mach-imx6q.c | 37 +++++++++++++++++++- drivers/dma/imx-sdma.c | 57 ++++++++++++++++++------------- drivers/i2c/busses/i2c-imx.c | 9 +++++ sound/soc/fsl/imx-audmux.c | 8 ++++ sound/soc/fsl/imx-sgtl5000.c | 61 +++++++++++++++++++++++++-------- 8 files changed, 208 insertions(+), 44 deletions(-) Thanks Richard -- 1.7.5.4

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[alsa-devel] [PATCH 1/2] ASoC: core: Allow DAIs to specify a base address
by Mark Brown 02 May '12

02 May '12

Devices with many DAIs are becoming more and more common, and generally the more modern devices have consistent register layouts between DAIs. Rather than have drivers open code lookups based on the DAI ID or cause uglification in UI by having register addresses for IDs provide a base address field they can use. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index c63985b..2704ab0 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -198,6 +198,7 @@ struct snd_soc_dai_driver { const char *name; unsigned int id; int ac97_control; + unsigned int base; /* DAI driver callbacks */ int (*probe)(struct snd_soc_dai *dai); -- 1.7.10

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[alsa-devel] [PATCH 8/8] ASoC: Ux500: Add machine-driver
by Ola Lilja 02 May '12

02 May '12

Add machine-driver for ST-Ericsson U8500 platform, including support for the AB8500-codec. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/ux500/Kconfig | 11 ++ sound/soc/ux500/Makefile | 3 + sound/soc/ux500/u8500.c | 118 +++++++++++++++++++++ sound/soc/ux500/ux500_ab8500.c | 225 ++++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/ux500_ab8500.h | 21 ++++ 5 files changed, 378 insertions(+), 0 deletions(-) create mode 100644 sound/soc/ux500/u8500.c create mode 100644 sound/soc/ux500/ux500_ab8500.c create mode 100644 sound/soc/ux500/ux500_ab8500.h diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 59da6ce..85031b0 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -21,3 +21,14 @@ config SND_SOC_UX500_PLAT_DMA help Say Y if you want to enable the Ux500 platform-driver. +config SND_SOC_UX500_AB8500 + tristate "Codec/Machine - AB8500 (MSA)" + depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500 + select SND_SOC_AB8500_CODEC + select SND_SOC_UX500_PLAT_MSP_I2S + select SND_SOC_UX500_PLAT_DMA + help + Select this to enable the Ux500 machine-driver + and the AB8500 codec-driver. It will also enable the platform-driver + Ux500. + diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 2e742b1..50b7c42 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -6,3 +6,6 @@ obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o snd-soc-ux500-plat-dma-objs := ux500_pcm.o obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o +snd-soc-ux500-mach-objs := u8500.o ux500_ab8500.o +obj-$(CONFIG_SND_SOC_UX500_AB8500) += snd-soc-ux500-mach.o + diff --git a/sound/soc/ux500/u8500.c b/sound/soc/ux500/u8500.c new file mode 100644 index 0000000..33348f7 --- /dev/null +++ b/sound/soc/ux500/u8500.c @@ -0,0 +1,118 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja (ola.o.lilja(a)stericsson.com) + * for ST-Ericsson. + * + * License terms: + * + * 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 <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/io.h> +#include <linux/spi/spi.h> + +#include <sound/soc.h> +#include <sound/initval.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" + +#ifdef CONFIG_SND_SOC_UX500_AB8500 +#include <ux500_ab8500.h> +#endif + +/* Define the whole U8500 soundcard, linking platform to the codec-drivers */ +struct snd_soc_dai_link u8500_dai_links[] = { + #ifdef CONFIG_SND_SOC_UX500_AB8500 + { + .name = "ab8500_0", + .stream_name = "ab8500_0", + .cpu_dai_name = "ux500-msp-i2s.1", + .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = ux500_ab8500_machine_init, + .ops = ux500_ab8500_ops, + }, + { + .name = "ab8500_1", + .stream_name = "ab8500_1", + .cpu_dai_name = "ux500-msp-i2s.3", + .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = NULL, + .ops = ux500_ab8500_ops, + }, + #endif +}; + +static struct snd_soc_card u8500_card = { + .name = "U8500-card", + .probe = NULL, + .dai_link = u8500_dai_links, + .num_links = ARRAY_SIZE(u8500_dai_links), +}; + + + + +static int __devinit u8500_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("%s: Enter.\n", __func__); + + u8500_card.dev = &pdev->dev; + + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", + __func__, u8500_card.name); + platform_set_drvdata(pdev, &u8500_card); + + snd_soc_card_set_drvdata(&u8500_card, NULL); + + dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n", + __func__, u8500_card.name, u8500_card.num_links); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n", + __func__, u8500_card.name, u8500_card.dai_link[0].name); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n", + __func__, u8500_card.name, u8500_card.dai_link[0].stream_name); + + ret = snd_soc_register_card(&u8500_card); + if (ret) + dev_err(&pdev->dev, "Error: snd_soc_register_card failed (%d)!\n", + ret); + + return ret; +} + +static int __devexit u8500_remove(struct platform_device *pdev) +{ + struct snd_soc_card *u8500_card = platform_get_drvdata(pdev); + + pr_debug("%s: Enter.\n", __func__); + + snd_soc_unregister_card(u8500_card); + platform_set_drvdata(pdev, NULL); + + return 0; +} + +static struct platform_driver snd_soc_u8500_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "snd-soc-u8500", + }, + .probe = u8500_probe, + .remove = __devexit_p(u8500_remove), +}; + +module_platform_driver(snd_soc_u8500_driver); + + diff --git a/sound/soc/ux500/ux500_ab8500.c b/sound/soc/ux500/ux500_ab8500.c new file mode 100644 index 0000000..e4b8c5a --- /dev/null +++ b/sound/soc/ux500/ux500_ab8500.c @@ -0,0 +1,225 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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/device.h> +#include <linux/io.h> + +#include <mach/hardware.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" +#include "../codecs/ab8500-codec.h" + +#define TX_SLOT_MONO 0x0008 +#define TX_SLOT_STEREO 0x000a +#define RX_SLOT_MONO 0x0001 +#define RX_SLOT_STEREO 0x0003 +#define TX_SLOT_8CH 0x00FF +#define RX_SLOT_8CH 0x00FF + +#define DEF_TX_SLOTS TX_SLOT_STEREO +#define DEF_RX_SLOTS RX_SLOT_MONO + +#define DRIVERMODE_NORMAL 0 +#define DRIVERMODE_CODEC_ONLY 1 + +/* Slot configuration */ +static unsigned int tx_slots = DEF_TX_SLOTS; +static unsigned int rx_slots = DEF_RX_SLOTS; + +static struct snd_kcontrol_new ux500_ab8500_ctrls[] = { + SOC_DAPM_PIN_SWITCH("Headset Left"), + SOC_DAPM_PIN_SWITCH("Headset Right"), + SOC_DAPM_PIN_SWITCH("Earpiece"), + SOC_DAPM_PIN_SWITCH("Speaker Left"), + SOC_DAPM_PIN_SWITCH("Speaker Right"), + SOC_DAPM_PIN_SWITCH("LineOut Left"), + SOC_DAPM_PIN_SWITCH("LineOut Right"), + SOC_DAPM_PIN_SWITCH("Vibra 1"), + SOC_DAPM_PIN_SWITCH("Vibra 2"), + SOC_DAPM_PIN_SWITCH("Mic 1"), + SOC_DAPM_PIN_SWITCH("Mic 2"), + SOC_DAPM_PIN_SWITCH("LineIn Left"), + SOC_DAPM_PIN_SWITCH("LineIn Right"), + SOC_DAPM_PIN_SWITCH("DMic 1"), + SOC_DAPM_PIN_SWITCH("DMic 2"), + SOC_DAPM_PIN_SWITCH("DMic 3"), + SOC_DAPM_PIN_SWITCH("DMic 4"), + SOC_DAPM_PIN_SWITCH("DMic 5"), + SOC_DAPM_PIN_SWITCH("DMic 6"), +}; + +/* ASoC */ + +void ux500_ab8500_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + + dev_dbg(dev, "%s: Enter\n", __func__); + + /* Reset slots configuration to default(s) */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tx_slots = DEF_TX_SLOTS; + else + rx_slots = DEF_RX_SLOTS; +} + +int ux500_ab8500_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; + struct device *dev = rtd->card->dev; + unsigned int fmt, fmt_if1; + int channels, ret = 0, slots, slot_width, driver_mode; + bool is_playback; + + dev_dbg(dev, "%s: Enter\n", __func__); + + dev_dbg(dev, "%s: substream->pcm->name = %s\n" + "substream->pcm->id = %s.\n" + "substream->name = %s.\n" + "substream->number = %d.\n", + __func__, + substream->pcm->name, + substream->pcm->id, + substream->name, + substream->number); + + channels = params_channels(params); + + /* Setup codec depending on driver-mode */ + driver_mode = (channels == 8) ? + DRIVERMODE_CODEC_ONLY : DRIVERMODE_NORMAL; + dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__, + (driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY"); + + ab8500_audio_set_bit_delay(codec_dai, 1); + + if (driver_mode == DRIVERMODE_NORMAL) { + ab8500_audio_set_word_length(codec_dai, 16); + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CONT; + } else { + ab8500_audio_set_word_length(codec_dai, 20); + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_GATED; + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(dev, "%s: ERROR: snd_soc_dai_set_fmt failed " + "for codec_dai (ret = %d)!\n", __func__, ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + dev_err(dev, "%s: ERROR: snd_soc_dai_set_fmt failed " + "for cpu_dai (ret = %d)!\n", __func__, ret); + return ret; + } + + /* Setup TDM-slots */ + + is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + switch (channels) { + case 1: + slots = 16; + slot_width = 16; + tx_slots = (is_playback) ? TX_SLOT_MONO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_MONO; + break; + case 2: + slots = 16; + slot_width = 16; + tx_slots = (is_playback) ? TX_SLOT_STEREO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO; + break; + case 8: + slots = 16; + slot_width = 16; + tx_slots = (is_playback) ? TX_SLOT_8CH : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_8CH; + break; + default: + return -EINVAL; + } + + dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots, slot_width); + if (ret) + return ret; + + dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots, slot_width); + if (ret) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dev_dbg(dev, "%s: Setup IF1 for FM-radio.\n", __func__); + fmt_if1 = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_I2S; + ret = ab8500_audio_setup_if1(codec_dai->codec, fmt_if1, 16, 1); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_ops ux500_ab8500_ops[] = { + { + .hw_params = ux500_ab8500_hw_params, + .shutdown = ux500_ab8500_shutdown, + } +}; + +int ux500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct device *dev = rtd->card->dev; + int status; + + dev_dbg(dev, "%s Enter.\n", __func__); + + /* Add controls */ + status = snd_soc_add_codec_controls(codec, ux500_ab8500_ctrls, + ARRAY_SIZE(ux500_ab8500_ctrls)); + if (status < 0) { + pr_err("%s: failed to add machine-controls (%d).\n", + __func__, status); + return status; + } + + status = snd_soc_dapm_enable_pin(&codec->dapm, "Headset Left"); + status |= snd_soc_dapm_enable_pin(&codec->dapm, "Headset Right"); + + return status; +} + diff --git a/sound/soc/ux500/ux500_ab8500.h b/sound/soc/ux500/ux500_ab8500.h new file mode 100644 index 0000000..76ec1e6 --- /dev/null +++ b/sound/soc/ux500/ux500_ab8500.h @@ -0,0 +1,21 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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. + */ + +#ifndef UX500_AB8500_H +#define UX500_AB8500_H + +extern struct snd_soc_ops ux500_ab8500_ops[]; + +int ux500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime); + +#endif -- 1.7.8.3

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Re: [alsa-devel] ASoC: wm9712: Microphone doesn't work, "Capture Volume" inverted
by Mark Brown 02 May '12

02 May '12

On Tue, May 01, 2012 at 10:07:57PM +0200, Hans J. Koch wrote: > On Tue, May 01, 2012 at 06:12:07PM +0100, Mark Brown wrote: > > The driver knows exactly which registers are volatile > Then this is the first complex chip I've ever seen where the datasheet covers That's... horrifying, really. The WM9712 really isn't at all complex by modern standards for that matter, but the idea that it's common to have devices with documented registers that might spontaneously change in normal operation doesn't seem at all clever. What happens if the configuration that was set was important? > all corner cases. And what about things like dirty supply voltages or EMI? If you've got that sort of problem with supply quality or EMI there's obviously a very serious system issue to contend with, one that typically requires highly aggressive action to resolve - you're starting to get into games like having to constantly poll the device and restart it if you detect that it fell over which isn't pleasant and most likely means you've already hit serious user visible consequences. Things like device initialisation will need to be redone, for example. I can only think of one driver that takes such measures, and it's actually relying on restore from register cache for part of the recovery :) That said, many devices will get a moderate degree of robustness against this through power saving - the closer the device can get to cold during idle the better from this point of view, and in many cases it's possible to completely power down the device when idle. A common idiom is to take the device down to cold when idle then dump the setup changes that don't need sequencing into the device from cache before doing the bits that need sequencing, especially where many of the register settings can be mapped direictly into userspace. Indeed I'm not sure a cache isn't a bad plan for detecting this sort of thing; if you find the cache values diverging from the physical values for non-volatile registers you know something's up. > Just to save a few clock cycles every now and then you happily accept the > risk of the driver simply stop working after an unusual event, without > any chances of recovery. For the sort of devices we're talking about here it's way more than a few clock cycles that are involved, you're talking about fairly slow and at times congested buses like I2C here. Never mind what happens when someone decides to use a bigbanged controller, which is more common than one might dream of. Obviously if you're on a bus with trivial I/O costs the tradeoffs vary but for the sort of buses we're talking about it's much easier to see a win. Indeed with many older devices readback just isn't physically supported in the first place so if you want to do any kind of readback based thing you're going to have to cache in some way. This was true for the majority of devices at the time WM9712 was new. > Sound is not my usual playing ground, but at least I learned now why the > sound system on my desktop refuses to work after a few days uptime... I don't actually think desktop drivers bother with this much, though ICBW. They relatively infrequently use the slow buses. I'd suggest reporting the issue, anyway - it's probably more constructive. > Fortunately, most other subsystems would never accept such code for mainline. This is actually a standard feature of the regmap API that drivers can use if they want, there's a bunch of device classes where it can offer a useful win (those that typically have slow control interfaces and not so mnay volatile registers).

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[alsa-devel] [PATCH] ASoC: wm_hubs: Cache multiple DCS offsets
by Mark Brown 01 May '12

01 May '12

Rather than invalidating the cached DCS value every time the headphone gain changes store multiple values, indexed by gain. This allows the optimisation we get from the cache to take effect more often. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 74 ++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/wm_hubs.h | 3 +- 2 files changed, 67 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 8a29843..dfe957a 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -143,12 +143,69 @@ static bool wm_hubs_dac_hp_direct(struct snd_soc_codec *codec) return true; } +struct wm_hubs_dcs_cache { + struct list_head list; + unsigned int left; + unsigned int right; + u16 dcs_cfg; +}; + +static bool wm_hubs_dcs_cache_get(struct snd_soc_codec *codec, + struct wm_hubs_dcs_cache **entry) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; + unsigned int left, right; + + left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME); + left &= WM8993_HPOUT1L_VOL_MASK; + + right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME); + right &= WM8993_HPOUT1R_VOL_MASK; + + list_for_each_entry(cache, &hubs->dcs_cache, list) { + if (cache->left != left || cache->right != right) + continue; + + *entry = cache; + return true; + } + + return false; +} + +static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; + + if (hubs->no_cache_dac_hp_direct) + return; + + cache = devm_kzalloc(codec->dev, sizeof(*cache), GFP_KERNEL); + if (!cache) { + dev_err(codec->dev, "Failed to allocate DCS cache entry\n"); + return; + } + + cache->left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME); + cache->left &= WM8993_HPOUT1L_VOL_MASK; + + cache->right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME); + cache->right &= WM8993_HPOUT1R_VOL_MASK; + + cache->dcs_cfg = dcs_cfg; + + list_add_tail(&cache->list, &hubs->dcs_cache); +} + /* * Startup calibration of the DC servo */ static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; s8 offset; u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg; @@ -163,10 +220,11 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* If we're using a digital only path and have a previously * callibrated DC servo offset stored then use that. */ - if (wm_hubs_dac_hp_direct(codec) && hubs->dac_hp_direct_dcs) { - dev_dbg(codec->dev, "Using cached DC servo offset %x\n", - hubs->dac_hp_direct_dcs); - snd_soc_write(codec, dcs_reg, hubs->dac_hp_direct_dcs); + if (wm_hubs_dac_hp_direct(codec) && + wm_hubs_dcs_cache_get(codec, &cache)) { + dev_dbg(codec->dev, "Using cached DCS offset %x for %d,%d\n", + cache->dcs_cfg, cache->left, cache->right); + snd_soc_write(codec, dcs_reg, cache->dcs_cfg); wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); @@ -241,8 +299,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* Save the callibrated offset if we're in class W mode and * therefore don't have any analogue signal mixed in. */ - if (wm_hubs_dac_hp_direct(codec) && !hubs->no_cache_dac_hp_direct) - hubs->dac_hp_direct_dcs = dcs_cfg; + if (wm_hubs_dac_hp_direct(codec)) + wm_hubs_dcs_cache_set(codec, dcs_cfg); } /* @@ -257,9 +315,6 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, ret = snd_soc_put_volsw(kcontrol, ucontrol); - /* Updating the analogue gains invalidates the DC servo cache */ - hubs->dac_hp_direct_dcs = 0; - /* If we're applying an offset correction then updating the * callibration would be likely to introduce further offsets. */ if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update) @@ -1057,6 +1112,7 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; + INIT_LIST_HEAD(&hubs->dcs_cache); init_completion(&hubs->dcs_done); snd_soc_dapm_add_routes(dapm, analogue_routes, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 71861fc..da2dc89 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -16,6 +16,7 @@ #include <linux/completion.h> #include <linux/interrupt.h> +#include <linux/list.h> #include <sound/control.h> struct snd_soc_codec; @@ -32,7 +33,7 @@ struct wm_hubs_data { int no_series_update; bool no_cache_dac_hp_direct; - u16 dac_hp_direct_dcs; + struct list_head dcs_cache; bool (*check_class_w_digital)(struct snd_soc_codec *); bool lineout1_se; -- 1.7.10

1 0

Re: [alsa-devel] ASoC: wm9712: Microphone doesn't work, "Capture Volume" inverted
by Mark Brown 01 May '12

01 May '12

On Tue, May 01, 2012 at 01:44:58AM +0200, Hans J. Koch wrote: > On Mon, Apr 30, 2012 at 10:43:33PM +0100, Mark Brown wrote: > > I'm not entirely sure you've looked at the code here... > Oh yes, sorry. I got distracted by your mentioning of the suppressed writes. > Actually, we have suppressed reads. ac97_read() does a "return cache[reg];" > for most registers, assuming the hardware register cannot change after a > ac97_write(). That might or might not be true, to me it looks dangerous. The driver knows exactly which registers are volatile and takes advantage of this - this is entirely idiomatic for devices like this and not really that hard to get right.

1 0

[alsa-devel] davinci-mcasp - external MCLK, CPU generates WCLK and BCLK, slave codec(s), full duplex issues
by Jérôme Carretero 01 May '12

01 May '12

Hi, Note: I opened a thread on TI's E2E forums too, the thread is at: http://e2e.ti.com/support/dsp/omap_applications_processors/f/42/t/185810.as… Linux is v3.3. Repeating what I have put there: - As we will be using multiple codecs on McASP, we have put the codecs as slaves. We are also using a 8.192MHz external master clock. - We have patched davinci-mcasp.c to handle this mode ("SND_SOC_DAIFMT_EM_CBS_CFS"), and refer to it for the CPU DAI configuration in the machine driver. - Our setup currently has an AIC3X codec, connected to the TX clocks of McASP. The RX clocks are not used, they are left hanging, so we need to enable the TX clocks even when recording. - We only plan to support 32kHz, so currently we have left an ugly hack in the code. - It almost works, but we have a nasty problem: - When performing a capture alone, the sound is recorded as expected - When performing a playback alone, no problem - When starting a playback then a capture, no problem - When starting a capture then a playback, the playback will hang, the stream won't accept more data; no perturbation on the recording side. A review of the McASP code WRT/ SPRUH77A (OMAP-L138 Technical Reference Manual) does not really help. Now, I performed some tracing: - patched the davinci-mcasp.c code, to trace register accesses See attached file. The patch also contains the differences explained on the e2e forum post. - produced the "yes" file by running the "playback then capture" scenario: dmesg -c cat /dev/urandom | pv -B 3200 | aplay -f S16 -r 32000 aplay -B 20000 -f S16 -r 32000 -C -V mono /dev/null dmesg -c | gzip > yes.gz - produced the "no" file by running the "capture then playback" scenario: dmesg -c aplay -B 20000 -f S16 -r 32000 -C -V mono /dev/null cat /dev/urandom | pv -B 3200 | aplay -f S16 -r 32000 dmesg -c | gzip > no.gz And I still have issues seing what could be wrong. Could you somehow help me ? Best regards, -- cJ PS: sorry for the resend, the initial message exceeded the allowed mailing list message size.

1 0

[alsa-devel] [PATCH] s3c2412-i2s: Fix dai registration
by Heiko Stübner 01 May '12

01 May '12

As s3c2412-i2s is using the s3c_i2sv2 it should call the more specialised s3c_i2sv2_register_dai instead of simply calling snd_soc_register_dai. Without this call the snd_soc_dai_ops structure isn't initialised correctly. Signed-off-by: Heiko Stuebner <heiko(a)sntech.de> --- sound/soc/samsung/s3c2412-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 7218507..79fbeea 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -166,7 +166,7 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { static __devinit int s3c2412_iis_dev_probe(struct platform_device *pdev) { - return snd_soc_register_dai(&pdev->dev, &s3c2412_i2s_dai); + return s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai); } static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev) -- 1.7.5.4

2 2

[alsa-devel] [PATCH 1/4] ASoC: wm8350: Don't use locally allocated codec struct
by Mark Brown 30 Apr '12

30 Apr '12

The core allocates the live copies, we shouldn't try to duplicate it and were buggy trying to do so as we were using uninitialised data for the control data. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8350.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 8c4c959..aa12c6b 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -60,7 +60,7 @@ struct wm8350_jack_data { }; struct wm8350_data { - struct snd_soc_codec codec; + struct wm8350 *wm8350; struct wm8350_output out1; struct wm8350_output out2; struct wm8350_jack_data hpl; @@ -1309,7 +1309,7 @@ static void wm8350_hp_work(struct wm8350_data *priv, struct wm8350_jack_data *jack, u16 mask) { - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report; @@ -1342,7 +1342,7 @@ static void wm8350_hpr_work(struct work_struct *work) static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; struct wm8350_jack_data *jack = NULL; switch (irq - wm8350->irq_base) { @@ -1427,7 +1427,7 @@ EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect); static irqreturn_t wm8350_mic_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report = 0; @@ -1536,6 +1536,8 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); + priv->wm8350 = wm8350; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) priv->supplies[i].supply = supply_names[i]; @@ -1544,7 +1546,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - wm8350->codec.codec = codec; codec->control_data = wm8350; /* Put the codec into reset if it wasn't already */ -- 1.7.10

1 3

Re: [alsa-devel] ASoC: wm9712: Microphone doesn't work, "Capture Volume" inverted
by Mark Brown 30 Apr '12

30 Apr '12

On Mon, Apr 30, 2012 at 09:40:55PM +0200, Hans J. Koch wrote: > On Thu, Apr 26, 2012 at 10:37:45PM +0100, Mark Brown wrote: > > On Thu, Apr 26, 2012 at 11:11:33PM +0200, Christoph Fritz wrote: > > > ac97_write() doesn't get called when I try to change "Item: Differential Source". > > Oh, right - you're not actually seeing any attempt to write. There > > won't be anything on the bus, then. Still, why are you identifying the > > shift there? The write will be suppresed if the register contents don't > > change which looks like what you're seeing here - though I can't see > [...] > I overlooked that one. What kind of horror show is that? It's definetly wrong > to make any assumptions about hardware register contents. A low-level write > function always has to write the given value to the given register. Everything > else, like suppressing a write because you _think_ a register hasn't changed, > will sooner or later lead to hard-to-find errors. > Any performance optimizations have to take place on the highest possible level. I'm not entirely sure you've looked at the code here...

1 0

[alsa-devel] [PATCH] ASoC: cs42l52: Remove duplicate module exit code
by Mark Brown 30 Apr '12

30 Apr '12

In the conversion to module_init_i2c() the original open coded module exit function was left. Remove it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs42l52.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index e7fefbd..703060d 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1287,13 +1287,6 @@ static struct i2c_driver cs42l52_i2c_driver = { .remove = __devexit_p(cs42l52_i2c_remove), }; -static void __exit cs42l52_exit(void) -{ - i2c_del_driver(&cs42l52_i2c_driver); -} - -module_exit(cs42l52_exit); - module_i2c_driver(cs42l52_i2c_driver); MODULE_DESCRIPTION("ASoC CS42L52 driver"); -- 1.7.10

2 1

[alsa-devel] [PATCH] ASoC: core: Fix dai_link dereference.
by Liam Girdwood 30 Apr '12

30 Apr '12

We should check dailess before dereferencing. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9bdfef4..7f1e012 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1218,7 +1218,7 @@ static int soc_post_component_init(struct snd_soc_card *card, #ifdef CONFIG_DEBUG_FS /* add DPCM sysfs entries */ - if (!dai_link->dynamic) + if (!dailess && !dai_link->dynamic) goto out; ret = soc_dpcm_debugfs_add(rtd); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ASoC: Add support for CS42L52 Codec
by Brian Austin 30 Apr '12

30 Apr '12

This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec v2 Changes: Fixes for regcache volatile and CHIPID Add platform data structures for MIC, Charge Pump, MICBIAS levels Add input support for Beep Generator Move L/R Swap into DAPM Use DAPM Routes to hook AIF widgets to DAI's Fix SND_SOC_BIAS cases Use regmap_patch for magic writes at probe Use devm_regmap_init_i2c() Use module_i2c_driver() Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Georgi Vlaev <joe(a)nucleusys.com> --- include/sound/cs42l52.h | 36 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l52.c | 1302 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l52.h | 274 ++++++++++ 5 files changed, 1618 insertions(+), 0 deletions(-) create mode 100644 include/sound/cs42l52.h create mode 100644 sound/soc/codecs/cs42l52.c create mode 100644 sound/soc/codecs/cs42l52.h diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h new file mode 100644 index 0000000..4c68955 --- /dev/null +++ b/include/sound/cs42l52.h @@ -0,0 +1,36 @@ +/* + * linux/sound/cs42l52.h -- Platform data for CS42L52 + * + * Copyright (c) 2012 Cirrus Logic Inc. + * + * 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. + */ + +#ifndef __CS42L52_H +#define __CS42L52_H + +struct cs42l52_platform_data { + + /* MICBIAS Level. Check datasheet Pg48 */ + unsigned int micbias_lvl; + + /* MICA mode selection 0=Single 1=Differential */ + unsigned int mica_cfg; + + /* MICB mode selection 0=Single 1=Differential */ + unsigned int micb_cfg; + + /* MICA Select 0=MIC1A 1=MIC2A */ + unsigned int mica_sel; + + /* MICB Select 0=MIC2A 1=MIC2B */ + unsigned int micb_sel; + + /* Charge Pump Freq. Check datasheet Pg73 */ + unsigned int chgfreq; + +}; + +#endif /* __CS42L52_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2e51eb0..0235532 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L52 if I2C select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI @@ -183,6 +184,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L52 + tristate + config SND_SOC_CS42L73 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index db61c44..d11ab90 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o @@ -123,6 +124,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c new file mode 100644 index 0000000..257579d --- /dev/null +++ b/sound/soc/codecs/cs42l52.c @@ -0,0 +1,1302 @@ +/* + * cs42l52.c -- CS42L52 ALSA SoC audio driver + * + * Copyright 2012 CirrusLogic, Inc. + * + * Author: Georgi Vlaev <joe(a)nucleusys.com> + * Author: Brian Austin <brian.austin(a)cirrus.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/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs42l52.h> +#include "cs42l52.h" + +struct sp_config { + u8 spc, format, spfs; + u32 srate; +}; + +struct cs42l52_private { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct device *dev; + struct sp_config config; + struct cs42l52_platform_data pdata; + u32 sysclk; + u8 mclksel; + u32 mclk; + u8 flags; +#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) + struct input_dev *beep; + struct work_struct beep_work; + int beep_rate; +#endif +}; + +static const struct reg_default cs42l52_reg_defaults[] = { + { CS42L52_PWRCTL1, 0x9F }, /* r02 PWRCTL 1 */ + { CS42L52_PWRCTL2, 0x07 }, /* r03 PWRCTL 2 */ + { CS42L52_PWRCTL3, 0xFF }, /* r04 PWRCTL 3 */ + { CS42L52_CLK_CTL, 0xA0 }, /* r05 Clocking Ctl */ + { CS42L52_IFACE_CTL1, 0x00 }, /* r06 Interface Ctl 1 */ + { CS42L52_ADC_PGA_A, 0x80 }, /* r08 Input A Select */ + { CS42L52_ADC_PGA_B, 0x80 }, /* r09 Input B Select */ + { CS42L52_ANALOG_HPF_CTL, 0xA5 }, /* r0A Analog HPF Ctl */ + { CS42L52_ADC_HPF_FREQ, 0x00 }, /* r0B ADC HPF Corner Freq */ + { CS42L52_ADC_MISC_CTL, 0x00 }, /* r0C Misc. ADC Ctl */ + { CS42L52_PB_CTL1, 0x60 }, /* r0D Playback Ctl 1 */ + { CS42L52_MISC_CTL, 0x02 }, /* r0E Misc. Ctl */ + { CS42L52_PB_CTL2, 0x00 }, /* r0F Playback Ctl 2 */ + { CS42L52_MICA_CTL, 0x00 }, /* r10 MICA Amp Ctl */ + { CS42L52_MICB_CTL, 0x00 }, /* r11 MICB Amp Ctl */ + { CS42L52_PGAA_CTL, 0x00 }, /* r12 PGAA Vol, Misc. */ + { CS42L52_PGAB_CTL, 0x00 }, /* r13 PGAB Vol, Misc. */ + { CS42L52_PASSTHRUA_VOL, 0x00 }, /* r14 Bypass A Vol */ + { CS42L52_PASSTHRUB_VOL, 0x00 }, /* r15 Bypass B Vol */ + { CS42L52_ADCA_VOL, 0x00 }, /* r16 ADCA Volume */ + { CS42L52_ADCB_VOL, 0x00 }, /* r17 ADCB Volume */ + { CS42L52_ADCA_MIXER_VOL, 0x80 }, /* r18 ADCA Mixer Volume */ + { CS42L52_ADCB_MIXER_VOL, 0x80 }, /* r19 ADCB Mixer Volume */ + { CS42L52_PCMA_MIXER_VOL, 0x00 }, /* r1A PCMA Mixer Volume */ + { CS42L52_PCMB_MIXER_VOL, 0x00 }, /* r1B PCMB Mixer Volume */ + { CS42L52_BEEP_FREQ, 0x00 }, /* r1C Beep Freq on Time */ + { 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_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */ + { CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */ + { CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */ + { CS42L52_SPKA_VOL, 0x00 }, /* r24 Speaker A Volume */ + { CS42L52_SPKB_VOL, 0x00 }, /* r25 Speaker B Volume */ + { CS42L52_ADC_PCM_MIXER, 0x00 }, /* r26 Channel Mixer and Swap */ + { CS42L52_LIMITER_CTL1, 0x00 }, /* r27 Limit Ctl 1 Thresholds */ + { CS42L52_LIMITER_CTL2, 0x7F }, /* r28 Limit Ctl 2 Release Rate */ + { CS42L52_LIMITER_AT_RATE, 0xC0 }, /* r29 Limiter Attack Rate */ + { CS42L52_ALC_CTL, 0x00 }, /* r2A ALC Ctl 1 Attack Rate */ + { CS42L52_ALC_RATE, 0x3F }, /* r2B ALC Release Rate */ + { CS42L52_ALC_THRESHOLD, 0x3f }, /* r2C ALC Thresholds */ + { CS42L52_NOISE_GATE_CTL, 0x00 }, /* r2D Noise Gate Ctl */ + { CS42L52_CLK_STATUS, 0x00 }, /* r2E Overflow and Clock Status */ + { CS42L52_BATT_COMPEN, 0x00 }, /* r2F battery Compensation */ + { CS42L52_BATT_LEVEL, 0x00 }, /* r30 VP Battery Level */ + { CS42L52_SPK_STATUS, 0x00 }, /* r31 Speaker Status */ + { CS42L52_TEM_CTL, 0x3B }, /* r32 Temp Ctl */ + { CS42L52_THE_FOLDBACK, 0x00 }, /* r33 Foldback */ +}; + +static bool cs42l52_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L52_CHIP: + case CS42L52_PWRCTL1: + case CS42L52_PWRCTL2: + case CS42L52_PWRCTL3: + case CS42L52_CLK_CTL: + case CS42L52_IFACE_CTL1: + case CS42L52_IFACE_CTL2: + case CS42L52_ADC_PGA_A: + case CS42L52_ADC_PGA_B: + case CS42L52_ANALOG_HPF_CTL: + case CS42L52_ADC_HPF_FREQ: + case CS42L52_ADC_MISC_CTL: + case CS42L52_PB_CTL1: + case CS42L52_MISC_CTL: + case CS42L52_PB_CTL2: + case CS42L52_MICA_CTL: + case CS42L52_MICB_CTL: + case CS42L52_PGAA_CTL: + case CS42L52_PGAB_CTL: + case CS42L52_PASSTHRUA_VOL: + case CS42L52_PASSTHRUB_VOL: + case CS42L52_ADCA_VOL: + case CS42L52_ADCB_VOL: + case CS42L52_ADCA_MIXER_VOL: + case CS42L52_ADCB_MIXER_VOL: + case CS42L52_PCMA_MIXER_VOL: + case CS42L52_PCMB_MIXER_VOL: + case CS42L52_BEEP_FREQ: + case CS42L52_BEEP_VOL: + case CS42L52_BEEP_TONE_CTL: + case CS42L52_TONE_CTL: + case CS42L52_MASTERA_VOL: + case CS42L52_MASTERB_VOL: + case CS42L52_HPA_VOL: + case CS42L52_HPB_VOL: + case CS42L52_SPKA_VOL: + case CS42L52_SPKB_VOL: + case CS42L52_ADC_PCM_MIXER: + case CS42L52_LIMITER_CTL1: + case CS42L52_LIMITER_CTL2: + case CS42L52_LIMITER_AT_RATE: + case CS42L52_ALC_CTL: + case CS42L52_ALC_RATE: + case CS42L52_ALC_THRESHOLD: + case CS42L52_NOISE_GATE_CTL: + case CS42L52_CLK_STATUS: + case CS42L52_BATT_COMPEN: + case CS42L52_BATT_LEVEL: + case CS42L52_SPK_STATUS: + case CS42L52_TEM_CTL: + case CS42L52_THE_FOLDBACK: + case CS42L52_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool cs42l52_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L52_IFACE_CTL2: + case CS42L52_CLK_STATUS: + case CS42L52_BATT_LEVEL: + case CS42L52_SPK_STATUS: + case CS42L52_CHARGE_PUMP: + return 1; + default: + return 0; + } +} + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(hpd_tlv, -9600, 50, 1); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); + +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const char * const cs42l52_adca_text[] = { + "Input1A", "Input2A", "Input3A", "Input4A", "PGA Input Left"}; + +static const char * const cs42l52_adcb_text[] = { + "Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"}; + +static const struct soc_enum adca_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5, + ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text); + +static const struct soc_enum adcb_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5, + ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text); + +static const struct snd_kcontrol_new adca_mux = + SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum); + +static const struct snd_kcontrol_new adcb_mux = + SOC_DAPM_ENUM("Right ADC Input Capture Mux", adcb_enum); + +static const char * const mic_bias_level_text[] = { + "0.5 +VA", "0.6 +VA", "0.7 +VA", + "0.8 +VA", "0.83 +VA", "0.91 +VA" +}; + +static const struct soc_enum mic_bias_level_enum = + SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0, + ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); + +static const char * const cs42l52_mic_text[] = { "Single", "Differential" }; + +static const struct soc_enum mica_enum = + SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5, + ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); + +static const struct soc_enum micb_enum = + SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5, + ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); + +static const struct snd_kcontrol_new mica_mux = + SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum); + +static const struct snd_kcontrol_new micb_mux = + SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum); + +static const char * const digital_output_mux_text[] = {"ADC", "DSP"}; + +static const struct soc_enum digital_output_mux_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6, + ARRAY_SIZE(digital_output_mux_text), + digital_output_mux_text); + +static const struct snd_kcontrol_new digital_output_mux = + SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum); + +static const char * const hp_gain_num_text[] = { + "0.3959", "0.4571", "0.5111", "0.6047", + "0.7099", "0.8399", "1.000", "1.1430" +}; + +static const struct soc_enum hp_gain_enum = + SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4, + ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text); + +static const char * const beep_pitch_text[] = { + "C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5", + "C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7" +}; + +static const struct soc_enum beep_pitch_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4, + ARRAY_SIZE(beep_pitch_text), beep_pitch_text); + +static const char * const beep_ontime_text[] = { + "86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s", + "1.80 s", "2.20 s", "2.50 s", "2.80 s", "3.20 s", + "3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s" +}; + +static const struct soc_enum beep_ontime_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0, + ARRAY_SIZE(beep_ontime_text), beep_ontime_text); + +static const char * const beep_offtime_text[] = { + "1.23 s", "2.58 s", "3.90 s", "5.20 s", + "6.60 s", "8.05 s", "9.35 s", "10.80 s" +}; + +static const struct soc_enum beep_offtime_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5, + ARRAY_SIZE(beep_offtime_text), beep_offtime_text); + +static const char * const beep_config_text[] = { + "Off", "Single", "Multiple", "Continuous" +}; + +static const struct soc_enum beep_config_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6, + ARRAY_SIZE(beep_config_text), beep_config_text); + +static const char * const beep_bass_text[] = { + "50 Hz", "100 Hz", "200 Hz", "250 Hz" +}; + +static const struct soc_enum beep_bass_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1, + ARRAY_SIZE(beep_bass_text), beep_bass_text); + +static const char * const beep_treble_text[] = { + "5 kHz", "7 kHz", "10 kHz", " 15 kHz" +}; + +static const struct soc_enum beep_treble_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3, + ARRAY_SIZE(beep_treble_text), beep_treble_text); + +static const char * const ng_threshold_text[] = { + "-34dB", "-37dB", "-40dB", "-43dB", + "-46dB", "-52dB", "-58dB", "-64dB" +}; + +static const struct soc_enum ng_threshold_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2, + ARRAY_SIZE(ng_threshold_text), ng_threshold_text); + +static const char * const cs42l52_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms"}; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0, + ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text); + +static const char * const cs42l52_ng_type_text[] = { + "Apply Specific", "Apply All" +}; + +static const struct soc_enum ng_type_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6, + ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text); + +static const char * const left_swap_text[] = { + "Left", "LR 2", "Right"}; + +static const char * const right_swap_text[] = { + "Right", "LR 2", "Left"}; + +static const unsigned int swap_values[] = { 0, 1, 3 }; + +static const struct soc_enum adca_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, + ARRAY_SIZE(left_swap_text), + left_swap_text, + swap_values); + +static const struct snd_kcontrol_new adca_mixer = + SOC_DAPM_ENUM("Route", adca_swap_enum); + +static const struct soc_enum pcma_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1, + ARRAY_SIZE(left_swap_text), + left_swap_text, + swap_values); + +static const struct snd_kcontrol_new pcma_mixer = + SOC_DAPM_ENUM("Route", pcma_swap_enum); + +static const struct soc_enum adcb_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1, + ARRAY_SIZE(right_swap_text), + right_swap_text, + swap_values); + +static const struct snd_kcontrol_new adcb_mixer = + SOC_DAPM_ENUM("Route", adcb_swap_enum); + +static const struct soc_enum pcmb_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1, + ARRAY_SIZE(right_swap_text), + right_swap_text, + swap_values); + +static const struct snd_kcontrol_new pcmb_mixer = + SOC_DAPM_ENUM("Route", pcmb_swap_enum); + + +static const struct snd_kcontrol_new passthrul_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 6, 1, 0); + +static const struct snd_kcontrol_new passthrur_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 7, 1, 0); + +static const struct snd_kcontrol_new spkl_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new spkr_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new hpl_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new hpr_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 6, 1, 1); + +static const struct snd_kcontrol_new cs42l52_snd_controls[] = { + + SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L52_MASTERA_VOL, + CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv), + + SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, + CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv), + + SOC_ENUM("Headphone Analog Gain", hp_gain_enum), + + SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, + CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv), + + SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, + CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), + + SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), + + SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL, + CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv), + + SOC_ENUM("MIC Bias Level", mic_bias_level_enum), + + SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, + CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv), + SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", + CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, + 6, 0x7f, 0x19, ipd_tlv), + + SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), + + SOC_DOUBLE_R("ADC Mixer Switch", CS42L52_ADCA_MIXER_VOL, + CS42L52_ADCB_MIXER_VOL, 7, 1, 1), + + SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv), + + SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", + CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, + 6, 0x7f, 0x19, hl_tlv), + SOC_DOUBLE_R("PCM Mixer Switch", + CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), + + SOC_ENUM("Beep Config", beep_config_enum), + SOC_ENUM("Beep Pitch", beep_pitch_enum), + SOC_ENUM("Beep on Time", beep_ontime_enum), + SOC_ENUM("Beep off Time", beep_offtime_enum), + SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), + SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), + SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), + + SOC_SINGLE("Tone Control Switch", CS42L52_BEEP_TONE_CTL, 0, 1, 1), + SOC_SINGLE_TLV("Treble Gain Volume", + CS42L52_TONE_CTL, 4, 15, 1, hl_tlv), + SOC_SINGLE_TLV("Bass Gain Volume", + CS42L52_TONE_CTL, 0, 15, 1, hl_tlv), + + /* Limiter */ + SOC_SINGLE_TLV("Limiter Max Threshold Volume", + CS42L52_LIMITER_CTL1, 5, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Cushion Threshold Volume", + CS42L52_LIMITER_CTL1, 2, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Release Rate Volume", + CS42L52_LIMITER_CTL2, 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Attack Rate Volume", + CS42L52_LIMITER_AT_RATE, 0, 63, 0, limiter_tlv), + + SOC_SINGLE("Limiter SR Switch", CS42L52_LIMITER_CTL1, 1, 1, 0), + SOC_SINGLE("Limiter ZC Switch", CS42L52_LIMITER_CTL1, 0, 1, 0), + SOC_SINGLE("Limiter Switch", CS42L52_LIMITER_CTL2, 7, 1, 0), + + /* ALC */ + SOC_SINGLE_TLV("ALC Attack Rate Volume", CS42L52_ALC_CTL, + 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Release Rate Volume", CS42L52_ALC_RATE, + 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L52_ALC_THRESHOLD, + 5, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L52_ALC_THRESHOLD, + 2, 7, 0, limiter_tlv), + + SOC_DOUBLE_R("ALC SR Capture Switch", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 7, 1, 1), + SOC_DOUBLE_R("ALC ZC Capture Switch", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 6, 1, 1), + SOC_DOUBLE("ALC Capture Switch", CS42L52_ALC_CTL, 6, 7, 1, 0), + + /* Noise gate */ + SOC_ENUM("NG Type Switch", ng_type_enum), + SOC_SINGLE("NG Enable Switch", CS42L52_NOISE_GATE_CTL, 6, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L52_NOISE_GATE_CTL, 5, 1, 1), + SOC_ENUM("NG Threshold", ng_threshold_enum), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_DOUBLE("HPF Switch", CS42L52_ANALOG_HPF_CTL, 5, 7, 1, 0), + + SOC_DOUBLE("Analog SR Switch", CS42L52_ANALOG_HPF_CTL, 1, 3, 1, 1), + SOC_DOUBLE("Analog ZC Switch", CS42L52_ANALOG_HPF_CTL, 0, 2, 1, 1), + SOC_SINGLE("Digital SR Switch", CS42L52_MISC_CTL, 1, 1, 0), + SOC_SINGLE("Digital ZC Switch", CS42L52_MISC_CTL, 0, 1, 0), + SOC_SINGLE("Deemphasis Switch", CS42L52_MISC_CTL, 2, 1, 0), + + SOC_SINGLE("Batt Compensation Switch", CS42L52_BATT_COMPEN, 7, 1, 0), + SOC_SINGLE("Batt VP Monitor Switch", CS42L52_BATT_COMPEN, 6, 1, 0), + SOC_SINGLE("Batt VP ref", CS42L52_BATT_COMPEN, 0, 0x0f, 0), + + SOC_SINGLE("PGA AIN1L Switch", CS42L52_ADC_PGA_A, 0, 1, 0), + SOC_SINGLE("PGA AIN1R Switch", CS42L52_ADC_PGA_B, 0, 1, 0), + SOC_SINGLE("PGA AIN2L Switch", CS42L52_ADC_PGA_A, 1, 1, 0), + SOC_SINGLE("PGA AIN2R Switch", CS42L52_ADC_PGA_B, 1, 1, 0), + + SOC_SINGLE("PGA AIN3L Switch", CS42L52_ADC_PGA_A, 2, 1, 0), + SOC_SINGLE("PGA AIN3R Switch", CS42L52_ADC_PGA_B, 2, 1, 0), + + SOC_SINGLE("PGA AIN4L Switch", CS42L52_ADC_PGA_A, 3, 1, 0), + SOC_SINGLE("PGA AIN4R Switch", CS42L52_ADC_PGA_B, 3, 1, 0), + + SOC_SINGLE("PGA MICA Switch", CS42L52_ADC_PGA_A, 4, 1, 0), + SOC_SINGLE("PGA MICB Switch", CS42L52_ADC_PGA_B, 4, 1, 0), + +}; + +static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), + SND_SOC_DAPM_INPUT("AIN4L"), + SND_SOC_DAPM_INPUT("AIN4R"), + SND_SOC_DAPM_INPUT("MICA"), + SND_SOC_DAPM_INPUT("MICB"), + SND_SOC_DAPM_SIGGEN("Beep"), + + SND_SOC_DAPM_AIF_OUT("AIFOUTL", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFOUTR", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux), + SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1), + SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", CS42L52_PWRCTL1, 4, 1, NULL, 0), + + SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adca_mux), + SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcb_mux), + + SND_SOC_DAPM_MUX("ADC Left Swap", SND_SOC_NOPM, + 0, 0, &adca_mixer), + SND_SOC_DAPM_MUX("ADC Right Swap", SND_SOC_NOPM, + 0, 0, &adcb_mixer), + + SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, + 0, 0, &digital_output_mux), + + SND_SOC_DAPM_PGA("PGA MICA", CS42L52_PWRCTL2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("PGA MICB", CS42L52_PWRCTL2, 2, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", CS42L52_PWRCTL2, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Charge Pump", CS42L52_PWRCTL1, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_IN("AIFINL", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFINR", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SWITCH("Bypass Left", CS42L52_MISC_CTL, + 6, 0, &passthrul_ctl), + SND_SOC_DAPM_SWITCH("Bypass Right", CS42L52_MISC_CTL, + 7, 0, &passthrur_ctl), + + SND_SOC_DAPM_MUX("PCM Left Swap", SND_SOC_NOPM, + 0, 0, &pcma_mixer), + SND_SOC_DAPM_MUX("PCM Right Swap", SND_SOC_NOPM, + 0, 0, &pcmb_mixer), + + SND_SOC_DAPM_SWITCH("HP Left Amp", SND_SOC_NOPM, 0, 0, &hpl_ctl), + SND_SOC_DAPM_SWITCH("HP Right Amp", SND_SOC_NOPM, 0, 0, &hpr_ctl), + + SND_SOC_DAPM_SWITCH("SPK Left Amp", SND_SOC_NOPM, 0, 0, &spkl_ctl), + SND_SOC_DAPM_SWITCH("SPK Right Amp", SND_SOC_NOPM, 0, 0, &spkr_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("SPKOUTA"), + SND_SOC_DAPM_OUTPUT("SPKOUTB"), + +}; + +static const struct snd_soc_dapm_route cs42l52_audio_map[] = { + + {"Capture", NULL, "AIFOUTL"}, + {"Capture", NULL, "AIFOUTL"}, + + {"AIFOUTL", NULL, "Output Mux"}, + {"AIFOUTR", NULL, "Output Mux"}, + + {"Output Mux", "ADC", "ADC Left"}, + {"Output Mux", "ADC", "ADC Right"}, + + {"ADC Left", NULL, "Charge Pump"}, + {"ADC Right", NULL, "Charge Pump"}, + + {"Charge Pump", NULL, "ADC Left Mux"}, + {"Charge Pump", NULL, "ADC Right Mux"}, + + {"ADC Left Mux", "Input1A", "AIN1L"}, + {"ADC Right Mux", "Input1B", "AIN1R"}, + {"ADC Left Mux", "Input2A", "AIN2L"}, + {"ADC Right Mux", "Input2B", "AIN2R"}, + {"ADC Left Mux", "Input3A", "AIN3L"}, + {"ADC Right Mux", "Input3B", "AIN3R"}, + {"ADC Left Mux", "Input4A", "AIN4L"}, + {"ADC Right Mux", "Input4B", "AIN4R"}, + {"ADC Left Mux", "PGA Input Left", "PGA Left"}, + {"ADC Right Mux", "PGA Input Right" , "PGA Right"}, + + {"PGA Left", "Switch", "AIN1L"}, + {"PGA Right", "Switch", "AIN1R"}, + {"PGA Left", "Switch", "AIN2L"}, + {"PGA Right", "Switch", "AIN2R"}, + {"PGA Left", "Switch", "AIN3L"}, + {"PGA Right", "Switch", "AIN3R"}, + {"PGA Left", "Switch", "AIN4L"}, + {"PGA Right", "Switch", "AIN4R"}, + + {"PGA Left", "Switch", "PGA MICA"}, + {"PGA MICA", NULL, "MICA"}, + + {"PGA Right", "Switch", "PGA MICB"}, + {"PGA MICB", NULL, "MICB"}, + + {"HPOUTA", NULL, "HP Left Amp"}, + {"HPOUTB", NULL, "HP Right Amp"}, + {"HP Left Amp", NULL, "Bypass Left"}, + {"HP Right Amp", NULL, "Bypass Right"}, + {"Bypass Left", "Switch", "PGA Left"}, + {"Bypass Right", "Switch", "PGA Right"}, + {"HP Left Amp", "Switch", "DAC Left"}, + {"HP Right Amp", "Switch", "DAC Right"}, + + {"SPKOUTA", NULL, "SPK Left Amp"}, + {"SPKOUTB", NULL, "SPK Right Amp"}, + + {"SPK Left Amp", NULL, "Beep"}, + {"SPK Right Amp", NULL, "Beep"}, + {"SPK Left Amp", "Switch", "Playback"}, + {"SPK Right Amp", "Switch", "Playback"}, + + {"DAC Left", NULL, "Beep"}, + {"DAC Right", NULL, "Beep"}, + {"DAC Left", NULL, "Playback"}, + {"DAC Right", NULL, "Playback"}, + + {"Output Mux", "DSP", "Playback"}, + {"Output Mux", "DSP", "Playback"}, + + {"AIFINL", NULL, "Playback"}, + {"AIFINR", NULL, "Playback"}, + +}; + +struct cs42l52_clk_para { + u32 mclk; + u32 rate; + u8 speed; + u8 group; + u8 videoclk; + u8 ratio; + u8 mclkdiv2; +}; + +static const struct cs42l52_clk_para clk_map_table[] = { + /*8k*/ + {12288000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 8000, CLK_QS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0}, + + /*11.025k*/ + {11289600, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /*16k*/ + {12288000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 16000, CLK_HS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 1}, + + /*22.05k*/ + {11289600, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 32k */ + {12288000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 32000, CLK_SS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0}, + + /* 44.1k */ + {11289600, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 48k */ + {12288000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_27M_MCLK, CLK_R_125, 1}, + + /* 88.2k */ + {11289600, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 96k */ + {12288000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1}, +}; + +static int cs42l52_get_clk(int mclk, int rate) +{ + int i, ret = 0; + u_int mclk1, mclk2 = 0; + + for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { + if (clk_map_table[i].rate == rate) { + mclk1 = clk_map_table[i].mclk; + if (abs(mclk - mclk1) < abs(mclk - mclk2)) { + mclk2 = mclk1; + ret = i; + } + } + } + if (ret > ARRAY_SIZE(clk_map_table)) + return -EINVAL; + return ret; +} + +static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) { + cs42l52->sysclk = freq; + } else { + dev_err(codec->dev, "Invalid freq paramter\n"); + return -EINVAL; + } + return 0; +} + +static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = CS42L52_IFACE_CTL1_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = CS42L52_IFACE_CTL1_SLAVE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= CS42L52_IFACE_CTL1_ADC_FMT_I2S | + CS42L52_IFACE_CTL1_DAC_FMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface |= CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J | + CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= CS42L52_IFACE_CTL1_DSP_MODE_EN; + break; + case SND_SOC_DAIFMT_DSP_B: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= CS42L52_IFACE_CTL1_INV_SCLK; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= CS42L52_IFACE_CTL1_INV_SCLK; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + ret = -EINVAL; + } + cs42l52->config.format = iface; + snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format); + + return 0; +} + +static int cs42l52_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) + snd_soc_update_bits(codec, CS42L52_PB_CTL1, + CS42L52_PB_CTL1_MUTE_MASK, + CS42L52_PB_CTL1_MUTE); + else + snd_soc_update_bits(codec, CS42L52_PB_CTL1, + CS42L52_PB_CTL1_MUTE_MASK, + CS42L52_PB_CTL1_UNMUTE); + + return 0; +} + +static int cs42l52_pcm_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 cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + u32 clk = 0; + int index; + + index = cs42l52_get_clk(cs42l52->sysclk, params_rate(params)); + if (index >= 0) { + cs42l52->sysclk = clk_map_table[index].mclk; + + clk |= (clk_map_table[index].speed << CLK_SPEED_SHIFT) | + (clk_map_table[index].group << CLK_32K_SR_SHIFT) | + (clk_map_table[index].videoclk << CLK_27M_MCLK_SHIFT) | + (clk_map_table[index].ratio << CLK_RATIO_SHIFT) | + clk_map_table[index].mclkdiv2; + + snd_soc_write(codec, CS42L52_CLK_CTL, clk); + } else { + dev_err(codec->dev, "can't get correct mclk\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42l52_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, CS42L52_PWRCTL1, + CS42L52_PWRCTL1_PDN_CODEC, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + regcache_cache_only(cs42l52->regmap, false); + regcache_sync(cs42l52->regmap); + } + snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL); + regcache_cache_only(cs42l52->regmap, true); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +#define CS42L52_RATES (SNDRV_PCM_RATE_8000_96000) + +#define CS42L52_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) + +static struct snd_soc_dai_ops cs42l52_ops = { + .hw_params = cs42l52_pcm_hw_params, + .digital_mute = cs42l52_digital_mute, + .set_fmt = cs42l52_set_fmt, + .set_sysclk = cs42l52_set_sysclk, +}; + +struct snd_soc_dai_driver cs42l52_dai = { + .name = "cs42l52", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L52_RATES, + .formats = CS42L52_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L52_RATES, + .formats = CS42L52_FORMATS, + }, + .ops = &cs42l52_ops, +}; + +static int cs42l52_suspend(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l52_resume(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +static int beep_rates[] = { + 261, 522, 585, 667, 706, 774, 889, 1000, + 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 +}; + +static void cs42l52_beep_work(struct work_struct *work) +{ + struct cs42l52_private *cs42l52 = + container_of(work, struct cs42l52_private, beep_work); + struct snd_soc_codec *codec = cs42l52->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int i; + int val = 0; + int best = 0; + + if (cs42l52->beep_rate) { + for (i = 0; i < ARRAY_SIZE(beep_rates); i++) { + if (abs(cs42l52->beep_rate - beep_rates[i]) < + abs(cs42l52->beep_rate - beep_rates[best])) + best = i; + } + + dev_dbg(codec->dev, "Set beep rate %dHz for requested %dHz\n", + beep_rates[best], cs42l52->beep_rate); + + val = (best << CS42L52_BEEP_RATE_SHIFT); + + snd_soc_dapm_enable_pin(dapm, "Beep"); + } else { + dev_dbg(codec->dev, "Disabling beep\n"); + snd_soc_dapm_disable_pin(dapm, "Beep"); + } + + snd_soc_update_bits(codec, CS42L52_BEEP_FREQ, + CS42L52_BEEP_RATE_MASK, val); + + snd_soc_dapm_sync(dapm); +} + +/* For usability define a way of injecting beep events for the device - + * many systems will not have a keyboard. + */ +static int cs42l52_beep_event(struct input_dev *dev, unsigned int type, + unsigned int code, int hz) +{ + struct snd_soc_codec *codec = input_get_drvdata(dev); + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "Beep event %x %x\n", code, hz); + + switch (code) { + case SND_BELL: + if (hz) + hz = 261; + case SND_TONE: + break; + default: + return -1; + } + + /* Kick the beep from a workqueue */ + cs42l52->beep_rate = hz; + schedule_work(&cs42l52->beep_work); + return 0; +} + +static ssize_t cs42l52_beep_set(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct cs42l52_private *cs42l52 = dev_get_drvdata(dev); + long int time; + int ret; + + ret = kstrtol(buf, 10, &time); + if (ret != 0) + return ret; + + input_event(cs42l52->beep, EV_SND, SND_TONE, time); + + return count; +} + +static DEVICE_ATTR(beep, 0200, NULL, cs42l52_beep_set); + +static void cs42l52_init_beep(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret; + + cs42l52->beep = input_allocate_device(); + if (!cs42l52->beep) { + dev_err(codec->dev, "Failed to allocate beep device\n"); + return; + } + + INIT_WORK(&cs42l52->beep_work, cs42l52_beep_work); + cs42l52->beep_rate = 0; + + cs42l52->beep->name = "CS42L52 Beep Generator"; + cs42l52->beep->phys = dev_name(codec->dev); + cs42l52->beep->id.bustype = BUS_I2C; + + cs42l52->beep->evbit[0] = BIT_MASK(EV_SND); + cs42l52->beep->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); + cs42l52->beep->event = cs42l52_beep_event; + cs42l52->beep->dev.parent = codec->dev; + input_set_drvdata(cs42l52->beep, codec); + + ret = input_register_device(cs42l52->beep); + if (ret != 0) { + input_free_device(cs42l52->beep); + cs42l52->beep = NULL; + dev_err(codec->dev, "Failed to register beep device\n"); + } + + ret = device_create_file(codec->dev, &dev_attr_beep); + if (ret != 0) { + dev_err(codec->dev, "Failed to create keyclick file: %d\n", + ret); + } +} + +static void cs42l52_free_beep(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + device_remove_file(codec->dev, &dev_attr_beep); + input_unregister_device(cs42l52->beep); + cancel_work_sync(&cs42l52->beep_work); + cs42l52->beep = NULL; + + snd_soc_update_bits(codec, CS42L52_BEEP_TONE_CTL, + CS42L52_BEEP_EN_MASK, 0); +} +#else +static void cs42l52_init_beep(struct snd_soc_codec *codec) +{ +} + +static void cs42l52_free_beep(struct snd_soc_codec *codec) +{ +} +#endif + +static int cs42l52_probe(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = cs42l52->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + regcache_cache_only(cs42l52->regmap, true); + + cs42l52_init_beep(codec); + + cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + cs42l52->sysclk = CS42L52_DEFAULT_CLK; + cs42l52->config.format = CS42L52_DEFAULT_FORMAT; + + /* Set Platform MICx CFG */ + snd_soc_update_bits(codec, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.mica_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + snd_soc_update_bits(codec, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.micb_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + /* if Single Ended, Get Mic_Select */ + if (cs42l52->pdata.mica_cfg) + snd_soc_update_bits(codec, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.mica_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + if (cs42l52->pdata.micb_cfg) + snd_soc_update_bits(codec, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.micb_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + + /* Set Platform Charge Pump Freq */ + snd_soc_update_bits(codec, CS42L52_CHARGE_PUMP, + CS42L52_CHARGE_PUMP_MASK, + cs42l52->pdata.chgfreq << + CS42L52_CHARGE_PUMP_SHIFT); + + /* Set Platform Bias Level */ + snd_soc_update_bits(codec, CS42L52_IFACE_CTL2, + CS42L52_IFACE_CTL2_BIAS_LVL, + cs42l52->pdata.micbias_lvl); + + return ret; +} + +static int cs42l52_remove(struct snd_soc_codec *codec) +{ + cs42l52_free_beep(codec); + cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { + .probe = cs42l52_probe, + .remove = cs42l52_remove, + .suspend = cs42l52_suspend, + .resume = cs42l52_resume, + .set_bias_level = cs42l52_set_bias_level, + + .dapm_widgets = cs42l52_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l52_dapm_widgets), + .dapm_routes = cs42l52_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l52_audio_map), + + .controls = cs42l52_snd_controls, + .num_controls = ARRAY_SIZE(cs42l52_snd_controls), +}; + +/* Current and threshold powerup sequence Pg37 */ +static const struct reg_default cs42l52_threshold_patch[] = { + + { 0x00, 0x99 }, + { 0x3E, 0xBA }, + { 0x47, 0x80 }, + { 0x32, 0xBB }, + { 0x32, 0x3B }, + { 0x00, 0x00 }, + +}; + +static struct regmap_config cs42l52_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L52_MAX_REGISTER, + .reg_defaults = cs42l52_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs42l52_reg_defaults), + .readable_reg = cs42l52_readable_register, + .volatile_reg = cs42l52_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs42l52_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l52_private *cs42l52; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private), + GFP_KERNEL); + if (cs42l52 == NULL) + return -ENOMEM; + cs42l52->dev = &i2c_client->dev; + + cs42l52->regmap = regmap_init_i2c(i2c_client, &cs42l52_regmap); + if (IS_ERR(cs42l52->regmap)) { + ret = PTR_ERR(cs42l52->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + i2c_set_clientdata(i2c_client, cs42l52); + + if (dev_get_platdata(&i2c_client->dev)) + memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev), + sizeof(cs42l52->pdata)); + + ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch, + ARRAY_SIZE(cs42l52_threshold_patch)); + if (ret != 0) + dev_warn(cs42l52->dev, "Failed to apply regmap patch: %d\n", + ret); + + ret = regmap_read(cs42l52->regmap, CS42L52_CHIP, &reg); + devid = reg & CS42L52_CHIP_ID_MASK; + if (devid != CS42L52_CHIP_ID) { + ret = -ENODEV; + dev_err(&i2c_client->dev, + "CS42L52 Device ID (%X). Expected %X\n", + devid, CS42L52_CHIP_ID); + goto err_regmap; + } + + regcache_cache_only(cs42l52->regmap, true); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l52, &cs42l52_dai, 1); + if (ret < 0) + goto err_regmap; + return 0; + +err_regmap: + regmap_exit(cs42l52->regmap); + +err: + return ret; +} + +static int cs42l52_i2c_remove(struct i2c_client *client) +{ + struct cs42l52_private *cs42l52 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(cs42l52->regmap); + + return 0; +} + +static const struct i2c_device_id cs42l52_id[] = { + { "cs42l52", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs42l52_id); + +static struct i2c_driver cs42l52_i2c_driver = { + .driver = { + .name = "cs42l52", + .owner = THIS_MODULE, + }, + .id_table = cs42l52_id, + .probe = cs42l52_i2c_probe, + .remove = __devexit_p(cs42l52_i2c_remove), +}; + +static void __exit cs42l52_exit(void) +{ + i2c_del_driver(&cs42l52_i2c_driver); +} + +module_exit(cs42l52_exit); + +module_i2c_driver(cs42l52_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS42L52 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com>"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h new file mode 100644 index 0000000..60985c0 --- /dev/null +++ b/sound/soc/codecs/cs42l52.h @@ -0,0 +1,274 @@ +/* + * cs42l52.h -- CS42L52 ALSA SoC audio driver + * + * Copyright 2012 CirrusLogic, Inc. + * + * Author: Georgi Vlaev <joe(a)nucleusys.com> + * Author: Brian Austin <brian.austin(a)cirrus.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. + * + */ + +#ifndef __CS42L52_H__ +#define __CS42L52_H__ + +#define CS42L52_NAME "CS42L52" +#define CS42L52_DEFAULT_CLK 12000000 +#define CS42L52_MIN_CLK 11000000 +#define CS42L52_MAX_CLK 27000000 +#define CS42L52_DEFAULT_FORMAT SNDRV_PCM_FMTBIT_S16_LE +#define CS42L52_DEFAULT_MAX_CHANS 2 +#define CS42L52_SYSCLK 1 + +#define CS42L52_CHIP_SWICTH (1 << 17) +#define CS42L52_ALL_IN_ONE (1 << 16) +#define CS42L52_CHIP_ONE 0x00 +#define CS42L52_CHIP_TWO 0x01 +#define CS42L52_CHIP_THR 0x02 +#define CS42L52_CHIP_MASK 0x0f + +#define CS42L52_FIX_BITS_CTL 0x00 +#define CS42L52_CHIP 0x01 +#define CS42L52_CHIP_ID 0xE0 +#define CS42L52_CHIP_ID_MASK 0xF8 +#define CS42L52_CHIP_REV_A0 0x00 +#define CS42L52_CHIP_REV_A1 0x01 +#define CS42L52_CHIP_REV_B0 0x02 +#define CS42L52_CHIP_REV_MASK 0x03 + +#define CS42L52_PWRCTL1 0x02 +#define CS42L52_PWRCTL1_PDN_ALL 0x9F +#define CS42L52_PWRCTL1_PDN_CHRG 0x80 +#define CS42L52_PWRCTL1_PDN_PGAB 0x10 +#define CS42L52_PWRCTL1_PDN_PGAA 0x08 +#define CS42L52_PWRCTL1_PDN_ADCB 0x04 +#define CS42L52_PWRCTL1_PDN_ADCA 0x02 +#define CS42L52_PWRCTL1_PDN_CODEC 0x01 + +#define CS42L52_PWRCTL2 0x03 +#define CS42L52_PWRCTL2_OVRDB (1 << 4) +#define CS42L52_PWRCTL2_OVRDA (1 << 3) +#define CS42L52_PWRCTL2_PDN_MICB (1 << 2) +#define CS42L52_PWRCTL2_PDN_MICB_SHIFT 2 +#define CS42L52_PWRCTL2_PDN_MICA (1 << 1) +#define CS42L52_PWRCTL2_PDN_MICA_SHIFT 1 +#define CS42L52_PWRCTL2_PDN_MICBIAS (1 << 0) +#define CS42L52_PWRCTL2_PDN_MICBIAS_SHIFT 0 + +#define CS42L52_PWRCTL3 0x04 +#define CS42L52_PWRCTL3_HPB_PDN_SHIFT 6 +#define CS42L52_PWRCTL3_HPB_ON_LOW 0x00 +#define CS42L52_PWRCTL3_HPB_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_HPB_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_HPB_ALWAYS_OFF 0x03 +#define CS42L52_PWRCTL3_HPA_PDN_SHIFT 4 +#define CS42L52_PWRCTL3_HPA_ON_LOW 0x00 +#define CS42L52_PWRCTL3_HPA_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_HPA_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_HPA_ALWAYS_OFF 0x03 +#define CS42L52_PWRCTL3_SPKB_PDN_SHIFT 2 +#define CS42L52_PWRCTL3_SPKB_ON_LOW 0x00 +#define CS42L52_PWRCTL3_SPKB_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_SPKB_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_PDN_SPKB (1 << 2) +#define CS42L52_PWRCTL3_PDN_SPKA (1 << 0) +#define CS42L52_PWRCTL3_SPKA_PDN_SHIFT 0 +#define CS42L52_PWRCTL3_SPKA_ON_LOW 0x00 +#define CS42L52_PWRCTL3_SPKA_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_SPKA_ALWAYS_ON 0x02 + +#define CS42L52_DEFAULT_OUTPUT_STATE 0x05 +#define CS42L52_PWRCTL3_CONF_MASK 0x03 + +#define CS42L52_CLK_CTL 0x05 +#define CLK_AUTODECT_ENABLE (1 << 7) +#define CLK_SPEED_SHIFT 5 +#define CLK_DS_MODE 0x00 +#define CLK_SS_MODE 0x01 +#define CLK_HS_MODE 0x02 +#define CLK_QS_MODE 0x03 +#define CLK_32K_SR_SHIFT 4 +#define CLK_32K 0x01 +#define CLK_NO_32K 0x00 +#define CLK_27M_MCLK_SHIFT 3 +#define CLK_27M_MCLK 0x01 +#define CLK_NO_27M 0x00 +#define CLK_RATIO_SHIFT 1 +#define CLK_R_128 0x00 +#define CLK_R_125 0x01 +#define CLK_R_132 0x02 +#define CLK_R_136 0x03 + +#define CS42L52_IFACE_CTL1 0x06 +#define CS42L52_IFACE_CTL1_MASTER (1 << 7) +#define CS42L52_IFACE_CTL1_SLAVE (0 << 7) +#define CS42L52_IFACE_CTL1_INV_SCLK (1 << 6) +#define CS42L52_IFACE_CTL1_ADC_FMT_I2S (1 << 5) +#define CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J (0 << 5) +#define CS42L52_IFACE_CTL1_DSP_MODE_EN (1 << 4) +#define CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J (0 << 2) +#define CS42L52_IFACE_CTL1_DAC_FMT_I2S (1 << 2) +#define CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J (2 << 2) +#define CS42L52_IFACE_CTL1_WL_32BIT (0x00) +#define CS42L52_IFACE_CTL1_WL_24BIT (0x01) +#define CS42L52_IFACE_CTL1_WL_20BIT (0x02) +#define CS42L52_IFACE_CTL1_WL_16BIT (0x03) +#define CS42L52_IFACE_CTL1_WL_MASK 0xFFFF + +#define CS42L52_IFACE_CTL2 0x07 +#define CS42L52_IFACE_CTL2_SC_MC_EQ (1 << 6) +#define CS42L52_IFACE_CTL2_LOOPBACK (1 << 5) +#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_EN (0 << 4) +#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_HIZ (1 << 4) +#define CS42L52_IFACE_CTL2_HP_SW_INV (1 << 3) +#define CS42L52_IFACE_CTL2_BIAS_LVL 0x07 + +#define CS42L52_ADC_PGA_A 0x08 +#define CS42L52_ADC_PGA_B 0x09 +#define CS42L52_ADC_SEL_SHIFT 5 +#define CS42L52_ADC_SEL_AIN1 0x00 +#define CS42L52_ADC_SEL_AIN2 0x01 +#define CS42L52_ADC_SEL_AIN3 0x02 +#define CS42L52_ADC_SEL_AIN4 0x03 +#define CS42L52_ADC_SEL_PGA 0x04 + +#define CS42L52_ANALOG_HPF_CTL 0x0A +#define CS42L52_HPF_CTL_ANLGSFTB (1 << 3) +#define CS42L52_HPF_CTL_ANLGSFTA (1 << 0) + +#define CS42L52_ADC_HPF_FREQ 0x0B +#define CS42L52_ADC_MISC_CTL 0x0C +#define CS42L52_ADC_MISC_CTL_SOURCE_DSP (1 << 6) + +#define CS42L52_PB_CTL1 0x0D +#define CS42L52_PB_CTL1_HP_GAIN_SHIFT 5 +#define CS42L52_PB_CTL1_HP_GAIN_03959 0x00 +#define CS42L52_PB_CTL1_HP_GAIN_04571 0x01 +#define CS42L52_PB_CTL1_HP_GAIN_05111 0x02 +#define CS42L52_PB_CTL1_HP_GAIN_06047 0x03 +#define CS42L52_PB_CTL1_HP_GAIN_07099 0x04 +#define CS42L52_PB_CTL1_HP_GAIN_08399 0x05 +#define CS42L52_PB_CTL1_HP_GAIN_10000 0x06 +#define CS42L52_PB_CTL1_HP_GAIN_11430 0x07 +#define CS42L52_PB_CTL1_INV_PCMB (1 << 3) +#define CS42L52_PB_CTL1_INV_PCMA (1 << 2) +#define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1) +#define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0) +#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD +#define CS42L52_PB_CTL1_MUTE 3 +#define CS42L52_PB_CTL1_UNMUTE 0 + +#define CS42L52_MISC_CTL 0x0E +#define CS42L52_MISC_CTL_DEEMPH (1 << 2) +#define CS42L52_MISC_CTL_DIGSFT (1 << 1) +#define CS42L52_MISC_CTL_DIGZC (1 << 0) + +#define CS42L52_PB_CTL2 0x0F +#define CS42L52_PB_CTL2_HPB_MUTE (1 << 7) +#define CS42L52_PB_CTL2_HPA_MUTE (1 << 6) +#define CS42L52_PB_CTL2_SPKB_MUTE (1 << 5) +#define CS42L52_PB_CTL2_SPKA_MUTE (1 << 4) +#define CS42L52_PB_CTL2_SPK_SWAP (1 << 2) +#define CS42L52_PB_CTL2_SPK_MONO (1 << 1) +#define CS42L52_PB_CTL2_SPK_MUTE50 (1 << 0) + +#define CS42L52_MICA_CTL 0x10 +#define CS42L52_MICB_CTL 0x11 +#define CS42L52_MIC_CTL_MIC_SEL_MASK 0xBF +#define CS42L52_MIC_CTL_MIC_SEL_SHIFT 6 +#define CS42L52_MIC_CTL_TYPE_MASK 0xDF +#define CS42L52_MIC_CTL_TYPE_SHIFT 5 + + +#define CS42L52_PGAA_CTL 0x12 +#define CS42L52_PGAB_CTL 0x13 +#define CS42L52_PGAX_CTL_VOL_12DB 24 +#define CS42L52_PGAX_CTL_VOL_6DB 12 /*step size 0.5db*/ + +#define CS42L52_PASSTHRUA_VOL 0x14 +#define CS42L52_PASSTHRUB_VOL 0x15 + +#define CS42L52_ADCA_VOL 0x16 +#define CS42L52_ADCB_VOL 0x17 +#define CS42L52_ADCX_VOL_24DB 24 /*step size 1db*/ +#define CS42L52_ADCX_VOL_12DB 12 +#define CS42L52_ADCX_VOL_6DB 6 + +#define CS42L52_ADCA_MIXER_VOL 0x18 +#define CS42L52_ADCB_MIXER_VOL 0x19 +#define CS42L52_ADC_MIXER_VOL_12DB 0x18 + +#define CS42L52_PCMA_MIXER_VOL 0x1A +#define CS42L52_PCMB_MIXER_VOL 0x1B + +#define CS42L52_BEEP_FREQ 0x1C +#define CS42L52_BEEP_VOL 0x1D +#define CS42L52_BEEP_TONE_CTL 0x1E +#define CS42L52_BEEP_RATE_SHIFT 4 +#define CS42L52_BEEP_RATE_MASK 0x0F + +#define CS42L52_TONE_CTL 0x1F +#define CS42L52_BEEP_EN_MASK 0x3F + +#define CS42L52_MASTERA_VOL 0x20 +#define CS42L52_MASTERB_VOL 0x21 + +#define CS42L52_HPA_VOL 0x22 +#define CS42L52_HPB_VOL 0x23 +#define CS42L52_DEFAULT_HP_VOL 0xF0 + +#define CS42L52_SPKA_VOL 0x24 +#define CS42L52_SPKB_VOL 0x25 +#define CS42L52_DEFAULT_SPK_VOL 0xF0 + +#define CS42L52_ADC_PCM_MIXER 0x26 + +#define CS42L52_LIMITER_CTL1 0x27 +#define CS42L52_LIMITER_CTL2 0x28 +#define CS42L52_LIMITER_AT_RATE 0x29 + +#define CS42L52_ALC_CTL 0x2A +#define CS42L52_ALC_CTL_ALCB_ENABLE_SHIFT 7 +#define CS42L52_ALC_CTL_ALCA_ENABLE_SHIFT 6 +#define CS42L52_ALC_CTL_FASTEST_ATTACK 0 + +#define CS42L52_ALC_RATE 0x2B +#define CS42L52_ALC_SLOWEST_RELEASE 0x3F + +#define CS42L52_ALC_THRESHOLD 0x2C +#define CS42L52_ALC_MAX_RATE_SHIFT 5 +#define CS42L52_ALC_MIN_RATE_SHIFT 2 +#define CS42L52_ALC_RATE_0DB 0 +#define CS42L52_ALC_RATE_3DB 1 +#define CS42L52_ALC_RATE_6DB 2 + +#define CS42L52_NOISE_GATE_CTL 0x2D +#define CS42L52_NG_ENABLE_SHIFT 6 +#define CS42L52_NG_THRESHOLD_SHIFT 2 +#define CS42L52_NG_MIN_70DB 2 +#define CS42L52_NG_DELAY_SHIFT 0 +#define CS42L52_NG_DELAY_100MS 1 + +#define CS42L52_CLK_STATUS 0x2E +#define CS42L52_BATT_COMPEN 0x2F + +#define CS42L52_BATT_LEVEL 0x30 +#define CS42L52_SPK_STATUS 0x31 +#define CS42L52_SPK_STATUS_PIN_SHIFT 3 +#define CS42L52_SPK_STATUS_PIN_HIGH 1 + +#define CS42L52_TEM_CTL 0x32 +#define CS42L52_TEM_CTL_SET 0x80 +#define CS42L52_THE_FOLDBACK 0x33 +#define CS42L52_CHARGE_PUMP 0x34 +#define CS42L52_CHARGE_PUMP_MASK 0xF0 +#define CS42L52_CHARGE_PUMP_SHIFT 4 +#define CS42L52_FIX_BITS1 0x3E +#define CS42L52_FIX_BITS2 0x47 + +#define CS42L52_MAX_REGISTER 0x34 + +#endif -- 1.7.9.1

2 1

Re: [alsa-devel] ASoC: dpcm: Add debugFS support for DPCM
by Dan Carpenter 30 Apr '12

30 Apr '12

Hello Liam Girdwood, The patch f86dcef87b77: "ASoC: dpcm: Add debugFS support for DPCM" from Apr 25, 2012, leads to the following Smatch warning: sound/soc/soc-core.c:1221 soc_post_component_init() error: potential null derefence 'dai_link'. 1143 static int soc_post_component_init(struct snd_soc_card *card, 1144 struct snd_soc_codec *codec, 1145 int num, int dailess) 1146 { 1147 struct snd_soc_dai_link *dai_link = NULL; 1148 struct snd_soc_aux_dev *aux_dev = NULL; 1149 struct snd_soc_pcm_runtime *rtd; 1150 const char *temp, *name; 1151 int ret = 0; 1152 1153 if (!dailess) { 1154 dai_link = &card->dai_link[num]; ^^^^^^^^ dai_link is only set sometimes; it depends on dailess. 1155 rtd = &card->rtd[num]; 1156 name = dai_link->name; 1157 } else { 1158 aux_dev = &card->aux_dev[num]; 1159 rtd = &card->rtd_aux[num]; 1160 name = aux_dev->name; 1161 } [snip] 1219 #ifdef CONFIG_DEBUG_FS 1220 /* add DPCM sysfs entries */ 1221 if (!dai_link->dynamic) ^^^^^^^^^^ Smatch complains about this. 1222 goto out; 1223 1224 ret = soc_dpcm_debugfs_add(rtd); 1225 if (ret < 0) 1226 dev_err(rtd->dev, "asoc: failed to add dpcm sysfs entries: %d\n", ret); 1227 1228 out: 1229 #endif regards, dan carpenter

2 1

Re: [alsa-devel] [PATCH 7/8] ASoC: codecs: Add AB8500 codec-driver
by Mark Brown 30 Apr '12

30 Apr '12

On Fri, Apr 20, 2012 at 11:33:22AM +0200, Ola Lilja wrote: This is massively better than previous versions! There's still some issues but hopefully not hard to correct. > @@ -12,6 +12,7 @@ config SND_SOC_ALL_CODECS > tristate "Build all ASoC CODEC drivers" > select SND_SOC_88PM860X if MFD_88PM860X > select SND_SOC_L3 > + select SND_SOC_AB8500_CODEC if SND_SOC_UX500 Shouldn't this depend on the MFD core for the device instead? > +static int regulators_init(struct device *dev) > +{ > + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); > + > + dev_dbg(dev, "%s: Enter\n", __func__); > + > + regulators_get_regulator(dev, &drvdata->reg_vaud, "V-AUD"); > + regulators_get_regulator(dev, &drvdata->reg_vamic1, "V-AMIC1"); > + regulators_get_regulator(dev, &drvdata->reg_vamic2, "V-AMIC2"); > + regulators_get_regulator(dev, &drvdata->reg_vdmic, "V-DMIC"); > + > + if (IS_ERR(drvdata->reg_vaud.consumer) || > + IS_ERR(drvdata->reg_vamic1.consumer) || > + IS_ERR(drvdata->reg_vamic2.consumer) || > + IS_ERR(drvdata->reg_vdmic.consumer)) { > + regulators_cleanup(drvdata); > + return -ENXIO; > + } This won't work with probe deferral - if we need to defer then the driver will squash down the -EPROBE_DEFER and fail totally. Not a big deal, though. > +static int dapm_audioclk_event(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *k, int event) > +{ We should add a variant of REGULATOR_SUPPLY for clocks too, have a framework thing that will own the clock - this is all generic code which gets repeated for each clock, we could easily factor it out into the core. > +static int dapm_audioreg_event(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *k, int event) > +{ > + struct device *dev = w->codec->dev; > + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); > + int status = 0; > + > + if (SND_SOC_DAPM_EVENT_ON(event)) > + status = regulator_enable(drvdata->reg_vaud.consumer); > + else > + regulator_disable(drvdata->reg_vaud.consumer); Similarly this is just a REGULATOR_SUPPLY. > +/* Mic 1b - Regulator select */ > +static const struct soc_enum enum_mic1breg_sel = SOC_ENUM_SINGLE(0, 0, 2, > + enum_micreg); > +static const struct snd_kcontrol_new dapm_mic1breg_mux[] = { > + SOC_DAPM_ENUM_VIRT("Mic 1b Regulator Select", > + enum_mic1breg_sel), > +}; Can you explain how this hardware works in more detail? It seems very odd to be changing the regulator used to supply something at runtime. > +static int mclk_input_control_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); > + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); > + unsigned int val; > + > + val = (ucontrol->value.enumerated.item[0] != 0); > + if (drvdata->mclk_sel == val) > + return 0; > + > + drvdata->mclk_sel = val; > + > + return 1; > +} This is really weird > +static const char * const enum_earselcm[] = {"0.95V", "1.10V", "1.27V", > + "1.58V"}; > +static SOC_ENUM_SINGLE_DECL(soc_enum_earselcm, > + AB8500_ANACONF1, AB8500_ANACONF1_EARSELCM, enum_earselcm); Some of this stuff looks awfully like it ought to be platform data... > +static const char * const enum_ensemicx[] = {"Differential", "Single Ended"}; > +static SOC_ENUM_SINGLE_DECL(soc_enum_ensemic1, > + AB8500_ANAGAIN1, AB8500_ANAGAINX_ENSEMICX, enum_ensemicx); > +static SOC_ENUM_SINGLE_DECL(soc_enum_lowpowmic1, > + AB8500_ANAGAIN1, AB8500_ANAGAINX_LOWPOWMICX, enum_dis_ena); This for example is normally fixed by the physical design and can't sensibly be varied at runtime. > + /* Digital interface - AD to slot mapping */ > + SOC_ENUM("Digital Interface AD To Slot 0 Map", soc_enum_adslot0map), > + SOC_ENUM("Digital Interface AD To Slot 1 Map", soc_enum_adslot1map), Can you usefully leave these fixed with a default configuration? There's been some chat about adding a framework feature for this but it's not there yet - lots of devices have similar features. If not then this code is fine. > +static int ab8500_codec_configure_audio_macrocell(struct snd_soc_codec *codec) > +{ > + u8 value8; > + unsigned int value; > + int status; > + > + status = ab8500_sysctrl_write(AB8500_STW4500CTRL3, > + AB8500_STW4500CTRL3_CLK32KOUT2DIS | > + AB8500_STW4500CTRL3_RESETAUDN, > + AB8500_STW4500CTRL3_RESETAUDN); > + if (status < 0) > + return status; > + > + status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC, > + (u8)AB8500_GPIO_DIR4_REG, > + &value8); > + if (status < 0) > + return status; > + value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT | > + GPIO31_DIR_OUTPUT; > + status |= abx500_set_register_interruptible(codec->dev, > + (u8)AB8500_MISC, > + (u8)AB8500_GPIO_DIR4_REG, > + value); Still not sure why this isn't platform data. > + /* Add controls */ > + status = snd_soc_add_codec_controls(codec, ab8500_ctrls, > + ARRAY_SIZE(ab8500_ctrls)); > + if (status < 0) { > + dev_err(dev, "%s: failed to add codec-controls (%d).\n", > + __func__, status); > + return status; > + } At least this one could be done from the driver struct. > + /* Add DAPM-widgets */ > + status = snd_soc_dapm_new_controls(&codec->dapm, ab8500_dapm_widgets, > + ARRAY_SIZE(ab8500_dapm_widgets)); > + if (status < 0) { > + dev_err(codec->dev, > + "%s: Failed to create DAPM controls (%d).\n", > + __func__, status); > + return status; > + } This could also be done from the driver struct. > + status = snd_soc_dapm_add_routes(&codec->dapm, ab8500_dapm_routes, > + ARRAY_SIZE(ab8500_dapm_routes)); > + if (status < 0) { > + dev_err(codec->dev, "%s: Failed to add DAPM routes (%d).\n", > + __func__, status); > + return status; > + } This too. > + if (IS_ERR(drvdata->clk_ptr_sysclk)) { > + dev_err(dev, > + "%s: ERROR: Clocks needed for streaming not available!", > + __func__); > + return -ENXIO; return PTR_ERR(). > + dev_info(&pdev->dev, "%s: Register codec.\n", __func__); Remove this or downgrade to debug, it's noisy and not adding much information.

2 6

[alsa-devel] [PATCH] ASoC: tlv312aic23: unbreak resume
by Eric Bénard 30 Apr '12

30 Apr '12

* commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to a bug preventing resumeof the codec as regmap expects a 9 bits data register but 0xFFFF is passed in tlv320aic23_set_bias_level and this values gets cached preventing any write to the TLV320AIC23_PWR register as the final value produced by regmap is (register << 9) | value * this patch solves the problem by only working on the 9 bits the register contains. Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/codecs/tlv320aic23.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index dfa41a9..e7de911 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -472,7 +472,7 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f; switch (level) { case SND_SOC_BIAS_ON: @@ -491,7 +491,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); - snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff); break; } codec->dapm.bias_level = level; -- 1.7.7.6

2 1

[alsa-devel] Required/preferred multichannel order for ALSA
by Ricardo Neri 29 Apr '12

29 Apr '12

Hi Takashi, list, I am implementing multichannel support for HDMI on Texas Instruments' OMAP4. I would like to know if ALSA mandates a specific channel order or has a preferred one. I tried to find some guidance in alsa-lib or in the alsa driver. All I could find is the order described in speaker-test: FL/FR/RL/RR/C/LFE/SL/SR. This order seems to be in use due to historical reasons [1]. It was also mentioned that an API to set/get the channel mapping was going to be implemented [2]; I tried to find it without success. As [1] and [2] are very old posts, I was wondering if the situation has changed. My question arises from the fact that HDMI audio uses the channel ordering defined in CEA-861 section 6.6.2, which is different from what speaker-test expects. This also different from the order that SMPTE 320M specifies. OMAP4 is able to alter the channel mapping, so I could match what ALSA expects if such required/preferred order exists. Thanks in advance for your comments. Ricardo [1].http://www.spinics.net/lists/alsa-devel/msg24444.html [2].http://www.spinics.net/lists/alsa-devel/msg24495.html

4 6

Re: [alsa-devel] ALSA: Updating Sample Rates in sound/core/pcm_native.c
by Clemens Ladisch 27 Apr '12

27 Apr '12

Jeremy Torres wrote: > Thanks for the quick reply! So, I followed Takashi's advice and added > the constraint list for the new sample rates and no longer changed > pcm.h/pcm_native.c. I can record/play at the new sample rates; > however, ALSA is performing rate conversion. For example, to run at > 144K, the slave is configured to run 192K, even though my hardware can > be configured to support 144K. This sounds like a bug in the driver. Unfortunately, my crystal ball is out of order, so you have to actually show the source code for me to be able to help you. Regards, Clemens

2 1

[alsa-devel] [PATCH] ASoC: da7210: Minor bugfix for non pll slave mode
by Ashish Chavan 27 Apr '12

27 Apr '12

This patch fixes a bug discovered during testing of non pll slave mode. Due to the bug chip was not getting correctly configured and as a result there was no sound output while playback. After applying this patch, both pll and non pll modes work fine. 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 | 13 +++++++------ 1 files changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index f92d1f7..10f52ff 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1159,8 +1159,9 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); - /* Set PLL Master clock range 10-20 MHz */ - snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ); + /* Set PLL Master clock range 10-20 MHz, enable PLL bypass */ + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | + DA7210_PLL_BYP); /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); @@ -1191,8 +1192,8 @@ static struct reg_default da7210_regmap_i2c_patch[] = { /* System controller master disable */ { DA7210_STARTUP1, 0x00 }, - /* make sure that DA7210 use bypass mode before start up */ - { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP }, + /* Set PLL Master clock range 10-20 MHz */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ }, /* to unlock */ { DA7210_A_HID_UNLOCK, 0x8B}, @@ -1290,8 +1291,8 @@ static struct reg_default da7210_regmap_spi_patch[] = { /* System controller master disable */ { DA7210_STARTUP1, 0x00 }, - /* make sure that DA7210 use bypass mode before start up */ - { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP }, + /* Set PLL Master clock range 10-20 MHz */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ }, /* to set PAGE1 of SPI register space */ { DA7210_PAGE_CONTROL, 0x80 }, -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: dpcm: Fixup debugFS for DPCM state.
by Liam Girdwood 27 Apr '12

27 Apr '12

Remove writable debugFS permission, use simple_open() and fix indentation. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-pcm.c | 17 +++++------------ 1 files changed, 5 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ca19152..0e68d2d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2389,12 +2389,6 @@ static char *dpcm_state_string(enum snd_soc_dpcm_state state) return "unknown"; } -static int dpcm_state_open_file(struct inode *inode, struct file *file) -{ - file->private_data = inode->i_private; - return 0; -} - static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, int stream, char *buf, size_t size) { @@ -2472,15 +2466,14 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE, buf + offset, out_count - offset); - ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); - - kfree(buf); + ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); - return ret; + kfree(buf); + return ret; } static const struct file_operations dpcm_state_fops = { - .open = dpcm_state_open_file, + .open = simple_open, .read = dpcm_state_read_file, .llseek = default_llseek, }; @@ -2496,7 +2489,7 @@ int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) return -EINVAL; } - rtd->debugfs_dpcm_state = debugfs_create_file("state", 0644, + rtd->debugfs_dpcm_state = debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root, rtd, &dpcm_state_fops); -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm_hubs: Special case headphones for digital paths in more use cases
by Mark Brown 27 Apr '12

27 Apr '12

The optimisations which we can do with caching the headphone DCS result in wm_hubs have only been enabled in cases where class W is enabled. However, there are more use cases which can benefit from the cache, especially with WM8994 series devices with their more advanced digital routing. Rather than keying off the class W information from the CODECs have a check in wm_hubs for a suitable path and use that to determine if we can deploy our headphone optimisations. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 2 -- sound/soc/codecs/wm8994.c | 4 +--- sound/soc/codecs/wm_hubs.c | 48 ++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/wm_hubs.h | 5 ++--- 4 files changed, 45 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index d256a93..8bb0059 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -852,7 +852,6 @@ static int class_w_put(struct snd_kcontrol *kcontrol, 0); } wm8993->class_w_users++; - wm8993->hubs_data.class_w = true; } /* Implement the change */ @@ -869,7 +868,6 @@ static int class_w_put(struct snd_kcontrol *kcontrol, WM8993_CP_DYN_V); } wm8993->class_w_users--; - wm8993->hubs_data.class_w = false; } dev_dbg(codec->dev, "Indirect DAC use count now %d\n", diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 587062b..5b7bb10 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -998,13 +998,11 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) WM8994_CP_DYN_PWR | WM8994_CP_DYN_SRC_SEL_MASK, source | WM8994_CP_DYN_PWR); - wm8994->hubs.class_w = true; } else { dev_dbg(codec->dev, "Class W disabled\n"); snd_soc_update_bits(codec, WM8994_CLASS_W_1, WM8994_CP_DYN_PWR, 0); - wm8994->hubs.class_w = false; } } @@ -3777,7 +3775,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.dcs_readback_mode = 2; wm8994->hubs.no_series_update = 1; wm8994->hubs.hp_startup_mode = 1; - wm8994->hubs.no_cache_class_w = true; + wm8994->hubs.no_cache_dac_hp_direct = true; wm8994->fll_byp = true; switch (wm8994->revision) { diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 6c028c4..df44083 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -109,6 +109,42 @@ irqreturn_t wm_hubs_dcs_done(int irq, void *data) } EXPORT_SYMBOL_GPL(wm_hubs_dcs_done); +static bool wm_hubs_dac_hp_direct(struct snd_soc_codec *codec) +{ + int reg; + + /* If we're going via the mixer we'll need to do additional checks */ + reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER1); + if (!(reg & WM8993_DACL_TO_HPOUT1L)) { + if (reg & ~WM8993_DACL_TO_MIXOUTL) { + dev_vdbg(codec->dev, "Analogue paths connected: %x\n", + reg & ~WM8993_DACL_TO_HPOUT1L); + return false; + } else { + dev_vdbg(codec->dev, "HPL connected to mixer\n"); + return false; + } + } else { + dev_vdbg(codec->dev, "HPL connected to DAC\n"); + } + + reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER2); + if (!(reg & WM8993_DACR_TO_HPOUT1R)) { + if (reg & ~WM8993_DACR_TO_MIXOUTR) { + dev_vdbg(codec->dev, "Analogue paths connected: %x\n", + reg & ~WM8993_DACR_TO_HPOUT1R); + return false; + } else { + dev_vdbg(codec->dev, "HPR connected to mixer\n"); + return false; + } + } else { + dev_vdbg(codec->dev, "HPR connected to DAC\n"); + } + + return true; +} + /* * Startup calibration of the DC servo */ @@ -129,10 +165,10 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* If we're using a digital only path and have a previously * callibrated DC servo offset stored then use that. */ - if (hubs->class_w && hubs->class_w_dcs) { + if (wm_hubs_dac_hp_direct(codec) && hubs->dac_hp_direct_dcs) { dev_dbg(codec->dev, "Using cached DC servo offset %x\n", - hubs->class_w_dcs); - snd_soc_write(codec, dcs_reg, hubs->class_w_dcs); + hubs->dac_hp_direct_dcs); + snd_soc_write(codec, dcs_reg, hubs->dac_hp_direct_dcs); wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); @@ -207,8 +243,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* Save the callibrated offset if we're in class W mode and * therefore don't have any analogue signal mixed in. */ - if (hubs->class_w && !hubs->no_cache_class_w) - hubs->class_w_dcs = dcs_cfg; + if (wm_hubs_dac_hp_direct(codec) && !hubs->no_cache_dac_hp_direct) + hubs->dac_hp_direct_dcs = dcs_cfg; } /* @@ -224,7 +260,7 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, ret = snd_soc_put_volsw(kcontrol, ucontrol); /* Updating the analogue gains invalidates the DC servo cache */ - hubs->class_w_dcs = 0; + hubs->dac_hp_direct_dcs = 0; /* If we're applying an offset correction then updating the * callibration would be likely to introduce further offsets. */ diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 5705276..8bb9f1b 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -30,9 +30,8 @@ struct wm_hubs_data { int series_startup; int no_series_update; - bool no_cache_class_w; - bool class_w; - u16 class_w_dcs; + bool no_cache_dac_hp_direct; + u16 dac_hp_direct_dcs; bool lineout1_se; bool lineout1n_ena; -- 1.7.10

1 2

[alsa-devel] Soc-DSP / Dynamic PCM
by Sebastien LEDUC 27 Apr '12

27 Apr '12

Hi I have seen some discussion on the mailing list about upcoming changes in ASoC for soc-dsp support / dynamic PCM. Could somebody explain what is the purpose of these changes? Is there a branch where I can have a look to the on-going work? Thanks Regards, Sebastien

2 1

[alsa-devel] [PATCH] ASoC: Add support for CS42L52 Codec
by Brian Austin 27 Apr '12

27 Apr '12

This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Georgi Vlaev <joe(a)nucleusys.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l52.c | 1079 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l52.h | 264 +++++++++++ 4 files changed, 1349 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs42l52.c create mode 100644 sound/soc/codecs/cs42l52.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2e51eb0..0235532 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L52 if I2C select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI @@ -183,6 +184,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L52 + tristate + config SND_SOC_CS42L73 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index db61c44..d11ab90 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o @@ -123,6 +124,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c new file mode 100644 index 0000000..a7a5ff5 --- /dev/null +++ b/sound/soc/codecs/cs42l52.c @@ -0,0 +1,1079 @@ +/* + * cs42l52.c -- CS42L52 ALSA SoC audio driver + * + * Copyright 2012 CirrusLogic, Inc. + * + * Author: Georgi Vlaev <joe(a)nucleusys.com> + * Author: Brian Austin <brian.austin(a)cirrus.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/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs42l52.h" + +struct sp_config { + u8 spc, format, spfs; + u32 srate; +}; + +struct cs42l52_private { + struct regmap *regmap; + struct sp_config config; + u32 sysclk; + u8 mclksel; + u32 mclk; + u8 flags; +}; + +static const struct reg_default cs42l52_reg_defaults[] = { + { 1, 0xE0 }, /* r01 ChipID */ + { 2, 0x9F }, /* r02 PWRCTL 1 */ + { 3, 0x07 }, /* r03 PWRCTL 2 */ + { 4, 0xFF }, /* r04 PWRCTL 3 */ + { 5, 0xA0 }, /* r05 Clocking Ctl */ + { 6, 0x00 }, /* r06 Interface Ctl 1 */ + { 7, 0x00 }, /* r07 Interface Ctl 2 */ + { 8, 0x80 }, /* r08 Input A Select */ + { 9, 0x80 }, /* r09 Input B Select */ + { 10, 0xA5 }, /* r0A Analog HPF Ctl */ + { 11, 0x00 }, /* r0B ADC HPF Corner Freq */ + { 12, 0x00 }, /* r0C Misc. ADC Ctl */ + { 13, 0x60 }, /* r0D Playback Ctl 1 */ + { 14, 0x02 }, /* r0E Misc. Ctl */ + { 15, 0x00 }, /* r0F Playback Ctl 2 */ + { 16, 0x00 }, /* r10 MICA Amp Ctl */ + { 17, 0x00 }, /* r11 MICB Amp Ctl */ + { 18, 0x00 }, /* r12 PGAA Vol, Misc. */ + { 19, 0x00 }, /* r13 PGAB Vol, Misc. */ + { 20, 0x00 }, /* r14 Bypass A Vol */ + { 21, 0x00 }, /* r15 Bypass B Vol */ + { 22, 0x00 }, /* r16 ADCA Volume */ + { 23, 0x00 }, /* r17 ADCB Volume */ + { 24, 0x80 }, /* r18 ADCA Mixer Volume */ + { 25, 0x80 }, /* r19 ADCB Mixer Volume */ + { 26, 0x00 }, /* r1A PCMA Mixer Volume */ + { 27, 0x00 }, /* r1B PCMB Mixer Volume */ + { 28, 0x00 }, /* r1C Beep Freq on Time */ + { 29, 0x00 }, /* r1D Beep Volume off Time */ + { 30, 0x00 }, /* r1E Beep Tone Cfg. */ + { 31, 0x00 }, /* r1F Tone Ctl */ + { 32, 0x88 }, /* r20 Master A Volume */ + { 33, 0x00 }, /* r21 Master B Volume */ + { 34, 0x00 }, /* r22 Headphone A Volume */ + { 35, 0x00 }, /* r23 Headphone B Volume */ + { 36, 0x00 }, /* r24 Speaker A Volume */ + { 37, 0x00 }, /* r25 Speaker B Volume */ + { 38, 0x00 }, /* r26 Channel Mixer and Swap */ + { 39, 0x00 }, /* r27 Limit Ctl 1 Thresholds */ + { 40, 0x7F }, /* r28 Limit Ctl 2 Release Rate */ + { 41, 0xC0 }, /* r29 Limiter Attack Rate */ + { 42, 0x00 }, /* r2A ALC Ctl 1 Attack Rate */ + { 43, 0x3F }, /* r2B ALC Release Rate */ + { 44, 0x3f }, /* r2C ALC Thresholds */ + { 45, 0x00 }, /* r2D Noise Gate Ctl */ + { 46, 0x00 }, /* r2E Overflow and Clock Status */ + { 47, 0x00 }, /* r2F battery Compensation */ + { 48, 0x00 }, /* r30 VP Battery Level */ + { 49, 0x00 }, /* r31 Speaker Status */ + { 50, 0x3B }, /* r32 Temp Ctl */ + { 51, 0x00 }, /* r33 Foldback */ + { 52, 0x5F }, /* r34 Change Pump Freq */ +}; + +static bool cs42l52_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L52_CHIP: + case CS42L52_PWRCTL1: + case CS42L52_PWRCTL2: + case CS42L52_PWRCTL3: + case CS42L52_CLK_CTL: + case CS42L52_IFACE_CTL1: + case CS42L52_IFACE_CTL2: + case CS42L52_ADC_PGA_A: + case CS42L52_ADC_PGA_B: + case CS42L52_ANALOG_HPF_CTL: + case CS42L52_ADC_HPF_FREQ: + case CS42L52_ADC_MISC_CTL: + case CS42L52_PB_CTL1: + case CS42L52_MISC_CTL: + case CS42L52_PB_CTL2: + case CS42L52_MICA_CTL: + case CS42L52_MICB_CTL: + case CS42L52_PGAA_CTL: + case CS42L52_PGAB_CTL: + case CS42L52_PASSTHRUA_VOL: + case CS42L52_PASSTHRUB_VOL: + case CS42L52_ADCA_VOL: + case CS42L52_ADCB_VOL: + case CS42L52_ADCA_MIXER_VOL: + case CS42L52_ADCB_MIXER_VOL: + case CS42L52_PCMA_MIXER_VOL: + case CS42L52_PCMB_MIXER_VOL: + case CS42L52_BEEP_FREQ: + case CS42L52_BEEP_VOL: + case CS42L52_BEEP_TONE_CTL: + case CS42L52_TONE_CTL: + case CS42L52_MASTERA_VOL: + case CS42L52_MASTERB_VOL: + case CS42L52_HPA_VOL: + case CS42L52_HPB_VOL: + case CS42L52_SPKA_VOL: + case CS42L52_SPKB_VOL: + case CS42L52_ADC_PCM_MIXER: + case CS42L52_LIMITER_CTL1: + case CS42L52_LIMITER_CTL2: + case CS42L52_LIMITER_AT_RATE: + case CS42L52_ALC_CTL: + case CS42L52_ALC_RATE: + case CS42L52_ALC_THRESHOLD: + case CS42L52_NOISE_GATE_CTL: + case CS42L52_CLK_STATUS: + case CS42L52_BATT_COMPEN: + case CS42L52_BATT_LEVEL: + case CS42L52_SPK_STATUS: + case CS42L52_TEM_CTL: + case CS42L52_THE_FOLDBACK: + case CS42L52_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(hpd_tlv, -9600, 50, 1); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); + +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const char * const cs42l52_adca_text[] = { + "Input1A", "Input2A", "Input3A", "Input4A", "PGA Input Left"}; + +static const char * const cs42l52_adcb_text[] = { + "Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"}; + +static const struct soc_enum adca_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5, + ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text); + +static const struct soc_enum adcb_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5, + ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text); + +static const struct snd_kcontrol_new adca_mux = + SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum); + +static const struct snd_kcontrol_new adcb_mux = + SOC_DAPM_ENUM("Right ADC Input Capture Mux", adcb_enum); + +static const char * const mic_bias_level_text[] = { + "0.5 +VA", "0.6 +VA", "0.7 +VA", + "0.8 +VA", "0.83 +VA", "0.91 +VA" +}; + +static const struct soc_enum mic_bias_level_enum = + SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0, + ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); + +static const char * const cs42l52_mic_text[] = { "Single", "Differential" }; + +static const struct soc_enum mica_enum = + SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5, + ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); + +static const struct soc_enum micb_enum = + SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5, + ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); + +static const struct snd_kcontrol_new mica_mux = + SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum); + +static const struct snd_kcontrol_new micb_mux = + SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum); + +static const char * const digital_output_mux_text[] = {"ADC", "DSP"}; + +static const struct soc_enum digital_output_mux_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6, + ARRAY_SIZE(digital_output_mux_text), + digital_output_mux_text); + +static const struct snd_kcontrol_new digital_output_mux = + SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum); + +static const char * const hp_gain_num_text[] = { + "0.3959", "0.4571", "0.5111", "0.6047", + "0.7099", "0.8399", "1.000", "1.1430" +}; + +static const struct soc_enum hp_gain_enum = + SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4, + ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text); + +static const char * const beep_pitch_text[] = { + "C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5", + "C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7" +}; + +static const struct soc_enum beep_pitch_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4, + ARRAY_SIZE(beep_pitch_text), beep_pitch_text); + +static const char * const beep_ontime_text[] = { + "86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s", + "1.80 s", "2.20 s", "2.50 s", "2.80 s", "3.20 s", + "3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s" +}; + +static const struct soc_enum beep_ontime_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0, + ARRAY_SIZE(beep_ontime_text), beep_ontime_text); + +static const char * const beep_offtime_text[] = { + "1.23 s", "2.58 s", "3.90 s", "5.20 s", + "6.60 s", "8.05 s", "9.35 s", "10.80 s" +}; + +static const struct soc_enum beep_offtime_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5, + ARRAY_SIZE(beep_offtime_text), beep_offtime_text); + +static const char * const beep_config_text[] = { + "Off", "Single", "Multiple", "Continuous" +}; + +static const struct soc_enum beep_config_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6, + ARRAY_SIZE(beep_config_text), beep_config_text); + +static const char * const beep_bass_text[] = { + "50 Hz", "100 Hz", "200 Hz", "250 Hz" +}; + +static const struct soc_enum beep_bass_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1, + ARRAY_SIZE(beep_bass_text), beep_bass_text); + +static const char * const beep_treble_text[] = { + "5 kHz", "7 kHz", "10 kHz", " 15 kHz" +}; + +static const struct soc_enum beep_treble_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3, + ARRAY_SIZE(beep_treble_text), beep_treble_text); + +static const char * const ng_threshold_text[] = { + "-34dB", "-37dB", "-40dB", "-43dB", + "-46dB", "-52dB", "-58dB", "-64dB" +}; + +static const struct soc_enum ng_threshold_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2, + ARRAY_SIZE(ng_threshold_text), ng_threshold_text); + +static const char * const cs42l52_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms"}; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0, + ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text); + +static const char * const cs42l52_ng_type_text[] = { + "Apply Specific", "Apply All" +}; + +static const struct soc_enum ng_type_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6, + ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text); + +static const char * const charge_pump_freq_text[] = { + "0", "1", "2", "3", "4", + "5", "6", "7", "8", "9", + "10", "11", "12", "13", "14", "15" }; + +static const struct soc_enum charge_pump_enum = + SOC_ENUM_SINGLE(CS42L52_CHARGE_PUMP, 4, + ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); + +static const char * const left_swap_text[] = { + "Left", "LR 2", "Right"}; + +static const char * const right_swap_text[] = { + "Right", "LR 2", "Left"}; + +static const unsigned int swap_values[] = { 0, 1, 3 }; + +static const struct soc_enum adca_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, + ARRAY_SIZE(left_swap_text), + left_swap_text, + swap_values); + +static const struct soc_enum pcma_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1, + ARRAY_SIZE(left_swap_text), + left_swap_text, + swap_values); + +static const struct soc_enum adcb_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1, + ARRAY_SIZE(right_swap_text), + right_swap_text, + swap_values); + +static const struct soc_enum pcmb_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1, + ARRAY_SIZE(right_swap_text), + right_swap_text, + swap_values); + +static const struct snd_kcontrol_new passthrul_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 6, 1, 0); + +static const struct snd_kcontrol_new passthrur_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 7, 1, 0); + +static const struct snd_kcontrol_new spkl_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new spkr_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new hpl_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new hpr_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 6, 1, 1); + +static const struct snd_kcontrol_new cs42l52_snd_controls[] = { + + SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L52_MASTERA_VOL, + CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv), + + SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, + CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv), + + SOC_ENUM("Headphone Analog Gain", hp_gain_enum), + + SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, + CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv), + + SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, + CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), + + SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), + + SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL, + CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv), + + SOC_ENUM("MIC Bias Level", mic_bias_level_enum), + + SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, + CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv), + SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", + CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, + 6, 0x7f, 0x19, ipd_tlv), + + SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), + + SOC_DOUBLE_R("ADC Mixer Switch", CS42L52_ADCA_MIXER_VOL, + CS42L52_ADCB_MIXER_VOL, 7, 1, 1), + + SOC_ENUM("ADC Left Swap", adca_swap_enum), + SOC_ENUM("ADC Right Swap", adcb_swap_enum), + + SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv), + + SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", + CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, + 6, 0x7f, 0x19, hl_tlv), + SOC_DOUBLE_R("PCM Mixer Switch", + CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), + + SOC_ENUM("PCM Left Swap", pcma_swap_enum), + SOC_ENUM("PCM Right Swap", pcmb_swap_enum), + + SOC_ENUM("Beep Pitch", beep_pitch_enum), + SOC_ENUM("Beep on Time", beep_ontime_enum), + SOC_ENUM("Beep off Time", beep_offtime_enum), + SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_ENUM("Beep Config", beep_config_enum), + SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), + SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), + SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), + + SOC_SINGLE("Tone Control Switch", CS42L52_BEEP_TONE_CTL, 0, 1, 1), + SOC_SINGLE_TLV("Treble Gain Volume", + CS42L52_TONE_CTL, 4, 15, 1, hl_tlv), + SOC_SINGLE_TLV("Bass Gain Volume", + CS42L52_TONE_CTL, 0, 15, 1, hl_tlv), + + /* Limiter */ + SOC_SINGLE_TLV("Limiter Max Threshold Volume", + CS42L52_LIMITER_CTL1, 5, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Cushion Threshold Volume", + CS42L52_LIMITER_CTL1, 2, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Release Rate Volume", + CS42L52_LIMITER_CTL2, 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Attack Rate Volume", + CS42L52_LIMITER_AT_RATE, 0, 63, 0, limiter_tlv), + + SOC_SINGLE("Limiter SR Switch", CS42L52_LIMITER_CTL1, 1, 1, 0), + SOC_SINGLE("Limiter ZC Switch", CS42L52_LIMITER_CTL1, 0, 1, 0), + SOC_SINGLE("Limiter Switch", CS42L52_LIMITER_CTL2, 7, 1, 0), + + /* ALC */ + SOC_SINGLE_TLV("ALC Attack Rate Volume", CS42L52_ALC_CTL, + 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Release Rate Volume", CS42L52_ALC_RATE, + 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L52_ALC_THRESHOLD, + 5, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L52_ALC_THRESHOLD, + 2, 7, 0, limiter_tlv), + + SOC_DOUBLE_R("ALC SR Capture Switch", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 7, 1, 1), + SOC_DOUBLE_R("ALC ZC Capture Switch", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 6, 1, 1), + SOC_DOUBLE("ALC Capture Switch", CS42L52_ALC_CTL, 6, 7, 1, 0), + + /* Noise gate */ + SOC_ENUM("NG Type Switch", ng_type_enum), + SOC_SINGLE("NG Enable Switch", CS42L52_NOISE_GATE_CTL, 6, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L52_NOISE_GATE_CTL, 5, 1, 1), + SOC_ENUM("NG Threshold", ng_threshold_enum), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_DOUBLE("HPF Switch", CS42L52_ANALOG_HPF_CTL, 5, 7, 1, 0), + + SOC_DOUBLE("Analog SR Switch", CS42L52_ANALOG_HPF_CTL, 1, 3, 1, 1), + SOC_DOUBLE("Analog ZC Switch", CS42L52_ANALOG_HPF_CTL, 0, 2, 1, 1), + SOC_SINGLE("Digital SR Switch", CS42L52_MISC_CTL, 1, 1, 0), + SOC_SINGLE("Digital ZC Switch", CS42L52_MISC_CTL, 0, 1, 0), + SOC_SINGLE("Deemphasis Switch", CS42L52_MISC_CTL, 2, 1, 0), + + SOC_SINGLE("Batt Compensation Switch", CS42L52_BATT_COMPEN, 7, 1, 0), + SOC_SINGLE("Batt VP Monitor Switch", CS42L52_BATT_COMPEN, 6, 1, 0), + SOC_SINGLE("Batt VP ref", CS42L52_BATT_COMPEN, 0, 0x0f, 0), + SOC_ENUM("Charge Pump Freq", charge_pump_enum), + + SOC_SINGLE("PGA AIN1L Switch", CS42L52_ADC_PGA_A, 0, 1, 0), + SOC_SINGLE("PGA AIN1R Switch", CS42L52_ADC_PGA_B, 0, 1, 0), + SOC_SINGLE("PGA AIN2L Switch", CS42L52_ADC_PGA_A, 1, 1, 0), + SOC_SINGLE("PGA AIN2R Switch", CS42L52_ADC_PGA_B, 1, 1, 0), + + SOC_SINGLE("PGA AIN3L Switch", CS42L52_ADC_PGA_A, 2, 1, 0), + SOC_SINGLE("PGA AIN3R Switch", CS42L52_ADC_PGA_B, 2, 1, 0), + + SOC_SINGLE("PGA AIN4L Switch", CS42L52_ADC_PGA_A, 3, 1, 0), + SOC_SINGLE("PGA AIN4R Switch", CS42L52_ADC_PGA_B, 3, 1, 0), + + SOC_SINGLE("PGA MICA Switch", CS42L52_ADC_PGA_A, 4, 1, 0), + SOC_SINGLE("PGA MICB Switch", CS42L52_ADC_PGA_B, 4, 1, 0), + +}; + +static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), + SND_SOC_DAPM_INPUT("AIN4L"), + SND_SOC_DAPM_INPUT("AIN4R"), + SND_SOC_DAPM_INPUT("MICA"), + SND_SOC_DAPM_INPUT("MICB"), + + SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux), + SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1), + SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", CS42L52_PWRCTL1, 4, 1, NULL, 0), + + SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adca_mux), + SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcb_mux), + + SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, + 0, 0, &digital_output_mux), + + SND_SOC_DAPM_PGA("PGA MICA", CS42L52_PWRCTL2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("PGA MICB", CS42L52_PWRCTL2, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", CS42L52_PWRCTL2, 0, 1, NULL, 0), + + SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SWITCH("Bypass Left", CS42L52_MISC_CTL, + 6, 0, &passthrul_ctl), + SND_SOC_DAPM_SWITCH("Bypass Right", CS42L52_MISC_CTL, + 7, 0, &passthrur_ctl), + + SND_SOC_DAPM_SWITCH("HP Left Amp", SND_SOC_NOPM, 0, 0, &hpl_ctl), + SND_SOC_DAPM_SWITCH("HP Right Amp", SND_SOC_NOPM, 0, 0, &hpr_ctl), + + SND_SOC_DAPM_SWITCH("SPK Left Amp", SND_SOC_NOPM, 0, 0, &spkl_ctl), + SND_SOC_DAPM_SWITCH("SPK Right Amp", SND_SOC_NOPM, 0, 0, &spkr_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("SPKOUTA"), + SND_SOC_DAPM_OUTPUT("SPKOUTB"), + +}; + +static const struct snd_soc_dapm_route cs42l52_audio_map[] = { + + {"AIFOUTL", NULL, "Output Mux"}, + {"AIFOUTR", NULL, "Output Mux"}, + + {"Output Mux", "ADC", "ADC Left"}, + {"Output Mux", "ADC", "ADC Right"}, + + {"ADC Left", NULL, "ADC Left Mux"}, + {"ADC Right", NULL, "ADC Right Mux"}, + + {"ADC Left Mux", "Input1A", "AIN1L"}, + {"ADC Right Mux", "Input1B", "AIN1R"}, + {"ADC Left Mux", "Input2A", "AIN2L"}, + {"ADC Right Mux", "Input2B", "AIN2R"}, + {"ADC Left Mux", "Input3A", "AIN3L"}, + {"ADC Right Mux", "Input3B", "AIN3R"}, + {"ADC Left Mux", "Input4A", "AIN4L"}, + {"ADC Right Mux", "Input4B", "AIN4R"}, + {"ADC Left Mux", "PGA Input Left", "PGA Left"}, + {"ADC Right Mux", "PGA Input Right" , "PGA Right"}, + + {"PGA Left", "Switch", "AIN1L"}, + {"PGA Right", "Switch", "AIN1R"}, + {"PGA Left", "Switch", "AIN2L"}, + {"PGA Right", "Switch", "AIN2R"}, + {"PGA Left", "Switch", "AIN3L"}, + {"PGA Right", "Switch", "AIN3R"}, + {"PGA Left", "Switch", "AIN4L"}, + {"PGA Right", "Switch", "AIN4R"}, + + {"PGA Left", "Switch", "PGA MICA"}, + {"PGA MICA", NULL, "MICA Mux"}, + {"MICA Mux", "Single", "MICA"}, + {"MICA Mux", "Differential", "MICA"}, + + {"PGA Right", "Switch", "PGA MICB"}, + {"PGA MICB", NULL, "MICB Mux"}, + {"MICB Mux", "Single", "MICB"}, + {"MICB Mux", "Differential", "MICB"}, + + {"HPOUTA", NULL, "HP Left Amp"}, + {"HPOUTB", NULL, "HP Right Amp"}, + {"HP Left Amp", NULL, "Bypass Left"}, + {"HP Right Amp", NULL, "Bypass Right"}, + {"Bypass Left", "Switch", "PGA Left"}, + {"Bypass Right", "Switch", "PGA Right"}, + {"HP Left Amp", "Switch", "DAC Left"}, + {"HP Right Amp", "Switch", "DAC Right"}, + + {"SPKOUTA", NULL, "SPK Left Amp"}, + {"SPKOUTB", NULL, "SPK Right Amp"}, + {"SPK Left Amp", "Switch", "AIFINL"}, + {"SPK Right Amp", "Switch", "AIFINR"}, + + {"DAC Left", NULL, "AIFINL"}, + {"DAC Right", NULL, "AIFINR"}, + + {"Output Mux", "DSP", "AIFINL"}, + {"Output Mux", "DSP", "AIFINR"}, + +}; + +struct cs42l52_clk_para { + u32 mclk; + u32 rate; + u8 speed; + u8 group; + u8 videoclk; + u8 ratio; + u8 mclkdiv2; +}; + +static const struct cs42l52_clk_para clk_map_table[] = { + /*8k*/ + {12288000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 8000, CLK_QS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0}, + + /*11.025k*/ + {11289600, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /*16k*/ + {12288000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 16000, CLK_HS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 1}, + + /*22.05k*/ + {11289600, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 32k */ + {12288000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 32000, CLK_SS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0}, + + /* 44.1k */ + {11289600, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 48k */ + {12288000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_27M_MCLK, CLK_R_125, 1}, + + /* 88.2k */ + {11289600, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 96k */ + {12288000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1}, +}; + +static int cs42l52_get_clk(int mclk, int rate) +{ + int i, ret = 0; + u_int mclk1, mclk2 = 0; + + for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { + if (clk_map_table[i].rate == rate) { + mclk1 = clk_map_table[i].mclk; + if (abs(mclk - mclk1) < abs(mclk - mclk2)) { + mclk2 = mclk1; + ret = i; + } + } + } + if (ret > ARRAY_SIZE(clk_map_table)) + return -EINVAL; + return ret; +} + +static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) { + cs42l52->sysclk = freq; + } else { + dev_err(codec->dev, "Invalid freq paramter\n"); + return -EINVAL; + } + return 0; +} + +static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = CS42L52_IFACE_CTL1_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = CS42L52_IFACE_CTL1_SLAVE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= CS42L52_IFACE_CTL1_ADC_FMT_I2S | + CS42L52_IFACE_CTL1_DAC_FMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface |= CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J | + CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= CS42L52_IFACE_CTL1_DSP_MODE_EN; + break; + case SND_SOC_DAIFMT_DSP_B: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= CS42L52_IFACE_CTL1_INV_SCLK; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= CS42L52_IFACE_CTL1_INV_SCLK; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + ret = -EINVAL; + } + cs42l52->config.format = iface; + snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format); + + return 0; +} + +static int cs42l52_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) + snd_soc_update_bits(codec, CS42L52_PB_CTL1, + CS42L52_PB_CTL1_MUTE_MASK, 0x03); + else + snd_soc_update_bits(codec, CS42L52_PB_CTL1, + CS42L52_PB_CTL1_MUTE_MASK, 0x00); + + return 0; +} + +static int cs42l52_pcm_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 cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + u32 clk = 0; + int index; + + index = cs42l52_get_clk(cs42l52->sysclk, params_rate(params)); + if (index >= 0) { + cs42l52->sysclk = clk_map_table[index].mclk; + + clk |= (clk_map_table[index].speed << CLK_SPEED_SHIFT) | + (clk_map_table[index].group << CLK_32K_SR_SHIFT) | + (clk_map_table[index].videoclk << CLK_27M_MCLK_SHIFT) | + (clk_map_table[index].ratio << CLK_RATIO_SHIFT) | + clk_map_table[index].mclkdiv2; + + snd_soc_write(codec, CS42L52_CLK_CTL, clk); + } else { + dev_err(codec->dev, "can't get correct mclk\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42l52_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, CS42L52_PWRCTL1, + CS42L52_PWRCTL1_PDN_CODEC | + CS42L52_PWRCTL1_PDN_CHRG, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + regcache_cache_only(cs42l52->regmap, false); + regcache_sync(cs42l52->regmap); + } + snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +#define CS42L52_RATES (SNDRV_PCM_RATE_8000_96000) + +#define CS42L52_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) + +static struct snd_soc_dai_ops cs42l52_ops = { + .hw_params = cs42l52_pcm_hw_params, + .digital_mute = cs42l52_digital_mute, + .set_fmt = cs42l52_set_fmt, + .set_sysclk = cs42l52_set_sysclk, +}; + +struct snd_soc_dai_driver cs42l52_dai = { + .name = "cs42l52", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L52_RATES, + .formats = CS42L52_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L52_RATES, + .formats = CS42L52_FORMATS, + }, + .ops = &cs42l52_ops, +}; + +static int cs42l52_suspend(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l52_resume(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* page 37 from cs42l52 datasheet */ +static void cs42l52_required_setup(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_ON); + + snd_soc_write(codec, CS42L52_FIX_BITS_CTL, 0x99); + snd_soc_write(codec, CS42L52_FIX_BITS1, 0xBA); + snd_soc_write(codec, CS42L52_FIX_BITS2, 0x80); + + snd_soc_update_bits(codec, CS42L52_TEM_CTL, CS42L52_TEM_CTL_SET, 1); + snd_soc_update_bits(codec, CS42L52_TEM_CTL, CS42L52_TEM_CTL_SET, 0); + + snd_soc_write(codec, CS42L52_FIX_BITS_CTL, 0x00); + + cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int cs42l52_probe(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = cs42l52->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + regcache_cache_only(cs42l52->regmap, true); + + cs42l52_required_setup(codec); + + cs42l52->sysclk = CS42L52_DEFAULT_CLK; + cs42l52->config.format = CS42L52_DEFAULT_FORMAT; + + cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return ret; +} + +static int cs42l52_remove(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { + .probe = cs42l52_probe, + .remove = cs42l52_remove, + .suspend = cs42l52_suspend, + .resume = cs42l52_resume, + .set_bias_level = cs42l52_set_bias_level, + + .dapm_widgets = cs42l52_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l52_dapm_widgets), + .dapm_routes = cs42l52_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l52_audio_map), + + .controls = cs42l52_snd_controls, + .num_controls = ARRAY_SIZE(cs42l52_snd_controls), +}; + +static struct regmap_config cs42l52_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L52_MAX_REGISTER, + .reg_defaults = cs42l52_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs42l52_reg_defaults), + .readable_reg = cs42l52_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs42l52_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l52_private *cs42l52; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private), + GFP_KERNEL); + if (cs42l52 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, cs42l52); + + cs42l52->regmap = regmap_init_i2c(i2c_client, &cs42l52_regmap); + if (IS_ERR(cs42l52->regmap)) { + ret = PTR_ERR(cs42l52->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + ret = regmap_read(cs42l52->regmap, CS42L52_CHIP, &reg); + devid = reg & CS42L52_CHIP_ID_MASK; + + if (devid != CS42L52_CHIP_ID) { + ret = -ENODEV; + dev_err(&i2c_client->dev, + "CS42L52 Device ID (%X). Expected %X\n", + devid, CS42L52_CHIP_ID); + goto err_regmap; + } + + regcache_cache_only(cs42l52->regmap, true); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l52, &cs42l52_dai, 1); + if (ret < 0) + goto err_regmap; + return 0; + +err_regmap: + regmap_exit(cs42l52->regmap); + +err: + return ret; +} + +static int cs42l52_i2c_remove(struct i2c_client *client) +{ + struct cs42l52_private *cs42l52 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(cs42l52->regmap); + + return 0; +} + +static const struct i2c_device_id cs42l52_id[] = { + { "cs42l52", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs42l52_id); + +static struct i2c_driver cs42l52_i2c_driver = { + .driver = { + .name = "cs42l52", + .owner = THIS_MODULE, + }, + .id_table = cs42l52_id, + .probe = cs42l52_i2c_probe, + .remove = __devexit_p(cs42l52_i2c_remove), +}; + +static int __init cs42l52_modinit(void) +{ + int ret; + ret = i2c_add_driver(&cs42l52_i2c_driver); + if (ret != 0) { + pr_err("Failed to register CS42L52 I2C driver: %d\n", ret); + return ret; + } + return 0; +} + +module_init(cs42l52_modinit); + +static void __exit cs42l52_exit(void) +{ + i2c_del_driver(&cs42l52_i2c_driver); +} + +module_exit(cs42l52_exit); + +MODULE_DESCRIPTION("ASoC CS42L52 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com>"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h new file mode 100644 index 0000000..6a598b5 --- /dev/null +++ b/sound/soc/codecs/cs42l52.h @@ -0,0 +1,264 @@ +/* + * cs42l52.h -- CS42L52 ALSA SoC audio driver + * + * Copyright 2012 CirrusLogic, Inc. + * + * Author: Georgi Vlaev <joe(a)nucleusys.com> + * Author: Brian Austin <brian.austin(a)cirrus.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. + * + */ + +#ifndef __CS42L52_H__ +#define __CS42L52_H__ + +#define CS42L52_NAME "CS42L52" +#define CS42L52_DEFAULT_CLK 12000000 +#define CS42L52_MIN_CLK 11000000 +#define CS42L52_MAX_CLK 27000000 +#define CS42L52_DEFAULT_FORMAT SNDRV_PCM_FMTBIT_S16_LE +#define CS42L52_DEFAULT_MAX_CHANS 2 +#define CS42L52_SYSCLK 1 + +#define CS42L52_CHIP_SWICTH (1 << 17) +#define CS42L52_ALL_IN_ONE (1 << 16) +#define CS42L52_CHIP_ONE 0x00 +#define CS42L52_CHIP_TWO 0x01 +#define CS42L52_CHIP_THR 0x02 +#define CS42L52_CHIP_MASK 0x0f + +#define CS42L52_FIX_BITS_CTL 0x00 +#define CS42L52_CHIP 0x01 +#define CS42L52_CHIP_ID 0xE0 +#define CS42L52_CHIP_ID_MASK 0xF8 +#define CS42L52_CHIP_REV_A0 0x00 +#define CS42L52_CHIP_REV_A1 0x01 +#define CS42L52_CHIP_REV_B0 0x02 +#define CS42L52_CHIP_REV_MASK 0x03 + +#define CS42L52_PWRCTL1 0x02 +#define CS42L52_PWRCTL1_PDN_ALL 0x9F +#define CS42L52_PWRCTL1_PDN_CHRG 0x80 +#define CS42L52_PWRCTL1_PDN_PGAB 0x10 +#define CS42L52_PWRCTL1_PDN_PGAA 0x08 +#define CS42L52_PWRCTL1_PDN_ADCB 0x04 +#define CS42L52_PWRCTL1_PDN_ADCA 0x02 +#define CS42L52_PWRCTL1_PDN_CODEC 0x01 + +#define CS42L52_PWRCTL2 0x03 +#define CS42L52_PWRCTL2_OVRDB (1 << 4) +#define CS42L52_PWRCTL2_OVRDA (1 << 3) +#define CS42L52_PWRCTL2_PDN_MICB (1 << 2) +#define CS42L52_PWRCTL2_PDN_MICB_SHIFT 2 +#define CS42L52_PWRCTL2_PDN_MICA (1 << 1) +#define CS42L52_PWRCTL2_PDN_MICA_SHIFT 1 +#define CS42L52_PWRCTL2_PDN_MICBIAS (1 << 0) +#define CS42L52_PWRCTL2_PDN_MICBIAS_SHIFT 0 + +#define CS42L52_PWRCTL3 0x04 +#define CS42L52_PWRCTL3_HPB_PDN_SHIFT 6 +#define CS42L52_PWRCTL3_HPB_ON_LOW 0x00 +#define CS42L52_PWRCTL3_HPB_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_HPB_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_HPB_ALWAYS_OFF 0x03 +#define CS42L52_PWRCTL3_HPA_PDN_SHIFT 4 +#define CS42L52_PWRCTL3_HPA_ON_LOW 0x00 +#define CS42L52_PWRCTL3_HPA_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_HPA_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_HPA_ALWAYS_OFF 0x03 +#define CS42L52_PWRCTL3_SPKB_PDN_SHIFT 2 +#define CS42L52_PWRCTL3_SPKB_ON_LOW 0x00 +#define CS42L52_PWRCTL3_SPKB_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_SPKB_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_PDN_SPKB (1 << 2) +#define CS42L52_PWRCTL3_PDN_SPKA (1 << 0) +#define CS42L52_PWRCTL3_SPKA_PDN_SHIFT 0 +#define CS42L52_PWRCTL3_SPKA_ON_LOW 0x00 +#define CS42L52_PWRCTL3_SPKA_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_SPKA_ALWAYS_ON 0x02 + +#define CS42L52_DEFAULT_OUTPUT_STATE 0x05 +#define CS42L52_PWRCTL3_CONF_MASK 0x03 + +#define CS42L52_CLK_CTL 0x05 +#define CLK_AUTODECT_ENABLE (1 << 7) +#define CLK_SPEED_SHIFT 5 +#define CLK_DS_MODE 0x00 +#define CLK_SS_MODE 0x01 +#define CLK_HS_MODE 0x02 +#define CLK_QS_MODE 0x03 +#define CLK_32K_SR_SHIFT 4 +#define CLK_32K 0x01 +#define CLK_NO_32K 0x00 +#define CLK_27M_MCLK_SHIFT 3 +#define CLK_27M_MCLK 0x01 +#define CLK_NO_27M 0x00 +#define CLK_RATIO_SHIFT 1 +#define CLK_R_128 0x00 +#define CLK_R_125 0x01 +#define CLK_R_132 0x02 +#define CLK_R_136 0x03 + +#define CS42L52_IFACE_CTL1 0x06 +#define CS42L52_IFACE_CTL1_MASTER (1 << 7) +#define CS42L52_IFACE_CTL1_SLAVE (0 << 7) +#define CS42L52_IFACE_CTL1_INV_SCLK (1 << 6) +#define CS42L52_IFACE_CTL1_ADC_FMT_I2S (1 << 5) +#define CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J (0 << 5) +#define CS42L52_IFACE_CTL1_DSP_MODE_EN (1 << 4) +#define CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J (0 << 2) +#define CS42L52_IFACE_CTL1_DAC_FMT_I2S (1 << 2) +#define CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J (2 << 2) +#define CS42L52_IFACE_CTL1_WL_32BIT (0x00) +#define CS42L52_IFACE_CTL1_WL_24BIT (0x01) +#define CS42L52_IFACE_CTL1_WL_20BIT (0x02) +#define CS42L52_IFACE_CTL1_WL_16BIT (0x03) +#define CS42L52_IFACE_CTL1_WL_MASK 0xFFFF + +#define CS42L52_IFACE_CTL2 0x07 +#define CS42L52_IFACE_CTL2_SC_MC_EQ (1 << 6) +#define CS42L52_IFACE_CTL2_LOOPBACK (1 << 5) +#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_EN (0 << 4) +#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_HIZ (1 << 4) +#define CS42L52_IFACE_CTL2_HP_SW_INV (1 << 3) + +#define CS42L52_ADC_PGA_A 0x08 +#define CS42L52_ADC_PGA_B 0x09 +#define CS42L52_ADC_SEL_SHIFT 5 +#define CS42L52_ADC_SEL_AIN1 0x00 +#define CS42L52_ADC_SEL_AIN2 0x01 +#define CS42L52_ADC_SEL_AIN3 0x02 +#define CS42L52_ADC_SEL_AIN4 0x03 +#define CS42L52_ADC_SEL_PGA 0x04 + +#define CS42L52_ANALOG_HPF_CTL 0x0A +#define CS42L52_HPF_CTL_ANLGSFTB (1 << 3) +#define CS42L52_HPF_CTL_ANLGSFTA (1 << 0) + +#define CS42L52_ADC_HPF_FREQ 0x0B +#define CS42L52_ADC_MISC_CTL 0x0C +#define CS42L52_ADC_MISC_CTL_SOURCE_DSP (1 << 6) + +#define CS42L52_PB_CTL1 0x0D +#define CS42L52_PB_CTL1_HP_GAIN_SHIFT 5 +#define CS42L52_PB_CTL1_HP_GAIN_03959 0x00 +#define CS42L52_PB_CTL1_HP_GAIN_04571 0x01 +#define CS42L52_PB_CTL1_HP_GAIN_05111 0x02 +#define CS42L52_PB_CTL1_HP_GAIN_06047 0x03 +#define CS42L52_PB_CTL1_HP_GAIN_07099 0x04 +#define CS42L52_PB_CTL1_HP_GAIN_08399 0x05 +#define CS42L52_PB_CTL1_HP_GAIN_10000 0x06 +#define CS42L52_PB_CTL1_HP_GAIN_11430 0x07 +#define CS42L52_PB_CTL1_INV_PCMB (1 << 3) +#define CS42L52_PB_CTL1_INV_PCMA (1 << 2) +#define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1) +#define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0) +#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD + +#define CS42L52_MISC_CTL 0x0E +#define CS42L52_MISC_CTL_DEEMPH (1 << 2) +#define CS42L52_MISC_CTL_DIGSFT (1 << 1) +#define CS42L52_MISC_CTL_DIGZC (1 << 0) + +#define CS42L52_PB_CTL2 0x0F +#define CS42L52_PB_CTL2_HPB_MUTE (1 << 7) +#define CS42L52_PB_CTL2_HPA_MUTE (1 << 6) +#define CS42L52_PB_CTL2_SPKB_MUTE (1 << 5) +#define CS42L52_PB_CTL2_SPKA_MUTE (1 << 4) +#define CS42L52_PB_CTL2_SPK_SWAP (1 << 2) +#define CS42L52_PB_CTL2_SPK_MONO (1 << 1) +#define CS42L52_PB_CTL2_SPK_MUTE50 (1 << 0) + +#define CS42L52_MICA_CTL 0x10 +#define CS42L52_MICB_CTL 0x11 +#define CS42L52_MIC_CTL_SEL_MIC1 (0 << 6) +#define CS42L52_MIC_CTL_SEL_MIC2 (1 << 6) +#define CS42L52_MIC_CTL_SEL_DIFF (1 << 5) + +#define CS42L52_PGAA_CTL 0x12 +#define CS42L52_PGAB_CTL 0x13 +#define CS42L52_PGAX_CTL_VOL_12DB 24 +#define CS42L52_PGAX_CTL_VOL_6DB 12 /*step size 0.5db*/ + +#define CS42L52_PASSTHRUA_VOL 0x14 +#define CS42L52_PASSTHRUB_VOL 0x15 + +#define CS42L52_ADCA_VOL 0x16 +#define CS42L52_ADCB_VOL 0x17 +#define CS42L52_ADCX_VOL_24DB 24 /*step size 1db*/ +#define CS42L52_ADCX_VOL_12DB 12 +#define CS42L52_ADCX_VOL_6DB 6 + +#define CS42L52_ADCA_MIXER_VOL 0x18 +#define CS42L52_ADCB_MIXER_VOL 0x19 +#define CS42L52_ADC_MIXER_VOL_12DB 0x18 + +#define CS42L52_PCMA_MIXER_VOL 0x1A +#define CS42L52_PCMB_MIXER_VOL 0x1B + +#define CS42L52_BEEP_FREQ 0x1C +#define CS42L52_BEEP_VOL 0x1D +#define CS42L52_BEEP_TONE_CTL 0x1E + +#define CS42L52_TONE_CTL 0x1F + +#define CS42L52_MASTERA_VOL 0x20 +#define CS42L52_MASTERB_VOL 0x21 + +#define CS42L52_HPA_VOL 0x22 +#define CS42L52_HPB_VOL 0x23 +#define CS42L52_DEFAULT_HP_VOL 0xF0 + +#define CS42L52_SPKA_VOL 0x24 +#define CS42L52_SPKB_VOL 0x25 +#define CS42L52_DEFAULT_SPK_VOL 0xF0 + +#define CS42L52_ADC_PCM_MIXER 0x26 + +#define CS42L52_LIMITER_CTL1 0x27 +#define CS42L52_LIMITER_CTL2 0x28 +#define CS42L52_LIMITER_AT_RATE 0x29 + +#define CS42L52_ALC_CTL 0x2A +#define CS42L52_ALC_CTL_ALCB_ENABLE (1 << 7) +#define CS42L52_ALC_CTL_ALCA_ENABLE (1 << 6) +#define CS42L52_ALC_CTL_FASTEST_ATTACK 0 + +#define CS42L52_ALC_RATE 0x2B +#define CS42L52_ALC_SLOWEST_RELEASE 0x3F + +#define CS42L52_ALC_THRESHOLD 0x2C +#define CS42L52_ALC_MAX_RATE_SHIFT 5 +#define CS42L52_ALC_MIN_RATE_SHIFT 2 +#define CS42L52_ALC_RATE_0DB 0 +#define CS42L52_ALC_RATE_3DB 1 +#define CS42L52_ALC_RATE_6DB 2 + +#define CS42L52_NOISE_GATE_CTL 0x2D +#define CS42L52_NG_ENABLE (1 << 6) +#define CS42L52_NG_THRESHOLD_SHIFT 2 +#define CS42L52_NG_MIN_70DB 2 +#define CS42L52_NG_DELAY_SHIFT 0 +#define CS42L52_NG_DELAY_100MS 1 + +#define CS42L52_CLK_STATUS 0x2E +#define CS42L52_BATT_COMPEN 0x2F + +#define CS42L52_BATT_LEVEL 0x30 +#define CS42L52_SPK_STATUS 0x31 +#define CS42L52_SPK_STATUS_PIN_SHIFT 3 +#define CS42L52_SPK_STATUS_PIN_HIGH 1 + +#define CS42L52_TEM_CTL 0x32 +#define CS42L52_TEM_CTL_SET 0x80 +#define CS42L52_THE_FOLDBACK 0x33 +#define CS42L52_CHARGE_PUMP 0x34 +#define CS42L52_FIX_BITS1 0x3E +#define CS42L52_FIX_BITS2 0x47 + +#define CS42L52_MAX_REGISTER 0x34 + +#endif -- 1.7.9.1

2 8

[alsa-devel] how to obtain hardware playback position after underrun?
by Maarten de Boer 27 Apr '12

27 Apr '12

Hello, I am trying to figure out how to obtain the hardware playback position after an underrun occured. When I have no underruns, I can simply keep track of the time by counting the samples processed. (I am using snd_pcm_mmap_commit). But in the unlucky event an underrun occurs, I would like to know what the playback position would have been without any underrun. Is this possible? I have been looking at the API docs, and at the source code, but to no avail. Thanks, Maarten

2 1

[alsa-devel] [PATCH] snd-usb-us122l: Fix missing NULL checks
by Karsten Wiese 27 Apr '12

27 Apr '12

Fix missing NULL checks in usb_stream_hwdep_poll() and usb_stream_hwdep_ioctl(). Wake up poll waiters before returning from usb_stream_hwdep_ioctl(). Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Cc: stable(a)kernel.org --- sound/usb/usx2y/us122l.c | 41 ++++++++++++++++++++--------------------- 1 files changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 91bb296..7abda8d 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -269,29 +269,26 @@ static unsigned int usb_stream_hwdep_poll(struct snd_hwdep *hw, struct file *file, poll_table *wait) { struct us122l *us122l = hw->private_data; - struct usb_stream *s = us122l->sk.s; unsigned *polled; unsigned int mask; poll_wait(file, &us122l->sk.sleep, wait); - switch (s->state) { - case usb_stream_ready: - if (us122l->first == file) - polled = &s->periods_polled; - else - polled = &us122l->second_periods_polled; - if (*polled != s->periods_done) { - *polled = s->periods_done; - mask = POLLIN | POLLOUT | POLLWRNORM; - break; + mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; + if (mutex_trylock(&us122l->mutex)) { + struct usb_stream *s = us122l->sk.s; + if (s && s->state == usb_stream_ready) { + if (us122l->first == file) + polled = &s->periods_polled; + else + polled = &us122l->second_periods_polled; + if (*polled != s->periods_done) { + *polled = s->periods_done; + mask = POLLIN | POLLOUT | POLLWRNORM; + } else + mask = 0; } - /* Fall through */ - mask = 0; - break; - default: - mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; - break; + mutex_unlock(&us122l->mutex); } return mask; } @@ -377,6 +374,7 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, { struct usb_stream_config *cfg; struct us122l *us122l = hw->private_data; + struct usb_stream *s; unsigned min_period_frames; int err = 0; bool high_speed; @@ -422,18 +420,18 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, snd_power_wait(hw->card, SNDRV_CTL_POWER_D0); mutex_lock(&us122l->mutex); + s = us122l->sk.s; if (!us122l->master) us122l->master = file; else if (us122l->master != file) { - if (memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg))) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg))) { err = -EIO; goto unlock; } us122l->slave = file; } - if (!us122l->sk.s || - memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg)) || - us122l->sk.s->state == usb_stream_xrun) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg)) || + s->state == usb_stream_xrun) { us122l_stop(us122l); if (!us122l_start(us122l, cfg->sample_rate, cfg->period_frames)) err = -EIO; @@ -444,6 +442,7 @@ unlock: mutex_unlock(&us122l->mutex); free: kfree(cfg); + wake_up_all(&us122l->sk.sleep); return err; } -- 1.7.2.3

2 5

[alsa-devel] [PATCH] ASoC: dapm: Move CODEC<->CODEC params off stack
by Mark Brown 27 Apr '12

27 Apr '12

Reduce our stack consumption by moving the params off the stack, they are reasonably large and might be an issue on platforms with small stacks. Reported-by: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 32 ++++++++++++++++++++------------ 1 file changed, 20 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 75b3ad6..522032e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2987,7 +2987,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_soc_dai *source, *sink; const struct snd_soc_pcm_stream *config = w->params; struct snd_pcm_substream substream; - struct snd_pcm_hw_params params; + struct snd_pcm_hw_params *params = NULL; u64 fmt; int ret; @@ -3017,17 +3017,21 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, } /* Currently very limited parameter selection */ - memset(&params, 0, sizeof(params)); - snd_mask_set(hw_param_mask(&params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); + params = kzalloc(sizeof(*params), GFP_KERNEL); + if (!params) { + ret = -ENOMEM; + goto out; + } + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); - hw_param_interval(&params, SNDRV_PCM_HW_PARAM_RATE)->min = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min = config->rate_min; - hw_param_interval(&params, SNDRV_PCM_HW_PARAM_RATE)->max = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max = config->rate_max; - hw_param_interval(&params, SNDRV_PCM_HW_PARAM_CHANNELS)->min + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min = config->channels_min; - hw_param_interval(&params, SNDRV_PCM_HW_PARAM_CHANNELS)->max + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max = config->channels_max; memset(&substream, 0, sizeof(substream)); @@ -3037,22 +3041,22 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, if (source->driver->ops && source->driver->ops->hw_params) { substream.stream = SNDRV_PCM_STREAM_CAPTURE; ret = source->driver->ops->hw_params(&substream, - &params, source); + params, source); if (ret != 0) { dev_err(source->dev, "hw_params() failed: %d\n", ret); - return ret; + goto out; } } if (sink->driver->ops && sink->driver->ops->hw_params) { substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - ret = sink->driver->ops->hw_params(&substream, &params, + ret = sink->driver->ops->hw_params(&substream, params, sink); if (ret != 0) { dev_err(sink->dev, "hw_params() failed: %d\n", ret); - return ret; + goto out; } } break; @@ -3061,12 +3065,14 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret = snd_soc_dai_digital_mute(sink, 0); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "Failed to unmute: %d\n", ret); + ret = 0; break; case SND_SOC_DAPM_PRE_PMD: ret = snd_soc_dai_digital_mute(sink, 1); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "Failed to mute: %d\n", ret); + ret = 0; break; default: @@ -3074,7 +3080,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, return -EINVAL; } - return 0; +out: + kfree(params); + return ret; } int snd_soc_dapm_new_pcm(struct snd_soc_card *card, -- 1.7.10

2 1

[alsa-devel] [PATCH 6/8] ASoC: Ux500: Add platform-driver
by Ola Lilja 27 Apr '12

27 Apr '12

Add platform-driver handling all DMA-activities. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/ux500/Kconfig | 9 +- sound/soc/ux500/Makefile | 7 +- sound/soc/ux500/ux500_pcm.c | 332 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/ux500_pcm.h | 42 ++++++ 4 files changed, 387 insertions(+), 3 deletions(-) create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index be33409..59da6ce 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -8,9 +8,16 @@ menuconfig SND_SOC_UX500 Say Y if you want to enable ASoC-support for any of the Ux500 platforms (e.g. U8500). -config SND_SOC_UX500_MSP_I2S +config SND_SOC_UX500_PLAT_MSP_I2S tristate "Platform - MSP (I2S)" depends on SND_SOC_UX500 help Say Y if you want to enable the Ux500 MSP I2S-driver. +config SND_SOC_UX500_PLAT_DMA + tristate "Platform - DB8500 (DMA)" + depends on SND_SOC_UX500 + select SND_SOC_DMAENGINE_PCM + help + Say Y if you want to enable the Ux500 platform-driver. + diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 6ffef5c..2e742b1 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -1,5 +1,8 @@ # Ux500 Platform Support -snd-soc-ux500-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o -obj-$(CONFIG_SND_SOC_UX500_MSP_I2S) += snd-soc-ux500-msp-i2s.o +snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o + +snd-soc-ux500-plat-dma-objs := ux500_pcm.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c new file mode 100644 index 0000000..62bcf02 --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.c @@ -0,0 +1,332 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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 <asm/page.h> + +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/slab.h> + +#include <plat/ste_dma40.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "ux500_msp_i2s.h" +#include "ux500_pcm.h" + +static struct snd_pcm_hardware ux500_pcm_hw_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK, + .rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static struct snd_pcm_hardware ux500_pcm_hw_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE, + .rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static void ux500_pcm_dma_hw_free(struct device *dev, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + + if (runtime->dma_area == NULL) + return; + + if (buf != &substream->dma_buffer) { + dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, + buf->addr); + kfree(runtime->dma_buffer_p); + } + + snd_pcm_set_runtime_buffer(substream, NULL); +} + +static int ux500_pcm_open(struct snd_pcm_substream *substream) +{ + int stream_id = substream->pstr->stream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct ux500_pcm *ux500_pcm_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct device *dev = dai->dev; + int ret; + struct ux500_msp_dma_params *dma_params; + u16 per_data_width, mem_data_width; + struct stedma40_chan_cfg *dma_cfg; + + dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_soc_stream_str(substream)); + + dev_dbg(dev, "%s: Set runtime hwparams.\n", __func__); + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_playback); + else + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_capture); + + /* 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) { + dev_err(dev, "%s: Error: snd_pcm_hw_constraints failed (%d)\n", + __func__, ret); + return ret; + } + + dev_dbg(dev, "%s: Init runtime private data.\n", __func__); + ux500_pcm_data = kzalloc(sizeof(struct ux500_pcm), GFP_KERNEL); + if (ux500_pcm_data == NULL) + return -ENOMEM; + ux500_pcm_data->msp_id = dai->id; + + runtime->private_data = ux500_pcm_data; + + dev_dbg(dev, "%s: Set hw-struct for %s.\n", __func__, + snd_soc_stream_str(substream)); + runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ? + ux500_pcm_hw_playback : ux500_pcm_hw_capture; + + mem_data_width = STEDMA40_HALFWORD_WIDTH; + + dma_params = snd_soc_dai_get_dma_data(dai, substream); + switch (dma_params->data_size) { + case 32: + per_data_width = STEDMA40_WORD_WIDTH; + break; + case 16: + per_data_width = STEDMA40_HALFWORD_WIDTH; + break; + case 8: + per_data_width = STEDMA40_BYTE_WIDTH; + break; + default: + per_data_width = STEDMA40_WORD_WIDTH; + dev_warn(rtd->platform->dev, + "%s: Unknown data-size (%d)! Assuming 32 bits.\n", + __func__, dma_params->data_size); + } + + dma_cfg = dma_params->dma_cfg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_cfg->src_info.data_width = mem_data_width; + dma_cfg->dst_info.data_width = per_data_width; + } else { + dma_cfg->src_info.data_width = per_data_width; + dma_cfg->dst_info.data_width = mem_data_width; + } + + + ret = snd_dmaengine_pcm_open(substream, stedma40_filter, dma_cfg); + if (ret) { + dev_dbg(dai->dev, + "%s: ERROR: snd_dmaengine_pcm_open failed (%d)!\n", + __func__, ret); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_cfg); + + return 0; +} + +static int ux500_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct ux500_pcm *ux500_pcm_data = substream->runtime->private_data; + + dev_dbg(dai->dev, "%s: Enter\n", __func__); + + snd_dmaengine_pcm_close(substream); + + kfree(ux500_pcm_data); + + return 0; +} + +static int ux500_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + int size; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + size = params_buffer_bytes(hw_params); + + if (buf) { + if (buf->bytes >= size) + goto out; + ux500_pcm_dma_hw_free(NULL, substream); + } + + if (substream->dma_buffer.area != NULL && + substream->dma_buffer.bytes >= size) { + buf = &substream->dma_buffer; + } else { + buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); + if (!buf) + goto nomem; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = NULL; + buf->area = dma_alloc_coherent(NULL, size, &buf->addr, + GFP_KERNEL); + buf->bytes = size; + buf->private_data = NULL; + + if (!buf->area) + goto free; + } + snd_pcm_set_runtime_buffer(substream, buf); + ret = 1; + out: + runtime->dma_bytes = size; + return ret; + + free: + kfree(buf); + nomem: + return -ENOMEM; +} + +static int ux500_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + ux500_pcm_dma_hw_free(NULL, substream); + + return 0; +} + +static int ux500_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__); + + return dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); +} + +static struct snd_pcm_ops ux500_pcm_ops = { + .open = ux500_pcm_open, + .close = ux500_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ux500_pcm_hw_params, + .hw_free = ux500_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = ux500_pcm_mmap +}; + +int ux500_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__); + + pcm->info_flags = 0; + strcpy(pcm->name, "UX500_PCM"); + dev_dbg(rtd->platform->dev, "%s: PCM-name: %s\n", __func__, pcm->name); + + return 0; +} + +static struct snd_soc_platform_driver ux500_pcm_soc_drv = { + .ops = &ux500_pcm_ops, + .pcm_new = ux500_pcm_new, +}; + +static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev) +{ + int ret; + + ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv); + if (ret < 0) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to register platform '%s' (%d)!\n", + __func__, pdev->name, ret); + return ret; + } + + return 0; +} + +static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver ux500_pcm_driver = { + .driver = { + .name = "ux500-pcm", + .owner = THIS_MODULE, + }, + + .probe = ux500_pcm_drv_probe, + .remove = __devexit_p(ux500_pcm_drv_remove), +}; +module_platform_driver(ux500_pcm_driver); + +MODULE_LICENSE("GPLv2"); + diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h new file mode 100644 index 0000000..19d766a --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.h @@ -0,0 +1,42 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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. + */ +#ifndef UX500_PCM_H +#define UX500_PCM_H + +#include <asm/page.h> + +#include <linux/workqueue.h> + +#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000 +#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000 +#define UX500_PLATFORM_MIN_RATE_CAPTURE 8000 +#define UX500_PLATFORM_MAX_RATE_CAPTURE 48000 + +#define UX500_PLATFORM_MIN_CHANNELS 1 +#define UX500_PLATFORM_MAX_CHANNELS 8 + +#define UX500_PLATFORM_PERIODS_BYTES_MIN 128 +#define UX500_PLATFORM_PERIODS_BYTES_MAX (64 * PAGE_SIZE) +#define UX500_PLATFORM_PERIODS_MIN 2 +#define UX500_PLATFORM_PERIODS_MAX 48 +#define UX500_PLATFORM_BUFFER_BYTES_MAX (2048 * PAGE_SIZE) + +extern struct snd_soc_platform ux500_soc_platform; + +struct ux500_pcm { + int msp_id; + int stream_id; +}; + +#endif -- 1.7.8.3

2 2

Re: [alsa-devel] [PATCH 5/8] ASoC: Ux500: Add MSP I2S-driver
by Mark Brown 27 Apr '12

27 Apr '12

On Fri, Apr 20, 2012 at 11:33:06AM +0200, Ola Lilja wrote: > Add driver for running I2S with the MSP-block. This depends on the change for the debug print function. Otherwise there's a bunch of relatively minor stuff but overall this looks generally good. > +static int ux500_msp_dai_startup(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > + int ret = 0; > + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); > + > + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, > + snd_soc_stream_str(substream)); > + > + /* Don't enable regulator if it's MSP1/3 */ Why not? > +static void ux500_msp_dai_shutdown(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > + if (drvdata->reg_enabled) { > + ret = regulator_disable(drvdata->reg_vape); > + if (ret < 0) > + dev_err(dai->dev, > + "%s: ERROR: Failed to disable regulator (%d)!\n", > + __func__, ret); > + drvdata->reg_enabled = 0; > + } This looks like the driver is going to get confused with bidirectional audio - if one direction stops then it'll turn the regulator off. > + /* Don't enable regulator if it's MSP1/3 */ > + if (!drvdata->reg_enabled && (drvdata->msp->id != MSP_1_I2S_CONTROLLER) > + && (drvdata->msp->id != MSP_3_I2S_CONTROLLER)) { This seems confused, you're enabling in multiple places... > +static int ux500_msp_dai_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + unsigned int mask, slots_active; > + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); > + > + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", > + __func__, dai->id, snd_soc_stream_str(substream)); > + > + switch (drvdata->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { > + case SND_SOC_DAIFMT_I2S: > + if (params_channels(params) != 2) { > + dev_err(dai->dev, > + "%s: Error: I2S requires ch = 2 (ch = %d)!\n", > + __func__, params_channels(params)); > + return -EINVAL; > + } Should really set up constraints for this, though in principle format can change at runtime (though it rarely does). Perhaps we should do something special if the configuration happens on init... > + if (params_channels(params) != slots_active) { > + dev_err(dai->dev, > + "%s: Error: Channels to slots mismatch (ch = %d, slots = %d)!\n", > + __func__, params_channels(params), > + slots_active); > + return -EINVAL; > + } Similarly here. > + drvdata->reg_vape = regulator_get(NULL, "v-ape"); > + if (IS_ERR(drvdata->reg_vape)) { No, this should be using the struct device. Regulators should always be requested in the context of their consumer. You could use devm_regulator_get() too (there's a clock one too, but only in -next). > + ret = ux500_msp_i2s_init_msp(pdev, &drvdata->msp, platform_data); > + if (!drvdata->msp) { Should be no need to cast away from void. > + > + ux500_msp_i2s_cleanup_msp(pdev, drvdata->msp); > + devm_kfree(&pdev->dev, drvdata); You're missing the point of devm_ here! :)

2 1

[alsa-devel] [PATCH 00/11] enable imx6q_sabrelite sgtl5000 audio support
by Richard Zhao 27 Apr '12

27 Apr '12

It's based on Dong Aisheng's early pinctrl patch and Sascha's v2 convert to common clk patch. You can also get at: https://github.com/riczhao/kernel-imx/tree/topics/audio Richard Zhao (11): dma: imx-sdma: make channel0 operations atomic ASoC: imx-sgtl5000: add of_node_put when probe fail. ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000 i2c: imx: add pinctrl support ARM: imx6q: move imx6q_sabrelite specific code to a dedicated function ARM: dts: imx6q-sabrelite: add ssi device ARM: dts: imx6q-sabrelite: add audmux device ASoC: imx-audmux: add pinctrl support ARM: imx6q: add ssi1 clk_lookup ARM: imx6q_sabrelite: clkdev_add cko1 for sgtl5000 ARM: dts: imx6q-sabrelite: add sound device imx6q-sabrelite-sgtl5000 arch/arm/boot/dts/imx6q-sabrelite.dts | 30 ++++++++++++++++ arch/arm/boot/dts/imx6q.dtsi | 46 +++++++++++++++++++++++-- arch/arm/mach-imx/clk-imx6q.c | 4 ++ arch/arm/mach-imx/mach-imx6q.c | 37 +++++++++++++++++++- drivers/dma/imx-sdma.c | 57 ++++++++++++++++++------------- drivers/i2c/busses/i2c-imx.c | 9 +++++ sound/soc/fsl/imx-audmux.c | 8 ++++ sound/soc/fsl/imx-sgtl5000.c | 61 +++++++++++++++++++++++++-------- 8 files changed, 208 insertions(+), 44 deletions(-) Thanks Richard -- 1.7.5.4

1 11

[alsa-devel] [PATCH] ASoC: dapm: Dynamically move CODEC<->CODEC params off stack
by Mark Brown 26 Apr '12

26 Apr '12

Reduce our stack consumption by moving the params off the stack, they are reasonably large and might be an issue on platforms with small stacks. Reported-by: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 32 ++++++++++++++++++++------------ 1 file changed, 20 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 75b3ad6..522032e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2987,7 +2987,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_soc_dai *source, *sink; const struct snd_soc_pcm_stream *config = w->params; struct snd_pcm_substream substream; - struct snd_pcm_hw_params params; + struct snd_pcm_hw_params *params = NULL; u64 fmt; int ret; @@ -3017,17 +3017,21 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, } /* Currently very limited parameter selection */ - memset(&params, 0, sizeof(params)); - snd_mask_set(hw_param_mask(&params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); + params = kzalloc(sizeof(*params), GFP_KERNEL); + if (!params) { + ret = -ENOMEM; + goto out; + } + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); - hw_param_interval(&params, SNDRV_PCM_HW_PARAM_RATE)->min = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min = config->rate_min; - hw_param_interval(&params, SNDRV_PCM_HW_PARAM_RATE)->max = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max = config->rate_max; - hw_param_interval(&params, SNDRV_PCM_HW_PARAM_CHANNELS)->min + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min = config->channels_min; - hw_param_interval(&params, SNDRV_PCM_HW_PARAM_CHANNELS)->max + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max = config->channels_max; memset(&substream, 0, sizeof(substream)); @@ -3037,22 +3041,22 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, if (source->driver->ops && source->driver->ops->hw_params) { substream.stream = SNDRV_PCM_STREAM_CAPTURE; ret = source->driver->ops->hw_params(&substream, - &params, source); + params, source); if (ret != 0) { dev_err(source->dev, "hw_params() failed: %d\n", ret); - return ret; + goto out; } } if (sink->driver->ops && sink->driver->ops->hw_params) { substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - ret = sink->driver->ops->hw_params(&substream, &params, + ret = sink->driver->ops->hw_params(&substream, params, sink); if (ret != 0) { dev_err(sink->dev, "hw_params() failed: %d\n", ret); - return ret; + goto out; } } break; @@ -3061,12 +3065,14 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret = snd_soc_dai_digital_mute(sink, 0); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "Failed to unmute: %d\n", ret); + ret = 0; break; case SND_SOC_DAPM_PRE_PMD: ret = snd_soc_dai_digital_mute(sink, 1); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "Failed to mute: %d\n", ret); + ret = 0; break; default: @@ -3074,7 +3080,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, return -EINVAL; } - return 0; +out: + kfree(params); + return ret; } int snd_soc_dapm_new_pcm(struct snd_soc_card *card, -- 1.7.10

1 0

[alsa-devel] [PATCH 1/5] ASoC: dpcm: Add Dynamic PCM core operations.
by Liam Girdwood 26 Apr '12

26 Apr '12

The Dynamic PCM core allows digital audio data to be dynamically routed between different ALSA PCMs and DAI links on SoC CPUs with on chip DSP devices. e.g. audio data could be played on pcm:0,0 and routed to any (or all) SoC DAI links. Dynamic PCM introduces the concept of Front End (FE) PCMs and Back End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that they can dynamically route digital audio data to any supported BE PCM. A BE PCM has no ALSA device, but represents a DAI link and it's substream and audio HW parameters. e.g. pcm:0,0 routing digital data to 2 external codecs. FE pcm:0,0 ----> BE (McBSP.0) ----> CODEC 0 +--> BE (McPDM.0) ----> CODEC 1 e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec. FE pcm:0,0 --- +--> BE (McBSP.0) ----> CODEC FE pcm:0,1 --- The digital audio routing is controlled by the usual ALSA method of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the routing based upon the mixer settings and configures the BE PCMs based on routing and the FE HW params. DPCM is designed so that most ASoC component drivers will need no modification at all. It's intended that existing CODEC, DAI and platform drivers can be used in DPCM based audio devices without any changes. However, there will be some cases where minor changes are required (e.g. for very tightly coupled HW) and there are helpers to support this too. Somethimes the HW params of a FE and BE do not match or are incompatible, so in these cases the machine driver can reconfigure any hw_params and make any DSP perform sample rate / format conversion. This patch adds the core DPCM code and contains :- o The FE and BE PCM operations. o FE and BE DAI link support. o FE and BE PCM creation. o BE support API. o BE and FE link management. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dpcm.h | 132 +++++ include/sound/soc.h | 17 + sound/soc/soc-core.c | 5 + sound/soc/soc-pcm.c | 1246 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 1370 insertions(+), 30 deletions(-) create mode 100644 include/sound/soc-dpcm.h diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h new file mode 100644 index 0000000..cd72c10 --- /dev/null +++ b/include/sound/soc-dpcm.h @@ -0,0 +1,132 @@ +/* + * linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support + * + * Author: Liam Girdwood <lrg(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 version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_SOC_DPCM_H +#define __LINUX_SND_SOC_DPCM_H + +#include <linux/list.h> +#include <sound/pcm.h> + +struct snd_soc_pcm_runtime; + +/* + * Types of runtime_update to perform. e.g. originated from FE PCM ops + * or audio route changes triggered by muxes/mixers. + */ +enum snd_soc_dpcm_update { + SND_SOC_DPCM_UPDATE_NO = 0, + SND_SOC_DPCM_UPDATE_BE, + SND_SOC_DPCM_UPDATE_FE, +}; + +/* + * Dynamic PCM Frontend -> Backend link management states. + */ +enum snd_soc_dpcm_link_state { + SND_SOC_DPCM_LINK_STATE_NEW = 0, /* newly created link */ + SND_SOC_DPCM_LINK_STATE_FREE, /* link to be dismantled */ +}; + +/* + * Dynamic PCM Frontend -> Backend link PCM states. + */ +enum snd_soc_dpcm_state { + SND_SOC_DPCM_STATE_NEW = 0, + SND_SOC_DPCM_STATE_OPEN, + SND_SOC_DPCM_STATE_HW_PARAMS, + SND_SOC_DPCM_STATE_PREPARE, + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_STOP, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_SUSPEND, + SND_SOC_DPCM_STATE_HW_FREE, + SND_SOC_DPCM_STATE_CLOSE, +}; + +/* + * Dynamic PCM trigger ordering. Triggering flexibility is required as some + * DSPs require triggering before/after their CPU platform and DAIs. + * + * i.e. some clients may want to manually order this call in their PCM + * trigger() whilst others will just use the regular core ordering. + */ +enum snd_soc_dpcm_trigger { + SND_SOC_DPCM_TRIGGER_PRE = 0, + SND_SOC_DPCM_TRIGGER_POST, +}; + +/* + * Dynamic PCM link + * This links together a FE and BE DAI at runtime and stores the link + * state information and the hw_params configuration. + */ +struct snd_soc_dpcm { + /* FE and BE DAIs*/ + struct snd_soc_pcm_runtime *be; + struct snd_soc_pcm_runtime *fe; + + /* link state */ + enum snd_soc_dpcm_link_state state; + + /* list of BE and FE for this DPCM link */ + struct list_head list_be; + struct list_head list_fe; + + /* hw params for this link - may be different for each link */ + struct snd_pcm_hw_params hw_params; +}; + +/* + * Dynamic PCM runtime data. + */ +struct snd_soc_dpcm_runtime { + struct list_head be_clients; + struct list_head fe_clients; + + int users; + struct snd_pcm_runtime *runtime; + struct snd_pcm_hw_params hw_params; + + /* state and update */ + enum snd_soc_dpcm_update runtime_update; + enum snd_soc_dpcm_state state; +}; + +/* can this BE stop and free */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + +/* can this BE perform a hw_params() */ +int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + +/* is the current PCM operation for this FE ? */ +int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); + +/* is the current PCM operation for this BE ? */ +int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + +/* get the substream for this BE */ +struct snd_pcm_substream * + snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream); + +/* get the BE runtime state */ +enum snd_soc_dpcm_state + snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream); + +/* set the BE runtime state */ +void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream, + enum snd_soc_dpcm_state state); + +/* internal use only */ +int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); + +#endif diff --git a/include/sound/soc.h b/include/sound/soc.h index afc3204..c2eb563 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -274,6 +274,7 @@ struct snd_soc_jack_zone; struct snd_soc_jack_pin; struct snd_soc_cache_ops; #include <sound/soc-dapm.h> +#include <sound/soc-dpcm.h> #ifdef CONFIG_GPIOLIB struct snd_soc_jack_gpio; @@ -760,23 +761,36 @@ struct snd_soc_dai_link { const char *cpu_dai_name; const struct device_node *cpu_dai_of_node; const char *codec_dai_name; + int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; unsigned int dai_fmt; /* format to set on init */ + enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ + /* Keep DAI active over suspend */ unsigned int ignore_suspend:1; /* Symmetry requirements */ unsigned int symmetric_rates:1; + /* Do not create a PCM for this DAI link (Backend link) */ + unsigned int no_pcm:1; + + /* This DAI link can route to other DAI links at runtime (Frontend)*/ + unsigned int dynamic:1; + /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_pcm_runtime *rtd); + /* optional hw_params re-writing for BE and FE sync */ + int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + /* machine stream operations */ struct snd_soc_ops *ops; }; @@ -908,6 +922,9 @@ struct snd_soc_pcm_runtime { unsigned int dev_registered:1; + /* Dynamic PCM BE runtime data */ + struct snd_soc_dpcm_runtime dpcm[2]; + long pmdown_time; /* runtime devices */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5d308e9..1a2626d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -39,6 +39,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-dpcm.h> #include <sound/initval.h> #define CREATE_TRACE_POINTS @@ -1161,6 +1162,10 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd->dev->init_name = name; dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); ret = device_add(rtd->dev); if (ret < 0) { dev_err(card->dev, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 26a60b4..458f116 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -22,12 +22,37 @@ #include <linux/pm_runtime.h> #include <linux/slab.h> #include <linux/workqueue.h> +#include <linux/export.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-dpcm.h> #include <sound/initval.h> +#define DPCM_MAX_BE_USERS 8 + +/* 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) +{ + struct snd_soc_dpcm *dpcm; + + list_for_each_entry(dpcm, &fe->dpcm[dir].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + + dev_dbg(be->dev, "pm: BE %s event %d dir %d\n", + be->dai_link->name, event, dir); + + snd_soc_dapm_stream_event(be, dir, event); + } + + snd_soc_dapm_stream_event(fe, dir, event); + + return 0; +} + static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, struct snd_soc_dai *soc_dai) { @@ -156,6 +181,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } + /* Dynamic PCM DAI links compat checks use dynamic capabilities */ + if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) + goto dynamic; + /* Check that the codec and cpu DAIs are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { runtime->hw.rate_min = @@ -248,6 +277,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, runtime->hw.rate_max); +dynamic: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; codec_dai->playback_active++; @@ -633,6 +663,1025 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } +/* connect a FE and BE */ +static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* only add new dpcms */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + if (dpcm->be == be && dpcm->fe == fe) + return 0; + } + + dpcm = kzalloc(sizeof(struct snd_soc_dpcm), GFP_KERNEL); + if (!dpcm) + return -ENOMEM; + + dpcm->be = be; + dpcm->fe = fe; + be->dpcm[stream].runtime = fe->dpcm[stream].runtime; + dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; + list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); + list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); + + dev_dbg(fe->dev, " connected new DPCM %s path %s %s %s\n", + stream ? "capture" : "playback", fe->dai_link->name, + stream ? "<-" : "->", be->dai_link->name); + + return 1; +} + +/* reparent a BE onto another FE */ +static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + struct snd_pcm_substream *fe_substream, *be_substream; + + /* reparent if BE is connected to other FEs */ + if (!be->dpcm[stream].users) + return; + + be_substream = snd_soc_dpcm_get_substream(be, stream); + + list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + if (dpcm->fe == fe) + continue; + + dev_dbg(fe->dev, " reparent %s path %s %s %s\n", + stream ? "capture" : "playback", + dpcm->fe->dai_link->name, + stream ? "<-" : "->", dpcm->be->dai_link->name); + + fe_substream = snd_soc_dpcm_get_substream(dpcm->fe, stream); + be_substream->runtime = fe_substream->runtime; + break; + } +} + +/* disconnect a BE and FE */ +static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm, *d; + + list_for_each_entry_safe(dpcm, d, &fe->dpcm[stream].be_clients, list_be) { + dev_dbg(fe->dev, "BE %s disconnect check for %s\n", + stream ? "capture" : "playback", + dpcm->be->dai_link->name); + + if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE) + continue; + + dev_dbg(fe->dev, " freed DSP %s path %s %s %s\n", + stream ? "capture" : "playback", fe->dai_link->name, + stream ? "<-" : "->", dpcm->be->dai_link->name); + + /* BEs still alive need new FE */ + dpcm_be_reparent(fe, dpcm->be, stream); + + list_del(&dpcm->list_be); + list_del(&dpcm->list_fe); + kfree(dpcm); + } +} + +/* get BE for DAI widget and stream */ +static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, + struct snd_soc_dapm_widget *widget, int stream) +{ + struct snd_soc_pcm_runtime *be; + int i; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < card->num_links; i++) { + be = &card->rtd[i]; + + if (be->cpu_dai->playback_widget == widget || + be->codec_dai->playback_widget == widget) + return be; + } + } else { + + for (i = 0; i < card->num_links; i++) { + be = &card->rtd[i]; + + if (be->cpu_dai->capture_widget == widget || + be->codec_dai->capture_widget == widget) + return be; + } + } + + dev_err(card->dev, "can't get %s BE for %s\n", + stream ? "capture" : "playback", widget->name); + return NULL; +} + +static inline struct snd_soc_dapm_widget * + rtd_get_cpu_widget(struct snd_soc_pcm_runtime *rtd, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + return rtd->cpu_dai->playback_widget; + else + return rtd->cpu_dai->capture_widget; +} + +static inline struct snd_soc_dapm_widget * + rtd_get_codec_widget(struct snd_soc_pcm_runtime *rtd, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + return rtd->codec_dai->playback_widget; + else + return rtd->codec_dai->capture_widget; +} + +static int widget_in_list(struct snd_soc_dapm_widget_list *list, + struct snd_soc_dapm_widget *widget) +{ + int i; + + for (i = 0; i < list->num_widgets; i++) { + if (widget == list->widgets[i]) + return 1; + } + + return 0; +} + +static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list_) +{ + struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dapm_widget_list *list; + int paths; + + list = kzalloc(sizeof(struct snd_soc_dapm_widget_list) + + sizeof(struct snd_soc_dapm_widget *), GFP_KERNEL); + if (list == NULL) + return -ENOMEM; + + /* get number of valid DAI paths and their widgets */ + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, &list); + + dev_dbg(fe->dev, "found %d audio %s paths\n", paths, + stream ? "capture" : "playback"); + + *list_ = list; + return paths; +} + +static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + +static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, + struct snd_soc_dapm_widget_list **list_) +{ + struct snd_soc_dpcm *dpcm; + struct snd_soc_dapm_widget_list *list = *list_; + struct snd_soc_dapm_widget *widget; + int prune = 0; + + /* Destroy any old FE <--> BE connections */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + /* is there a valid CPU DAI widget for this BE */ + widget = rtd_get_cpu_widget(dpcm->be, stream); + + /* prune the BE if it's no longer in our active list */ + if (widget && widget_in_list(list, widget)) + continue; + + /* is there a valid CODEC DAI widget for this BE */ + widget = rtd_get_codec_widget(dpcm->be, stream); + + /* prune the BE if it's no longer in our active list */ + if (widget && widget_in_list(list, widget)) + continue; + + dev_dbg(fe->dev, "pruning %s BE %s for %s\n", + stream ? "capture" : "playback", + dpcm->be->dai_link->name, fe->dai_link->name); + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + prune++; + } + + dev_dbg(fe->dev, "found %d old BE paths for pruning\n", prune); + return prune; +} + +static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, + struct snd_soc_dapm_widget_list **list_) +{ + struct snd_soc_card *card = fe->card; + struct snd_soc_dapm_widget_list *list = *list_; + struct snd_soc_pcm_runtime *be; + int i, new = 0, err; + + /* Create any new FE <--> BE connections */ + for (i = 0; i < list->num_widgets; i++) { + + if (list->widgets[i]->id != snd_soc_dapm_dai) + continue; + + /* is there a valid BE rtd for this widget */ + be = dpcm_get_be(card, list->widgets[i], stream); + if (!be) { + dev_err(fe->dev, "no BE found for %s\n", + list->widgets[i]->name); + continue; + } + + /* make sure BE is a real BE */ + if (!be->dai_link->no_pcm) + continue; + + /* don't connect if FE is not running */ + if (!fe->dpcm[stream].runtime) + continue; + + /* newly connected FE and BE */ + err = dpcm_be_connect(fe, be, stream); + if (err < 0) { + dev_err(fe->dev, "can't connect %s\n", + list->widgets[i]->name); + break; + } else if (err == 0) /* already connected */ + continue; + + /* new */ + be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + new++; + } + + dev_dbg(fe->dev, "found %d new BE paths\n", new); + return new; +} + +/* + * Find the corresponding BE DAIs that source or sink audio to this + * FE substream. + */ +static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list, int new) +{ + if (new) + return dpcm_add_paths(fe, stream, list); + else + return dpcm_prune_paths(fe, stream, list); +} + +static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->be->dpcm[stream].runtime_update = + SND_SOC_DPCM_UPDATE_NO; +} + +static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, + int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* disable any enabled and non active backends */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + if (be->dpcm[stream].users == 0) + dev_err(be->dev, "no users %s at close - state %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (--be->dpcm[stream].users != 0) + continue; + + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) + continue; + + soc_pcm_close(be_substream); + be_substream->runtime = NULL; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + } +} + +static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + int err, count = 0; + + /* only startup BE DAIs that are either sinks or sources to this FE DAI */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* first time the dpcm is open ? */ + if (be->dpcm[stream].users == DPCM_MAX_BE_USERS) + dev_err(be->dev, "too many users %s at open %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (be->dpcm[stream].users++ != 0) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_NEW) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE)) + continue; + + dev_dbg(be->dev, "dpcm: open BE %s\n", be->dai_link->name); + + be_substream->runtime = be->dpcm[stream].runtime; + err = soc_pcm_open(be_substream); + if (err < 0) { + dev_err(be->dev, "BE open failed %d\n", err); + be->dpcm[stream].users--; + if (be->dpcm[stream].users < 0) + dev_err(be->dev, "no users %s at unwind %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + goto unwind; + } + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; + count++; + } + + return count; + +unwind: + /* disable any enabled and non active backends */ + list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + if (be->dpcm[stream].users == 0) + dev_err(be->dev, "no users %s at close %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (--be->dpcm[stream].users != 0) + continue; + + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) + continue; + + soc_pcm_close(be_substream); + be_substream->runtime = NULL; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + } + + return err; +} + +void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + runtime->hw.rate_min = cpu_dai_drv->playback.rate_min; + runtime->hw.rate_max = cpu_dai_drv->playback.rate_max; + runtime->hw.channels_min = cpu_dai_drv->playback.channels_min; + runtime->hw.channels_max = cpu_dai_drv->playback.channels_max; + runtime->hw.formats &= cpu_dai_drv->playback.formats; + runtime->hw.rates = cpu_dai_drv->playback.rates; + } else { + runtime->hw.rate_min = cpu_dai_drv->capture.rate_min; + runtime->hw.rate_max = cpu_dai_drv->capture.rate_max; + runtime->hw.channels_min = cpu_dai_drv->capture.channels_min; + runtime->hw.channels_max = cpu_dai_drv->capture.channels_max; + runtime->hw.formats &= cpu_dai_drv->capture.formats; + runtime->hw.rates = cpu_dai_drv->capture.rates; + } +} + +static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_pcm_runtime *runtime = fe_substream->runtime; + int stream = fe_substream->stream, ret = 0; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_startup(fe, fe_substream->stream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: failed to start some BEs %d\n", ret); + goto be_err; + } + + dev_dbg(fe->dev, "dpcm: open FE %s\n", fe->dai_link->name); + + /* start the DAI frontend */ + ret = soc_pcm_open(fe_substream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: failed to start FE %d\n", ret); + goto unwind; + } + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; + + dpcm_set_fe_runtime(fe_substream); + snd_pcm_limit_hw_rates(runtime); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return 0; + +unwind: + dpcm_be_dai_startup_unwind(fe, fe_substream->stream); +be_err: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return ret; +} + +static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* only shutdown BEs that are either sinks or sources to this FE DAI */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + if (be->dpcm[stream].users == 0) + dev_err(be->dev, "no users %s at close - state %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (--be->dpcm[stream].users != 0) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN)) + continue; + + dev_dbg(be->dev, "dpcm: close BE %s\n", + dpcm->fe->dai_link->name); + + soc_pcm_close(be_substream); + be_substream->runtime = NULL; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + } + return 0; +} + +static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + /* shutdown the BEs */ + dpcm_be_dai_shutdown(fe, substream->stream); + + dev_dbg(fe->dev, "dpcm: close FE %s\n", fe->dai_link->name); + + /* now shutdown the frontend */ + soc_pcm_close(substream); + + /* run the stream event for each BE */ + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return 0; +} + +static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* only hw_params backends that are either sinks or sources + * to this frontend DAI */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* only free hw when no longer used - check all FEs */ + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + dev_dbg(be->dev, "dpcm: hw_free BE %s\n", + dpcm->fe->dai_link->name); + + soc_pcm_hw_free(be_substream); + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; + } + + return 0; +} + +int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int err, stream = substream->stream; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + dev_dbg(fe->dev, "dpcm: hw_free FE %s\n", fe->dai_link->name); + + /* call hw_free on the frontend */ + err = soc_pcm_hw_free(substream); + if (err < 0) + dev_err(fe->dev,"dpcm: hw_free FE %s failed\n", + fe->dai_link->name); + + /* only hw_params backends that are either sinks or sources + * to this frontend DAI */ + err = dpcm_be_dai_hw_free(fe, stream); + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + + mutex_unlock(&fe->card->mutex); + return 0; +} + +static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + int ret; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* only allow hw_params() if no connected FEs are running */ + if (!snd_soc_dpcm_can_be_params(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE)) + continue; + + dev_dbg(be->dev, "dpcm: hw_params BE %s\n", + dpcm->fe->dai_link->name); + + /* copy params for each dpcm */ + memcpy(&dpcm->hw_params, &fe->dpcm[stream].hw_params, + sizeof(struct snd_pcm_hw_params)); + + /* perform any hw_params fixups */ + if (be->dai_link->be_hw_params_fixup) { + ret = be->dai_link->be_hw_params_fixup(be, + &dpcm->hw_params); + if (ret < 0) { + dev_err(be->dev, + "dpcm: hw_params BE fixup failed %d\n", + ret); + goto unwind; + } + } + + ret = soc_pcm_hw_params(be_substream, &dpcm->hw_params); + if (ret < 0) { + dev_err(dpcm->be->dev, + "dpcm: hw_params BE failed %d\n", ret); + goto unwind; + } + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS; + } + return 0; + +unwind: + /* disable any enabled and non active backends */ + list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* only allow hw_free() if no connected FEs are running */ + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + soc_pcm_hw_free(be_substream); + } + + return ret; +} + +int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int ret, stream = substream->stream; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + memcpy(&fe->dpcm[substream->stream].hw_params, params, + sizeof(struct snd_pcm_hw_params)); + ret = dpcm_be_dai_hw_params(fe, substream->stream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: hw_params BE failed %d\n", ret); + goto out; + } + + dev_dbg(fe->dev, "dpcm: hw_params FE %s rate %d chan %x fmt %d\n", + fe->dai_link->name, params_rate(params), + params_channels(params), params_format(params)); + + /* call hw_params on the frontend */ + ret = soc_pcm_hw_params(substream, params); + if (ret < 0) { + dev_err(fe->dev,"dpcm: hw_params FE failed %d\n", ret); + dpcm_be_dai_hw_free(fe, stream); + } else + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS; + +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_unlock(&fe->card->mutex); + return ret; +} + +static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, + struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + dev_dbg(dpcm->be->dev, "dpcm: trigger BE %s cmd %d\n", + dpcm->fe->dai_link->name, cmd); + + ret = soc_pcm_trigger(substream, cmd); + if (ret < 0) + dev_err(dpcm->be->dev,"dpcm: trigger BE failed %d\n", ret); + + return ret; +} + +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) +{ + struct snd_soc_dpcm *dpcm; + int ret = 0; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_RESUME: + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_STOP: + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + continue; + + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) + continue; + + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + continue; + + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; + break; + } + } + + return ret; +} +EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); + +int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream, ret; + enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + switch (trigger) { + case SND_SOC_DPCM_TRIGGER_PRE: + /* call trigger on the frontend before the backend. */ + + dev_dbg(fe->dev, "dpcm: pre trigger FE %s cmd %d\n", + fe->dai_link->name, cmd); + + ret = soc_pcm_trigger(substream, cmd); + if (ret < 0) { + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret); + goto out; + } + + ret = dpcm_be_dai_trigger(fe, substream->stream, cmd); + break; + case SND_SOC_DPCM_TRIGGER_POST: + /* call trigger on the frontend after the backend. */ + + ret = dpcm_be_dai_trigger(fe, substream->stream, cmd); + if (ret < 0) { + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret); + goto out; + } + + dev_dbg(fe->dev, "dpcm: post trigger FE %s cmd %d\n", + fe->dai_link->name, cmd); + + ret = soc_pcm_trigger(substream, cmd); + break; + default: + dev_err(fe->dev, "dpcm: invalid trigger cmd %d for %s\n", cmd, + fe->dai_link->name); + ret = -EINVAL; + goto out; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; + break; + } + +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return ret; +} + +static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + int ret = 0; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + dev_dbg(be->dev, "dpcm: prepare BE %s\n", + dpcm->fe->dai_link->name); + + ret = soc_pcm_prepare(be_substream); + if (ret < 0) { + dev_err(be->dev, "dpcm: backend prepare failed %d\n", + ret); + break; + } + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; + } + return ret; +} + +int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream, ret = 0; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + + dev_dbg(fe->dev, "dpcm: prepare FE %s\n", fe->dai_link->name); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + /* there is no point preparing this FE if there are no BEs */ + if (list_empty(&fe->dpcm[stream].be_clients)) { + dev_err(fe->dev, "dpcm: no backend DAIs enabled for %s\n", + fe->dai_link->name); + ret = -EINVAL; + goto out; + } + + ret = dpcm_be_dai_prepare(fe, substream->stream); + if (ret < 0) + goto out; + + /* call prepare on the frontend */ + ret = soc_pcm_prepare(substream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: prepare FE %s failed\n", + fe->dai_link->name); + goto out; + } + + /* run the stream event for each BE */ + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; + +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_unlock(&fe->card->mutex); + + return ret; +} + + +int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) +{ + struct snd_soc_dpcm *dpcm; + struct list_head *clients = + &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients; + + list_for_each_entry(dpcm, clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai *dai = be->codec_dai; + struct snd_soc_dai_driver *drv = dai->driver; + + if (be->dai_link->ignore_suspend) + continue; + + dev_dbg(be->dev, "BE digital mute %s\n", be->dai_link->name); + + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, mute); + } + + return 0; +} + +int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_soc_dpcm *dpcm; + struct snd_soc_dapm_widget_list *list; + int ret; + int stream = fe_substream->stream; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime = fe_substream->runtime; + + if (dpcm_path_get(fe, stream, &list) <= 0) { + dev_warn(fe->dev, "asoc: %s no valid %s route\n", + fe->dai_link->name, stream ? "capture" : "playback"); + mutex_unlock(&fe->card->mutex); + return -EINVAL; + } + + /* calculate valid and active FE <-> BE dpcms */ + dpcm_process_paths(fe, stream, &list, 1); + + ret = dpcm_fe_dai_startup(fe_substream); + if (ret < 0) { + /* clean up all links */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + fe->dpcm[stream].runtime = NULL; + } + + dpcm_clear_pending_state(fe, stream); + dpcm_path_put(&list); + mutex_unlock(&fe->card->mutex); + return ret; +} + +int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = fe_substream->stream, ret; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + ret = dpcm_fe_dai_shutdown(fe_substream); + + /* mark FE's links ready to prune */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + + fe->dpcm[stream].runtime = NULL; + mutex_unlock(&fe->card->mutex); + return ret; +} + /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -640,56 +1689,92 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_pcm_ops *soc_pcm_ops = &rtd->ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; - soc_pcm_ops->open = soc_pcm_open; - soc_pcm_ops->close = soc_pcm_close; - soc_pcm_ops->hw_params = soc_pcm_hw_params; - soc_pcm_ops->hw_free = soc_pcm_hw_free; - soc_pcm_ops->prepare = soc_pcm_prepare; - soc_pcm_ops->trigger = soc_pcm_trigger; - soc_pcm_ops->pointer = soc_pcm_pointer; - - /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - - if (codec_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min) - capture = 1; - - dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); - ret = snd_pcm_new(rtd->card->snd_card, new_name, - num, playback, capture, &pcm); + if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { + if (cpu_dai->driver->playback.channels_min) + playback = 1; + if (cpu_dai->driver->capture.channels_min) + capture = 1; + } else { + if (codec_dai->driver->playback.channels_min) + playback = 1; + if (codec_dai->driver->capture.channels_min) + capture = 1; + } + + /* create the PCM */ + if (rtd->dai_link->no_pcm) { + snprintf(new_name, sizeof(new_name), "(%s)", + rtd->dai_link->stream_name); + + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, + playback, capture, &pcm); + } else { + if (rtd->dai_link->dynamic) + snprintf(new_name, sizeof(new_name), "%s (*)", + rtd->dai_link->stream_name); + else + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + + ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, + capture, &pcm); + } if (ret < 0) { printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); return ret; } + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name); /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); rtd->pcm = pcm; pcm->private_data = rtd; + + if (rtd->dai_link->no_pcm) { + if (playback) + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + if (capture) + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + goto out; + } + + /* ASoC PCM operations */ + if (rtd->dai_link->dynamic) { + rtd->ops.open = dpcm_fe_dai_open; + rtd->ops.hw_params = dpcm_fe_dai_hw_params; + rtd->ops.prepare = dpcm_fe_dai_prepare; + rtd->ops.trigger = dpcm_fe_dai_trigger; + rtd->ops.hw_free = dpcm_fe_dai_hw_free; + rtd->ops.close = dpcm_fe_dai_close; + rtd->ops.pointer = soc_pcm_pointer; + } else { + rtd->ops.open = soc_pcm_open; + rtd->ops.hw_params = soc_pcm_hw_params; + rtd->ops.prepare = soc_pcm_prepare; + rtd->ops.trigger = soc_pcm_trigger; + rtd->ops.hw_free = soc_pcm_hw_free; + rtd->ops.close = soc_pcm_close; + rtd->ops.pointer = soc_pcm_pointer; + } + if (platform->driver->ops) { - soc_pcm_ops->mmap = platform->driver->ops->mmap; - soc_pcm_ops->pointer = platform->driver->ops->pointer; - soc_pcm_ops->ioctl = platform->driver->ops->ioctl; - soc_pcm_ops->copy = platform->driver->ops->copy; - soc_pcm_ops->silence = platform->driver->ops->silence; - soc_pcm_ops->ack = platform->driver->ops->ack; - soc_pcm_ops->page = platform->driver->ops->page; + rtd->ops.ack = platform->driver->ops->ack; + rtd->ops.copy = platform->driver->ops->copy; + rtd->ops.silence = platform->driver->ops->silence; + rtd->ops.page = platform->driver->ops->page; + rtd->ops.mmap = platform->driver->ops->mmap; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &rtd->ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); @@ -700,7 +1785,108 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) } pcm->private_free = platform->driver->pcm_free; +out: printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } + +/* is the current PCM operation for this FE ? */ +int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream) +{ + if (fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) + return 1; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_fe_can_update); + +/* is the current PCM operation for this BE ? */ +int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + if ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) || + ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_BE) && + be->dpcm[stream].runtime_update)) + return 1; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_can_update); + +/* get the substream for this BE */ +struct snd_pcm_substream * + snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream) +{ + return be->pcm->streams[stream].substream; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream); + +/* get the BE runtime state */ +enum snd_soc_dpcm_state + snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream) +{ + return be->dpcm[stream].state; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_get_state); + +/* set the BE runtime state */ +void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, + int stream, enum snd_soc_dpcm_state state) +{ + be->dpcm[stream].state = state; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_set_state); + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + int state; + + list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + + if (dpcm->fe == fe) + continue; + + state = dpcm->fe->dpcm[stream].state; + if (state == SND_SOC_DPCM_STATE_START || + state == SND_SOC_DPCM_STATE_PAUSED || + state == SND_SOC_DPCM_STATE_SUSPEND) + return 0; + } + + /* it's safe to free/stop this BE DAI */ + return 1; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); + +/* + * We can only change hw params a BE DAI if any of it's FE are not prepared, + * running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + int state; + + list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + + if (dpcm->fe == fe) + continue; + + state = dpcm->fe->dpcm[stream].state; + if (state == SND_SOC_DPCM_STATE_START || + state == SND_SOC_DPCM_STATE_PAUSED || + state == SND_SOC_DPCM_STATE_SUSPEND || + state == SND_SOC_DPCM_STATE_PREPARE) + return 0; + } + + /* it's safe to change hw_params */ + return 1; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -- 1.7.5.4

2 8

[alsa-devel] [PATCH] ASoC: pcm: Add pcm operation for pcm ioctl.
by Liam Girdwood 26 Apr '12

26 Apr '12

Provide an ioctl marshaller for ASoC platform drivers. This will use the default ALSA handler if no platform handler exists. This is also required for DPCM BE PCMs as snd_pcm_info() will call the ioctl as part of stream startup. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-pcm.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5bd8270..4e3be82 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1647,6 +1647,17 @@ out: return ret; } +static int soc_pcm_ioctl(struct snd_pcm_substream *substream, + unsigned int cmd, void *arg) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + + if (platform->driver->ops->ioctl) + return platform->driver->ops->ioctl(substream, cmd, arg); + return snd_pcm_lib_ioctl(substream, cmd, arg); +} + static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_pcm_substream *substream = @@ -2051,6 +2062,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->ops.hw_free = dpcm_fe_dai_hw_free; rtd->ops.close = dpcm_fe_dai_close; rtd->ops.pointer = soc_pcm_pointer; + rtd->ops.ioctl = soc_pcm_ioctl; } else { rtd->ops.open = soc_pcm_open; rtd->ops.hw_params = soc_pcm_hw_params; @@ -2059,6 +2071,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->ops.hw_free = soc_pcm_hw_free; rtd->ops.close = soc_pcm_close; rtd->ops.pointer = soc_pcm_pointer; + rtd->ops.ioctl = soc_pcm_ioctl; } if (platform->driver->ops) { -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm8994: Tune debounce rates for jack detect mode
by Mark Brown 26 Apr '12

26 Apr '12

Use a slightly larger debounce when identifying accessory type and a slightly smaller one when detecting buttons in response to user feedback from large scale testing. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9d50e76..9754904 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -70,8 +70,8 @@ static const struct wm8958_micd_rate micdet_rates[] = { static const struct wm8958_micd_rate jackdet_rates[] = { { 32768, true, 0, 1 }, { 32768, false, 0, 1 }, - { 44100 * 256, true, 7, 10 }, - { 44100 * 256, false, 7, 10 }, + { 44100 * 256, true, 10, 10 }, + { 44100 * 256, false, 7, 8 }, }; static void wm8958_micd_set_rate(struct snd_soc_codec *codec) @@ -3481,6 +3481,8 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) if (present) { dev_dbg(codec->dev, "Jack detected\n"); + wm8958_micd_set_rate(codec); + snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); -- 1.7.10

1 2

[alsa-devel] [PATCH] ASoC: mxs-saif: Use devm_clk_get()
by Fabio Estevam 26 Apr '12

26 Apr '12

Using devm_clk_get() can make the error handling path easier. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 20 +++++++------------- 1 files changed, 7 insertions(+), 13 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 53f4fd8..bfc1f9e 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -650,7 +650,7 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->master_id = saif->id; } - saif->clk = clk_get(&pdev->dev, NULL); + saif->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { ret = PTR_ERR(saif->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", @@ -663,16 +663,14 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->base = devm_request_and_ioremap(&pdev->dev, iores); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENODEV; - goto failed_get_resource; + return -ENODEV; } dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) { - ret = -ENODEV; dev_err(&pdev->dev, "failed to get dma resource: %d\n", ret); - goto failed_get_resource; + return -ENODEV; } saif->dma_param.chan_num = dmares->start; @@ -681,7 +679,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = saif->irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); - goto failed_get_resource; + return saif->irq; } saif->dev = &pdev->dev; @@ -689,15 +687,14 @@ static int mxs_saif_probe(struct platform_device *pdev) "mxs-saif", saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); - goto failed_get_resource; + return ret; } saif->dma_param.chan_irq = platform_get_irq(pdev, 1); if (saif->dma_param.chan_irq < 0) { - ret = saif->dma_param.chan_irq; dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", ret); - goto failed_get_resource; + return saif->dma_param.chan_irq; } platform_set_drvdata(pdev, saif); @@ -705,7 +702,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); - goto failed_get_resource; + return ret; } saif->soc_platform_pdev = platform_device_alloc( @@ -728,8 +725,6 @@ failed_pdev_add: platform_device_put(saif->soc_platform_pdev); failed_pdev_alloc: snd_soc_unregister_dai(&pdev->dev); -failed_get_resource: - clk_put(saif->clk); return ret; } @@ -740,7 +735,6 @@ static int __devexit mxs_saif_remove(struct platform_device *pdev) platform_device_unregister(saif->soc_platform_pdev); snd_soc_unregister_dai(&pdev->dev); - clk_put(saif->clk); return 0; } -- 1.7.1

2 3

[alsa-devel] [PATCH] ASoC: soc-core: Remove unused variable 'min'
by Fabio Estevam 26 Apr '12

26 Apr '12

commit 4183eed2 (ASoC: core: Add signed multi register control) introduced the variable 'min',but it is not used. Remove it to fix the following build warning: sound/soc/soc-core.c: In function 'snd_soc_put_xr_sx': sound/soc/soc-core.c:2990: warning: unused variable 'min' Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/soc-core.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c46ece5..5b0688e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2987,7 +2987,6 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, unsigned int regwmask = (1<<regwshift)-1; unsigned int invert = mc->invert; unsigned long mask = (1UL<<mc->nbits)-1; - long min = mc->min; long max = mc->max; long val = ucontrol->value.integer.value[0]; unsigned int i, regval, regmask; -- 1.7.1

2 1

Re: [alsa-devel] [PATCH 10/10] OMAPDSS: HDMI: Implement DSS driver interface for audio
by Ricardo Neri 26 Apr '12

26 Apr '12

+alsa-devel list On 04/25/2012 01:19 AM, Tomi Valkeinen wrote: > On Tue, 2012-04-24 at 23:48 -0500, Ricardo Neri wrote: >> On 04/23/2012 08:01 AM, Tomi Valkeinen wrote: >>> On Wed, 2012-03-28 at 16:38 -0600, Ricardo Neri wrote: >>>> Implement the DSS device driver audio support interface in the HDMI >>>> panel driver and generic driver. The implementation relies on the >>>> IP-specific functions that are defined at DSS probe time. >>>> >>>> A HW-safe spinlock is used to protect the audio functions. This is because >>> >>> What is a "HW-safe spinlock"? >> Sorry, I meant a spinlock that disables the HW irqs when held:hardirq-safe. >> >>> >>>> the audio functions may be called while holding a lock; in such case, >>>> the panel's driver mutex is not suitable. Functions should be used >>>> to set registers and should not wait for any other event. >>> >>> Are you sure this is the only option? What lock is being held? >> For instance, ALSA calls the start audio function while holding a >> hardirq-safe readlock. Hence, when reaching the HDMI panel start >> function, a lock is held and irqs are disabled. Using a mutex, that >> might sleep, is not correct; nor it is using an hardirq-unsafe spinlock. >> Otherwise, deadlocks and/or inverse lock ordering may arise. This >> situation was signaled by lockdep. >> >> IMHO, as the DSS device driver does not know who is going to use it (at >> least the audio part), it should not assume that no locks are held when >> its functions are called. >> >>> While a spinlock may be ok for now, quite often enabling/disabling things do not >>> happen immediately,and it's much easier to do the wait synchronously. >> I don't understand this comment. To me, holding a lock until the >> enabling function returns is synchronous. Would you please clarify? > > I meant that quite often when enabling things on hardware it takes time > until the HW is actually up and running. Perhaps a regulator needs to be > started, or such. And it's usually simpler to wait for the HW to be up > synchronously in the enable function, instead of some kind of > asynchronous mechanism. And if a function waits synchronously, a mutex > is better than spinlock. > > And in that sense it's often better to define (define meaning, adding a > comment, or just mentally taking note about it) that the functions in > the API may sleep, so that the driver is free to do what is best for the > case, and it's also future-proof in a way that you can easily add > function calls that sleep to the functions in the future. > > But you are also right in your previous comment, it's better to define > functions so that they never sleep, as that gives the callers the > freedom to call the funcs in atomic context. > > Perhaps we can have audio_start() that never sleeps, it just enables a > few bits that start the audio. But how about audio_enable()? Is it > possible that on some OMAP version it would need to enable a regulator, > or set a gpio that's in an external i2c controlled mux chip, or such? I think it might be possible to have such a scenario if, for instance, an external chip is used to support DisplayPort on OMAP4/5. As DisplayPort can support audio-only use-cases, it would have to enable the adapter chip (but HDMI output would have to be enabled to feed the chip, though). > > If so, we need to make sure it's not currently called in an atomic > context, because it would break if the function will sleep in the > future. And with "make sure" I just mean that we check the code and keep > it in mind. Or perhaps adding a comment in the header, that says > "audio_enable may sleep, other audio functions do not sleep" or such. I revisited the ALSA code. Only the audio_start function is atomic. Although ALSA may not be the only user, it makes sense to me to think that they will follow a similar approach in terms of locks. Hence, based on that and on the reasons you describe (audio_enable potentially taking too long to return), Rephrasing what you stated, a mutex may be used for the enable/disable and config operations. Only start/stop would be protected by a spinlock. This should be described in comments in the header file. Does it make sense to you? BR, Ricardo > > Tomi

2 1

[alsa-devel] imx sgtl5000:BUG: scheduling while atomic: speaker-test/1225/0x00000003
by Richard Zhao 26 Apr '12

26 Apr '12

[ 13.901890] eth0: no IPv6 routers present [ 70.295079] BUG: scheduling while atomic: speaker-test/1225/0x00000003 [ 70.304293] Modules linked in: [ 70.310389] [<c0015a68>] (unwind_backtrace+0x0/0xec) from [<c0585810>] (dump_stack+0x20/0x24) [ 70.310421] [<c0585810>] (dump_stack+0x20/0x24) from [<c0585f6c>] (__schedule_bug+0x54/0x60) [ 70.310452] [<c0585f6c>] (__schedule_bug+0x54/0x60) from [<c058db64>] (__schedule+0x84/0x748) [ 70.310476] [<c058db64>] (__schedule+0x84/0x748) from [<c058e310>] (schedule+0x8c/0x90) [ 70.310500] [<c058e310>] (schedule+0x8c/0x90) from [<c058c504>] (schedule_timeout+0x2fc/0x35c) [ 70.310541] [<c058c504>] (schedule_timeout+0x2fc/0x35c) from [<c058d970>] (wait_for_common+0xd8/0x16c) [ 70.310567] [<c058d970>] (wait_for_common+0xd8/0x16c) from [<c058da44>] (wait_for_completion_timeout+0x1c/0x20) [ 70.310604] [<c058da44>] (wait_for_completion_timeout+0x1c/0x20) from [<c03352bc>] (sdma_run_channel+0x70/0x8c) [ 70.310634] [<c03352bc>] (sdma_run_channel+0x70/0x8c) from [<c0335614>] (sdma_load_context+0x188/0x19c) [ 70.310661] [<c0335614>] (sdma_load_context+0x188/0x19c) from [<c0335688>] (sdma_prep_dma_cyclic+0x60/0x1a8) [ 70.310694] [<c0335688>] (sdma_prep_dma_cyclic+0x60/0x1a8) from [<c04760f4>] (snd_dmaengine_pcm_trigger+0xa0/0x134) [ 70.310729] [<c04760f4>] (snd_dmaengine_pcm_trigger+0xa0/0x134) from [<c0483e68>] (soc_pcm_trigger+0x74/0xb0) [ 70.310770] [<c0483e68>] (soc_pcm_trigger+0x74/0xb0) from [<c046e670>] (snd_pcm_do_start+0x34/0x40) [ 70.310797] [<c046e670>] (snd_pcm_do_start+0x34/0x40) from [<c046e288>] (snd_pcm_action_single+0x48/0x74) [ 70.310823] [<c046e288>] (snd_pcm_action_single+0x48/0x74) from [<c046e334>] (snd_pcm_action+0x80/0x8c) [ 70.310850] [<c046e334>] (snd_pcm_action+0x80/0x8c) from [<c04707f0>] (snd_pcm_start+0x24/0x2c) [ 70.310877] [<c04707f0>] (snd_pcm_start+0x24/0x2c) from [<c0474058>] (snd_pcm_lib_write1+0x2a8/0x340) [ 70.310900] [<c0474058>] (snd_pcm_lib_write1+0x2a8/0x340) from [<c04741d0>] (snd_pcm_lib_write+0x70/0x7c) [ 70.311007] [<c04741d0>] (snd_pcm_lib_write+0x70/0x7c) from [<c04703f8>] (snd_pcm_playback_ioctl1+0x114/0x428) [ 70.311038] [<c04703f8>] (snd_pcm_playback_ioctl1+0x114/0x428) from [<c0470744>] (snd_pcm_playback_ioctl+0x38/0x44) [ 70.311077] [<c0470744>] (snd_pcm_playback_ioctl+0x38/0x44) from [<c0107234>] (do_vfs_ioctl+0x4f4/0x568) [ 70.311105] [<c0107234>] (do_vfs_ioctl+0x4f4/0x568) from [<c0107308>] (sys_ioctl+0x60/0x84) [ 70.311138] [<c0107308>] (sys_ioctl+0x60/0x84) from [<c000df40>] (ret_fast_syscall+0x0/0x30) [ 80.301928] ALSA sound/core/pcm_lib.c:1853 playback write error (DMA or IRQ trouble?) kernel: 3.4rc3 with ASoC/for-next and some my patches. Thanks Richard

2 3

[alsa-devel] [snd-usb-audio] frequent "delay: estimated xxx, actual yyy" messages
by Guillaume Clement 25 Apr '12

25 Apr '12

Hello, I am using an USB soundcard (ID : 046d:0a24 : Logitech Speaker Lapdesk N550). During playback, I have frequent messages like these : delay: estimated 795, actual 662 delay: estimated 133, actual 0 delay: estimated 751, actual 662 delay: estimated 89, actual 0 When those messages appear, I am hearing some "cracks" from the soundcard. I've tried to make some research on this, I only found some messages that hint about some interrupt problems. I'm not using any closed source driver and I can reproduce the issue with any computer running Linux. On interesting thing to notice is that it happens only when the CPU is near idle. To workaround the problem, I can just do something like : nice -n 20 dd if=/dev/zero of=/dev/null bs=1k count=500G This would keep my CPU busy and I have no more messages nor cracks. This does not seem to be a recent regression since I've experienced this since I bought the lapdesk in december or so. I am still experiencing this issue with the latest rc (linux-3.4rc3 at the moment) I tried using different .asoundrc setups, tried to install pulseaudio (usually I'm using bare alsa) and still experience the problem. Below is some information that might help locating the problem. I can make some test if needed and even code if I have some information on where the problem might me located. Note: those commands are typed during playback. $ cat /proc/asound/N550/stream0 Logitech, Inc. Logitech Speaker Lapdesk N550 at usb-0000:00:1d.0-2, full speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 URBs = 2 [ 20 20 ] Packet Size = 200 Momentary freq = 44100 Hz (0x2c.199a) Interface 1 Altset 1 Format: S16_LE Channels: 2 Endpoint: 1 OUT (ADAPTIVE) Rates: 44100, 48000 $ cat /proc/asound/N550/pcm0p/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 $ cat /proc/asound/N550/pcm0p/sub0/hw_params access: RW_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 44100 (44100/1) period_size: 5513 buffer_size: 22050 $ cat /proc/asound/N550/pcm0p/sub0/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 $ cat /proc/asound/N550/pcm0p/sub0/status state: RUNNING owner_pid : 1210 trigger_time: 3264.436419606 tstamp : 3302.883104696 delay : 20904 avail : 2646 avail_max : 21026 ----- hw_ptr : 1695160 appl_ptr : 1714564 $ cat /proc/asound/N550/pcm0p/sub0/sw_params tstamp_mode: NONE period_step: 1 avail_min: 5513 start_threshold: 16537 stop_threshold: 22050 silence_threshold: 0 silence_size: 0 boundary: 1445068800

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[alsa-devel] [PATCH] fix bug 5579: invalid long long format specifier
by John Spencer 25 Apr '12

25 Apr '12

the attached patch fixes an invalid long long format specifier issue (%L was used by mistake, which is the long double specifier). this lead to abort() on 64bit system, making some of the alsa-utils unusable. --JS

3 3

[alsa-devel] [PATCH v2] ASoC: omap-pcm: Free dma buffers in case of error.
by Oleg Matcovschi 25 Apr '12

25 Apr '12

Change-Id: I4ba9de0de4681332539246ccc5e11a7a8fb32e79 Signed-off-by: Oleg Matcovschi <oleg.matcovschi(a)ti.com> --- v1: initial revision v2: resending patch including maintainers sound/soc/omap/omap-pcm.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index a2a464f..380a5bd 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -414,6 +414,10 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) } out: + /* free preallocated buffers in case of error */ + if (ret) + omap_pcm_free_dma_buffers(pcm); + return ret; } -- 1.7.4.1

5 4

[alsa-devel] Writing an ASoC codec driver for Beaglebone McASP
by Ulrik Kjems 25 Apr '12

25 Apr '12

I am working on connecting audio from the beaglebone McASP to a codec on an eval board, supporting a general TDM format. I know for sure the setup is viable because I have the system running in a bare-metal configuration using Starterware. I want to make it work under Linux ALSA/ASoC. I am willing to share the source code if anyone is interested in working on this as well. To do this I need to configure/write a driver for ALSA/ASoC, and my goal is to be able to perform multi-channel sound capture and playback using ALSA. The current kernel (3.2.14) davinci platform driver (davinci-evm.c) apparantly supports TI's AM3358 EVM board, which uses the same ARM3358 Sitara processor as the Beaglebone. The EVM board uses a similar (but not identical) McASP config (and a TLV320AIC3x codec, I believe). When booting the kernel on the Beaglebone the ALSA driver fails to register with the McASP. I can see in soc-core.c around line 825 that four client devices must be found: codec, codec_dai, platform, and platform_dai. By modifying sound/soc/davinci/davinci-evm.c to reflect the fact that for beaglebone McASP 0 is being used (as opposed to the EVM board where mcasp 1 is in use), I can make the platform register: static struct snd_soc_dai_link am335x_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", // .cpu_dai_name = "davinci-mcasp.1", .cpu_dai_name = "davinci-mcasp.0", // <<<<<<<Changed .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-001b", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, }; Now, all I need is for the codec to register. If I look at /sys/kernel/debug/asoc/ I can see the davinci platform now (the above change enables this) ~# cat /sys/kernel/debug/asoc/platforms davinci-pcm-audio ~# cat /sys/kernel/debug/asoc/dais davinci-mcasp.0 ~# cat /sys/kernel/debug/asoc/codecs # No codecs! ~# Of course, I dont have the codec tlv320aic3x-hifi on my system,- but for now I would be happy configuring my codec manually (as I said its on an eval board with a config software tool I can use in place of I2C) so that I could get sound through the system. So my question is, how do I go about convincing ALSA that its OK not to have the codec? Is there a generic codec I could use as a starting point (e.g. setting codec_dai_name in the above struct to something else)? I dont understand the kernel code well enough to see why and how the kernel can detect the absence of the TLV codec above - I dont seem to be getting any calls to any of the functions inside sound/soc/codecs/tlv320aic3x.c except aic3x_modinit(void) - how on earth does the system know the codec is not there ?

1 0

[alsa-devel] Logical device name for 4-channel microphones?
by Tanu Kaskinen 25 Apr '12

25 Apr '12

Hi, What is the best source of information about device names ("front", "surround40" etc)? Is it the DeviceNames wiki page[1]? That page was slightly out of date, so I updated and expanded it (review would be welcome). I'm looking for information specifically about capture device names. The wiki page didn't help me there, as it only listed playback device names. My problem is that I'd like to make Pulseaudio handle 4-channel webcam microphones[2], but I'm not aware of any input device names for 4-channel devices. Are there such device names? If not, what should be done? Pulseaudio could just use "hw", but I'd prefer there to be a standard device name for 4-channel mics. AFAIK, at playback side, some surround implementations are using multiple devices on one card, and using "hw" wouldn't work in those cases, while e.g. "surround40" can use the multiple devices behind the scenes. I would guess that the same could be the case with some multichannel microphones, and therefore a logical name like "fourchannelmic" would be better than using "hw". [1] http://www.alsa-project.org/main/index.php/DeviceNames [2] https://bugs.freedesktop.org/show_bug.cgi?id=45813 -- Tanu

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[alsa-devel] [PATCH 1/3] ASoC: SSM2602: Add sysclk based rate constraints
by Lars-Peter Clausen 25 Apr '12

25 Apr '12

Not all advertised rates are available for all sysclk frequencies. Add additional sysclk based rate constraints. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm2602.c | 38 ++++++++++++++++++++++++++++++++++---- 1 file changed, 34 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 4c94fd2..dcedd74 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -53,6 +53,7 @@ enum ssm2602_type { /* codec private data */ struct ssm2602_priv { unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; enum snd_soc_control_type control_type; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; @@ -195,6 +196,24 @@ static const struct snd_soc_dapm_route ssm2604_routes[] = { {"ADC", NULL, "Line Input"}, }; +static const unsigned int ssm2602_rates_12288000[] = { + 8000, 32000, 48000, 96000, +}; + +static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { + .list = ssm2602_rates_12288000, + .count = ARRAY_SIZE(ssm2602_rates_12288000), +}; + +static const unsigned int ssm2602_rates_11289600[] = { + 8000, 44100, 88200, +}; + +static struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { + .list = ssm2602_rates_11289600, + .count = ARRAY_SIZE(ssm2602_rates_11289600), +}; + struct ssm2602_coeff { u32 mclk; u32 rate; @@ -320,6 +339,12 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, } else ssm2602->master_substream = substream; + if (ssm2602->sysclk_constraints) { + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + ssm2602->sysclk_constraints); + } + return 0; } @@ -361,16 +386,21 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; switch (freq) { - case 11289600: - case 12000000: case 12288000: - case 16934400: case 18432000: - ssm2602->sysclk = freq; + ssm2602->sysclk_constraints = &ssm2602_constraints_12288000; + break; + case 11289600: + case 16934400: + ssm2602->sysclk_constraints = &ssm2602_constraints_11289600; + break; + case 12000000: + ssm2602->sysclk_constraints = NULL; break; default: return -EINVAL; } + ssm2602->sysclk = freq; } else { unsigned int mask; -- 1.7.9.5

2 3

[alsa-devel] [PATCH v2] ASoC: bf5xx-ssm2602: Setup sysclock in init callback
by Lars-Peter Clausen 25 Apr '12

25 Apr '12

The sysclock is fixed, so just set it up once in the init callback instead of setting it repeatably in the hw_params callback. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Since v1: * Fix duplicate use of bf5xx_ssm2602_init function name. Sorry, I amended the wrong commit before sending out v1... --- sound/soc/blackfin/bf5xx-ssm2602.c | 37 ++++-------------------------------- 1 file changed, 4 insertions(+), 33 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index b39ad35..7dbeef1 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -44,16 +44,8 @@ static struct snd_soc_card bf5xx_ssm2602; -static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int bf5xx_ssm2602_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int clk = 0; - int ret = 0; - - pr_debug("%s rate %d format %x\n", __func__, params_rate(params), - params_format(params)); /* * If you are using a crystal source which frequency is not 12MHz * then modify the below case statement with frequency of the crystal. @@ -61,31 +53,10 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, * If you are using the SPORT to generate clocking then this is * where to do it. */ - - switch (params_rate(params)) { - case 8000: - case 16000: - case 48000: - case 96000: - case 11025: - case 22050: - case 44100: - clk = 12000000; - break; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk, + return snd_soc_dai_set_sysclk(rtd->codec_dai, SSM2602_SYSCLK, 12000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; } -static struct snd_soc_ops bf5xx_ssm2602_ops = { - .hw_params = bf5xx_ssm2602_hw_params, -}; - /* CODEC is master for BCLK and LRC in this configuration. */ #define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ SND_SOC_DAIFMT_CBM_CFM) @@ -98,7 +69,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .codec_dai_name = "ssm2602-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", - .ops = &bf5xx_ssm2602_ops, + .init = bf5xx_ssm2602_dai_init, .dai_fmt = BF5XX_SSM2602_DAIFMT, }, { @@ -108,7 +79,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .codec_dai_name = "ssm2602-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", - .ops = &bf5xx_ssm2602_ops, + .init = bf5xx_ssm2602_dai_init, .dai_fmt = BF5XX_SSM2602_DAIFMT, }, }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: bf5xx-ssm2602: Set DAI format
by Lars-Peter Clausen 25 Apr '12

25 Apr '12

Commit 980b0bc69 ("ASoC: blackfin: Use dai_fmt") converted the blackfin ASoC machine drivers to use the dai_links dai_fmt field to setup their DAI format. For the bf5xx-ssm2602 the commit removed the manual call to snd_soc_dai_set_fmt, but missed to set the dai_links dai_fmt field. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- The commit that introduced the problem is only in 3.4-rc --- sound/soc/blackfin/bf5xx-ssm2602.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index df3ac73..b39ad35 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -99,6 +99,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, { .name = "ssm2602", @@ -108,6 +109,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, }; -- 1.7.9.5

2 2

[alsa-devel] snd_usb_hardware definition
by Amit Nagal 25 Apr '12

25 Apr '12

Hi , In H/W Descriptor snd_usb_hardware (sound/usb/pcm.c) , the following values are used : static struct snd_pcm_hardware snd_usb_hardware = { .... .buffer_bytes_max = 1024 * 1024, .period_bytes_min = 64, .period_bytes_max = 512 * 1024, .periods_min = 2, .periods_max = 1024, }; can i be guided what criterion is used to set the values of above fields ( periods_min, periods_max , period_bytes_min , period_bytes_max , buffer_bytes_max) as defined above for a snd usb h/w . Regards Amit Nagal

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Make sure we disable FLL bypass when stopping the FLL
by Mark Brown 25 Apr '12

25 Apr '12

From: Kyung-Kwee Ryu <Kyung-Kwee.Ryu(a)wolfsonmicro.com> If FLL bypass is left enabled when we disable the CODEC then the output clock will be left running which consumes a small amount of additional current. Only enable bypass when there is an output. Signed-off-by: Kyung-Kwee Ryu <Kyung-Kwee.Ryu(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4cebac4..746d3db 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1982,7 +1982,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, WM8994_FLL1_ENA, 0); if (wm8994->fll_byp && src == WM8994_FLL_SRC_BCLK && - freq_in == freq_out) { + freq_in == freq_out && freq_out) { dev_dbg(codec->dev, "Bypassing FLL%d\n", id + 1); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, WM8958_FLL1_BYP, WM8958_FLL1_BYP); -- 1.7.10

1 0

Re: [alsa-devel] [PATCH] ASoC: sgtl5000: Enable VAG when DAC/ADC up
by Mark Brown 25 Apr '12

25 Apr '12

On Thu, Mar 29, 2012 at 03:17:00PM +0800, Zeng Zhaoming wrote: > So the code should be: > + {"MIC_IN", NULL, "Mic Bias}, /* mic bias --> mic_in */ > {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ > -+ {"Mic Bias", NULL, "MIC_IN"}, /* mic_in --> mic bias */ > -+ {"Capture Mux", "MIC_IN", "Mic Bias"}, /* mic bias --> adc_mux */ > and platform code > + {"Mic Bias", NULL, "MIC"}, > - {"MIC_IN", NULL, "MIC"}, No, the bias should be a supply for the microphone - ensuring that this happens is the whole point of the patch I sent the other day.

3 5

[alsa-devel] [PATCH] ALSA: snd-usb: fix some typos in endpoint.c documentation
by Daniel Mack 24 Apr '12

24 Apr '12

Also be more specific about some details while at it. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/endpoint.c | 58 ++++++++++++++++++++++++++++--------------------- 1 files changed, 33 insertions(+), 25 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 12e5a95..286a8d2 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -40,25 +40,26 @@ * USB endpoint and its streaming. * * There are functions to activate and deactivate the streaming URBs and - * optinal callbacks to let the pcm logic handle the actual content of the + * optional callbacks to let the pcm logic handle the actual content of the * packets for playback and record. Thus, the bus streaming and the audio * handlers are fully decoupled. * - * There are two different types of endpoints in for audio applications. + * There are two different types of endpoints in audio applications. * * SND_USB_ENDPOINT_TYPE_DATA handles full audio data payload for both * inbound and outbound traffic. * - * SND_USB_ENDPOINT_TYPE_SYNC are for inbound traffic only and expect the - * payload to carry Q16.16 formatted sync information (3 or 4 bytes). + * SND_USB_ENDPOINT_TYPE_SYNC endpoints are for inbound traffic only and + * expect the payload to carry Q10.14 / Q16.16 formatted sync information + * (3 or 4 bytes). * - * Each endpoint has to be configured (by calling - * snd_usb_endpoint_set_params()) before it can be used. + * Each endpoint has to be configured prior to being used by calling + * snd_usb_endpoint_set_params(). * * The model incorporates a reference counting, so that multiple users * can call snd_usb_endpoint_start() and snd_usb_endpoint_stop(), and * only the first user will effectively start the URBs, and only the last - * one will tear them down again. + * one to stop it will tear the URBs down again. */ /* @@ -120,7 +121,7 @@ static const char *usb_error_string(int err) /** * snd_usb_endpoint_implicit_feedback_sink: Report endpoint usage type * - * @ep: The endpoint + * @ep: The snd_usb_endpoint * * Determine whether an endpoint is driven by an implicit feedback * data endpoint source. @@ -278,17 +279,18 @@ static inline void prepare_inbound_urb(struct snd_usb_endpoint *ep, } /* - * Send output urbs that have been prepared previously. Urbs are dequeued + * Send output urbs that have been prepared previously. URBs are dequeued * from ep->ready_playback_urbs and in case there there aren't any available * or there are no packets that have been prepared, this function does * nothing. * - * The reason why the functionality of sending and preparing urbs is separated - * is that host controllers don't guarantee an ordering in returing inbound - * and outbound packets to their submitters. + * The reason why the functionality of sending and preparing URBs is separated + * is that host controllers don't guarantee the order in which they return + * inbound and outbound packets to their submitters. * * This function is only used for implicit feedback endpoints. For endpoints - * driven by sync endpoints, urbs are submitted from their completion handler. + * driven by dedicated sync endpoints, URBs are immediately re-submitted + * from their completion handler. */ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) { @@ -391,7 +393,7 @@ exit_clear: } /** - * snd_usb_add_endpoint: Add an endpoint to an audio chip + * snd_usb_add_endpoint: Add an endpoint to an USB audio chip * * @chip: The chip * @alts: The USB host interface @@ -745,11 +747,14 @@ out_of_memory: } /** - * snd_usb_endpoint_set_params: configure an snd_endpoint + * snd_usb_endpoint_set_params: configure an snd_usb_endpoint * - * @ep: the endpoint to configure + * @ep: the snd_usb_endpoint to configure + * @hw_params: the hardware parameters + * @fmt: the USB audio format information + * @sync_ep: the sync endpoint to use, if any * - * Determine the number of of URBs to be used on this endpoint. + * Determine the number of URBs to be used on this endpoint. * An endpoint must be configured before it can be started. * An endpoint that is already running can not be reconfigured. */ @@ -807,7 +812,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, * @ep: the endpoint to start * * A call to this function will increment the use count of the endpoint. - * In case this not already running, the URBs for this endpoint will be + * In case it is not already running, the URBs for this endpoint will be * submitted. Otherwise, this function does nothing. * * Must be balanced to calls of snd_usb_endpoint_stop(). @@ -840,7 +845,8 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) /* * If this endpoint has a data endpoint as implicit feedback source, * don't start the urbs here. Instead, mark them all as available, - * wait for the record urbs to arrive and queue from that context. + * wait for the record urbs to return and queue the playback urbs + * from that context. */ set_bit(EP_FLAG_RUNNING, &ep->flags); @@ -892,7 +898,7 @@ __error: * * A call to this function will decrement the use count of the endpoint. * In case the last user has requested the endpoint stop, the URBs will - * actually deactivated. + * actually be deactivated. * * Must be balanced to calls of snd_usb_endpoint_start(). */ @@ -994,7 +1000,8 @@ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) return -EBUSY; } -/** snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint +/** + * snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint * * @ep: the list header of the endpoint to free * @@ -1032,8 +1039,9 @@ void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, /* * In case the endpoint is operating in implicit feedback mode, prepare - * and a new outbound URB that has the same layout as the received - * packet and add it to the list of pending urbs. + * a new outbound URB that has the same layout as the received packet + * and add it to the list of pending urbs. queue_pending_output_urbs() + * will take care of them later. */ if (snd_usb_endpoint_implict_feedback_sink(ep) && ep->use_count != 0) { @@ -1063,8 +1071,8 @@ void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, /* * Iterate through the inbound packet and prepare the lengths * for the output packet. The OUT packet we are about to send - * will have the same amount of payload than the IN packet we - * just received. + * will have the same amount of payload bytes than the IN + * packet we just received. */ out_packet->packets = in_ctx->packets; -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/1] ASoC: Core: check of_property_count_strings failure
by Richard Zhao 24 Apr '12

24 Apr '12

Signed-off-by: Richard Zhao <richard.zhao(a)freescale.com> --- sound/soc/soc-core.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 20e55531..ce4d462 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3683,10 +3683,10 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, int i, ret; num_routes = of_property_count_strings(np, propname); - if (num_routes & 1) { + if (num_routes < 0 || num_routes & 1) { dev_err(card->dev, - "Property '%s's length is not even\n", - propname); + "Property '%s' does not exist or its length is not even\n", + propname); return -EINVAL; } num_routes /= 2; -- 1.7.5.4

3 2

[alsa-devel] Playback underrun recovery
by Paul D. DeRocco 24 Apr '12

24 Apr '12

I've got a synth program outputting directly to the main hw PCM device, with no intervening plugins. It uses about 75-80% of the time on one core, so it normally doesn't underrun. Once in a while, other burdens on the system trigger an underrun, and my program never recovers. My buffer contains two periods of 256 stereo frames (with 32-bit samples), which at 96KHz comes out to a little over 5.3ms total. When it fails, its CPU usage goes up to 100% on that core, and it spits out bursts of 5.3ms of audio (two full periods, not one), with about 4ms of silence between them. My top level code consists of a loop that generates one period at a time: int buf[512]; while (true) { generate_samples(buf, 256); // generate 256 stereo frames if (snd_pcm_writei(h, buf, 256) < 0) snd_pcm_prepare(h); } The funny thing is that if I break to the debugger and then restart, I get exactly one underrun as you would expect, and then it recovers. What could cause it to get into a state where it fails every other time around the loop, and then sucks up milliseconds of CPU time before it gets going again, thus guaranteeing another underrun? -- Ciao, Paul D. DeRocco Paul mailto:pderocco@ix.netcom.com

2 1

[alsa-devel] ASoC: sh: fsi: Build proble in 3.4-rc4
by Rafael J. Wysocki 24 Apr '12

24 Apr '12

Hi all, There is a build problem in sound/soc/sh/fsi.c which seems to be fixed by the following patch. I'm unsure if that is the right fix, though. Thanks, Rafael --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) Index: linux/sound/soc/sh/fsi.c =================================================================== --- linux.orig/sound/soc/sh/fsi.c +++ linux/sound/soc/sh/fsi.c @@ -1003,7 +1003,7 @@ static void fsi_dma_do_tasklet(unsigned desc = chan->device->device_prep_slave_sg(chan, &sg, 1, dir, DMA_PREP_INTERRUPT | - DMA_CTRL_ACK); + DMA_CTRL_ACK, NULL); if (!desc) { dev_err(dai->dev, "device_prep_slave_sg() fail\n"); return;

3 3

[PATCH] fix bug 5579: invalid long long format specifier
by John Spencer 24 Apr '12

24 Apr '12

Per POSIX: L Specifies that a following a, A, e, E, f, F, g, or G conversion specifier applies to a long double argument. L is only intended to be used with long doubles, not long long ints. the proper way is to use "ll" instead. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- src/conf.c | 6 +++--- test/midiloop.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/src/conf.c b/src/conf.c index 5b1b5a6..32446a2 100644 --- a/src/conf.c +++ b/src/conf.c @@ -496,7 +496,7 @@ static int safe_strtoll(const char *str, long long *val) if (!*str) return -EINVAL; errno = 0; - if (sscanf(str, "%Li%n", &v, &endidx) < 1) + if (sscanf(str, "%lli%n", &v, &endidx) < 1) return -EINVAL; if (str[endidx]) return -EINVAL; @@ -1378,7 +1378,7 @@ static int _snd_config_save_node_value(snd_config_t *n, snd_output_t *out, snd_output_printf(out, "%ld", n->u.integer); break; case SND_CONFIG_TYPE_INTEGER64: - snd_output_printf(out, "%Ld", n->u.integer64); + snd_output_printf(out, "%lld", n->u.integer64); break; case SND_CONFIG_TYPE_REAL: snd_output_printf(out, "%-16g", n->u.real); @@ -2630,7 +2630,7 @@ int snd_config_get_ascii(const snd_config_t *config, char **ascii) { char res[32]; int err; - err = snprintf(res, sizeof(res), "%Li", config->u.integer64); + err = snprintf(res, sizeof(res), "%lli", config->u.integer64); if (err < 0 || err == sizeof(res)) { assert(0); return -ENOMEM; diff --git a/test/midiloop.c b/test/midiloop.c index ee2e563..d6548b5 100644 --- a/test/midiloop.c +++ b/test/midiloop.c @@ -175,7 +175,7 @@ int main(int argc, char** argv) printf("output.status.xruns = %zi\n", snd_rawmidi_status_get_xruns(ostat)); diff = timediff(end, start); - printf("Time diff: %Liusec (%Li bytes/sec)\n", diff, ((long long)opos * 1000000) / diff); + printf("Time diff: %lliusec (%lli bytes/sec)\n", diff, ((long long)opos * 1000000) / diff); if (verbose) { fprintf(stderr,"Closing\n"); -- 1.7.3.4 --------------040707000502000402000200--

1 0

[alsa-devel] [PATCH 4/8] arm: ux500: Add support for MSP I2S-devices
by Ola Lilja 24 Apr '12

24 Apr '12

Create devices for the MSP-blocks (MSP0, MSP1, MSP2 and MSP3) and associate it with the correct clocks in the clock-framework. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- arch/arm/mach-ux500/Makefile | 3 +- arch/arm/mach-ux500/board-mop500-msp.c | 250 ++++++++++++++++++++++++++++++++ arch/arm/mach-ux500/board-mop500-msp.h | 14 ++ arch/arm/mach-ux500/board-mop500.c | 4 + arch/arm/mach-ux500/clock.c | 8 +- arch/arm/mach-ux500/devices-db8500.h | 10 -- arch/arm/mach-ux500/include/mach/msp.h | 29 ++++ 7 files changed, 304 insertions(+), 14 deletions(-) create mode 100644 arch/arm/mach-ux500/board-mop500-msp.c create mode 100644 arch/arm/mach-ux500/board-mop500-msp.h create mode 100644 arch/arm/mach-ux500/include/mach/msp.h diff --git a/arch/arm/mach-ux500/Makefile b/arch/arm/mach-ux500/Makefile index 465b9ec..11729bc 100644 --- a/arch/arm/mach-ux500/Makefile +++ b/arch/arm/mach-ux500/Makefile @@ -11,7 +11,8 @@ obj-$(CONFIG_MACH_MOP500) += board-mop500.o board-mop500-sdi.o \ board-mop500-regulators.o \ board-mop500-uib.o board-mop500-stuib.o \ board-mop500-u8500uib.o \ - board-mop500-pins.o + board-mop500-pins.o \ + board-mop500-msp.o obj-$(CONFIG_MACH_U5500) += board-u5500.o board-u5500-sdi.o obj-$(CONFIG_SMP) += platsmp.o headsmp.o obj-$(CONFIG_HOTPLUG_CPU) += hotplug.o diff --git a/arch/arm/mach-ux500/board-mop500-msp.c b/arch/arm/mach-ux500/board-mop500-msp.c new file mode 100644 index 0000000..c8f6300 --- /dev/null +++ b/arch/arm/mach-ux500/board-mop500-msp.c @@ -0,0 +1,250 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * License terms: GNU General Public License (GPL), version 2 + */ + +#include <linux/platform_device.h> +#include <linux/init.h> +#include <linux/gpio.h> +#include <plat/gpio-nomadik.h> + +#include <plat/pincfg.h> +#include <plat/ste_dma40.h> + +#include <mach/devices.h> +#include <ste-dma40-db8500.h> +#include <mach/hardware.h> +#include <mach/irqs.h> +#include <mach/msp.h> + +#include "board-mop500.h" +#include "devices-db8500.h" +#include "pins-db8500.h" + +/* MSP1/3 Tx/Rx usage protection */ +static DEFINE_SPINLOCK(msp_rxtx_lock); + +/* Reference Count */ +static int msp_rxtx_ref; + +static pin_cfg_t mop500_msp1_pins_init[] = { + GPIO33_MSP1_TXD | PIN_OUTPUT_LOW | PIN_SLPM_WAKEUP_DISABLE, + GPIO34_MSP1_TFS | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_DISABLE, + GPIO35_MSP1_TCK | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_DISABLE, + GPIO36_MSP1_RXD | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_DISABLE, +}; + +static pin_cfg_t mop500_msp1_pins_exit[] = { + GPIO33_MSP1_TXD | PIN_OUTPUT_LOW | PIN_SLPM_WAKEUP_ENABLE, + GPIO34_MSP1_TFS | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_ENABLE, + GPIO35_MSP1_TCK | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_ENABLE, + GPIO36_MSP1_RXD | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_ENABLE, +}; + +int msp13_i2s_init(void) +{ + int retval = 0; + unsigned long flags; + + spin_lock_irqsave(&msp_rxtx_lock, flags); + if (msp_rxtx_ref == 0) + retval = nmk_config_pins( + ARRAY_AND_SIZE(mop500_msp1_pins_init)); + if (!retval) + msp_rxtx_ref++; + spin_unlock_irqrestore(&msp_rxtx_lock, flags); + + return retval; +} + +int msp13_i2s_exit(void) +{ + int retval = 0; + unsigned long flags; + + spin_lock_irqsave(&msp_rxtx_lock, flags); + WARN_ON(!msp_rxtx_ref); + msp_rxtx_ref--; + if (msp_rxtx_ref == 0) + retval = nmk_config_pins_sleep( + ARRAY_AND_SIZE(mop500_msp1_pins_exit)); + spin_unlock_irqrestore(&msp_rxtx_lock, flags); + + return retval; +} + +static struct stedma40_chan_cfg msp0_dma_rx = { + .high_priority = true, + .dir = STEDMA40_PERIPH_TO_MEM, + + .src_dev_type = DB8500_DMA_DEV31_MSP0_RX_SLIM0_CH0_RX, + .dst_dev_type = STEDMA40_DEV_DST_MEMORY, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct stedma40_chan_cfg msp0_dma_tx = { + .high_priority = true, + .dir = STEDMA40_MEM_TO_PERIPH, + + .src_dev_type = STEDMA40_DEV_DST_MEMORY, + .dst_dev_type = DB8500_DMA_DEV31_MSP0_TX_SLIM0_CH0_TX, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct msp_i2s_platform_data msp0_platform_data = { + .id = MSP_I2S_0, + .msp_i2s_dma_rx = &msp0_dma_rx, + .msp_i2s_dma_tx = &msp0_dma_tx, +}; + +static struct stedma40_chan_cfg msp1_dma_rx = { + .high_priority = true, + .dir = STEDMA40_PERIPH_TO_MEM, + + .src_dev_type = DB8500_DMA_DEV30_MSP3_RX, + .dst_dev_type = STEDMA40_DEV_DST_MEMORY, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct stedma40_chan_cfg msp1_dma_tx = { + .high_priority = true, + .dir = STEDMA40_MEM_TO_PERIPH, + + .src_dev_type = STEDMA40_DEV_DST_MEMORY, + .dst_dev_type = DB8500_DMA_DEV30_MSP1_TX, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct msp_i2s_platform_data msp1_platform_data = { + .id = MSP_I2S_1, + .msp_i2s_dma_rx = NULL, + .msp_i2s_dma_tx = &msp1_dma_tx, + .msp_i2s_init = msp13_i2s_init, + .msp_i2s_exit = msp13_i2s_exit, +}; + +static struct stedma40_chan_cfg msp2_dma_rx = { + .high_priority = true, + .dir = STEDMA40_PERIPH_TO_MEM, + + .src_dev_type = DB8500_DMA_DEV14_MSP2_RX, + .dst_dev_type = STEDMA40_DEV_DST_MEMORY, + + /* MSP2 DMA doesn't work with PSIZE == 4 on DB8500v2 */ + .src_info.psize = STEDMA40_PSIZE_LOG_1, + .dst_info.psize = STEDMA40_PSIZE_LOG_1, + + /* data_width is set during configuration */ +}; + +static struct stedma40_chan_cfg msp2_dma_tx = { + .high_priority = true, + .dir = STEDMA40_MEM_TO_PERIPH, + + .src_dev_type = STEDMA40_DEV_DST_MEMORY, + .dst_dev_type = DB8500_DMA_DEV14_MSP2_TX, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + .use_fixed_channel = true, + .phy_channel = 1, + + /* data_width is set during configuration */ +}; + +static int db8500_add_msp_i2s(struct device *parent, int id, + resource_size_t base, int irq, + struct msp_i2s_platform_data *pdata) +{ + struct platform_device *pdev; + struct resource res[] = { + DEFINE_RES_MEM(base, SZ_4K), + DEFINE_RES_IRQ(irq), + }; + + pr_info("Register platform-device 'ux500-msp-i2s', id %d, irq %d\n", + id, irq); + pdev = platform_device_register_resndata(parent, "ux500-msp-i2s", id, + res, ARRAY_SIZE(res), + pdata, sizeof(*pdata)); + if (!pdev) { + pr_err("Failed to register platform-device 'ux500-msp-i2s.%d'!\n", + id); + return -EIO; + } + + return 0; +} + +/* Platform device for ASoC U8500 machine */ +static struct platform_device snd_soc_u8500 = { + .name = "snd-soc-u8500", + .id = 0, + .dev = { + .platform_data = NULL, + }, +}; + +/* Platform device for Ux500-PCM */ +static struct platform_device ux500_pcm = { + .name = "ux500-pcm", + .id = 0, + .dev = { + .platform_data = NULL, + }, +}; + +static struct msp_i2s_platform_data msp2_platform_data = { + .id = MSP_I2S_2, + .msp_i2s_dma_rx = &msp2_dma_rx, + .msp_i2s_dma_tx = &msp2_dma_tx, +}; + +static struct msp_i2s_platform_data msp3_platform_data = { + .id = MSP_I2S_3, + .msp_i2s_dma_rx = &msp1_dma_rx, + .msp_i2s_dma_tx = NULL, + .msp_i2s_init = msp13_i2s_init, + .msp_i2s_exit = msp13_i2s_exit, +}; + +int mop500_msp_init(struct device *parent) +{ + int ret; + + pr_info("%s: Register platform-device 'snd-soc-u8500'.\n", __func__); + platform_device_register(&snd_soc_u8500); + + pr_info("Initialize MSP I2S-devices.\n"); + ret = db8500_add_msp_i2s(parent, 0, U8500_MSP0_BASE, IRQ_DB8500_MSP0, + &msp0_platform_data); + ret |= db8500_add_msp_i2s(parent, 1, U8500_MSP1_BASE, IRQ_DB8500_MSP1, + &msp1_platform_data); + ret |= db8500_add_msp_i2s(parent, 2, U8500_MSP2_BASE, IRQ_DB8500_MSP2, + &msp2_platform_data); + ret |= db8500_add_msp_i2s(parent, 3, U8500_MSP3_BASE, IRQ_DB8500_MSP1, + &msp3_platform_data); + + pr_info("%s: Register platform-device 'ux500-pcm'\n", __func__); + platform_device_register(&ux500_pcm); + + return ret; +} diff --git a/arch/arm/mach-ux500/board-mop500-msp.h b/arch/arm/mach-ux500/board-mop500-msp.h new file mode 100644 index 0000000..6fcfb5e --- /dev/null +++ b/arch/arm/mach-ux500/board-mop500-msp.h @@ -0,0 +1,14 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * for ST-Ericsson. + * + * License terms: + * + * 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. + */ + +void mop500_msp_init(struct device *parent); diff --git a/arch/arm/mach-ux500/board-mop500.c b/arch/arm/mach-ux500/board-mop500.c index 9dfaab3..4cbed08 100644 --- a/arch/arm/mach-ux500/board-mop500.c +++ b/arch/arm/mach-ux500/board-mop500.c @@ -53,6 +53,7 @@ #include "devices-db8500.h" #include "board-mop500.h" #include "board-mop500-regulators.h" +#include "board-mop500-msp.h" static struct gpio_led snowball_led_array[] = { { @@ -631,6 +632,7 @@ static void __init mop500_init_machine(void) mop500_i2c_init(parent); mop500_sdi_init(parent); mop500_spi_init(parent); + mop500_msp_init(parent); mop500_uart_init(parent); i2c0_devs = ARRAY_SIZE(mop500_i2c0_devices); @@ -662,6 +664,7 @@ static void __init snowball_init_machine(void) mop500_i2c_init(parent); snowball_sdi_init(parent); mop500_spi_init(parent); + mop500_msp_init(parent); mop500_uart_init(parent); i2c0_devs = ARRAY_SIZE(mop500_i2c0_devices); @@ -699,6 +702,7 @@ static void __init hrefv60_init_machine(void) mop500_i2c_init(parent); hrefv60_sdi_init(parent); mop500_spi_init(parent); + mop500_msp_init(parent); mop500_uart_init(parent); i2c0_devs = ARRAY_SIZE(mop500_i2c0_devices); diff --git a/arch/arm/mach-ux500/clock.c b/arch/arm/mach-ux500/clock.c index cc87f77..9c2ffcdf 100644 --- a/arch/arm/mach-ux500/clock.c +++ b/arch/arm/mach-ux500/clock.c @@ -336,6 +336,7 @@ static DEFINE_PRCMU_CLK(uiccclk, 0x4, 1, UICCCLK); /* v1 */ */ /* Peripheral Cluster #1 */ +static DEFINE_PRCC_CLK(1, msp3, 11, 10, &clk_msp1clk); static DEFINE_PRCC_CLK(1, i2c4, 10, 9, &clk_i2cclk); static DEFINE_PRCC_CLK(1, gpio0, 9, -1, NULL); static DEFINE_PRCC_CLK(1, slimbus0, 8, 8, &clk_slimclk); @@ -405,7 +406,7 @@ static struct clk_lookup u8500_clks[] = { CLK(slimbus0, "slimbus0", NULL), CLK(i2c2, "nmk-i2c.2", NULL), CLK(sdi0, "sdi0", NULL), - CLK(msp0, "msp0", NULL), + CLK(msp0, "ux500-msp-i2s.0", NULL), CLK(i2c1, "nmk-i2c.1", NULL), CLK(uart1, "uart1", NULL), CLK(uart0, "uart0", NULL), @@ -455,7 +456,8 @@ static struct clk_lookup u8500_clks[] = { /* Peripheral Cluster #1 */ CLK(i2c4, "nmk-i2c.4", NULL), CLK(spi3, "spi3", NULL), - CLK(msp1, "msp1", NULL), + CLK(msp1, "ux500-msp-i2s.1", NULL), + CLK(msp3, "ux500-msp-i2s.3", NULL), /* Peripheral Cluster #2 */ CLK(gpio1, "gpio.6", NULL), @@ -465,7 +467,7 @@ static struct clk_lookup u8500_clks[] = { CLK(spi0, "spi0", NULL), CLK(sdi3, "sdi3", NULL), CLK(sdi1, "sdi1", NULL), - CLK(msp2, "msp2", NULL), + CLK(msp2, "ux500-msp-i2s.2", NULL), CLK(sdi4, "sdi4", NULL), CLK(pwl, "pwl", NULL), CLK(spi1, "spi1", NULL), diff --git a/arch/arm/mach-ux500/devices-db8500.h b/arch/arm/mach-ux500/devices-db8500.h index 9fd93e9..e22b786 100644 --- a/arch/arm/mach-ux500/devices-db8500.h +++ b/arch/arm/mach-ux500/devices-db8500.h @@ -34,7 +34,6 @@ db8500_add_ssp(struct device *parent, const char *name, resource_size_t base, return dbx500_add_amba_device(parent, name, base, irq, pdata, 0); } - #define db8500_add_i2c0(parent, pdata) \ dbx500_add_i2c(parent, 0, U8500_I2C0_BASE, IRQ_DB8500_I2C0, pdata) #define db8500_add_i2c1(parent, pdata) \ @@ -46,15 +45,6 @@ db8500_add_ssp(struct device *parent, const char *name, resource_size_t base, #define db8500_add_i2c4(parent, pdata) \ dbx500_add_i2c(parent, 4, U8500_I2C4_BASE, IRQ_DB8500_I2C4, pdata) -#define db8500_add_msp0_i2s(parent, pdata) \ - dbx500_add_msp_i2s(parent, 0, U8500_MSP0_BASE, IRQ_DB8500_MSP0, pdata) -#define db8500_add_msp1_i2s(parent, pdata) \ - dbx500_add_msp_i2s(parent, 1, U8500_MSP1_BASE, IRQ_DB8500_MSP1, pdata) -#define db8500_add_msp2_i2s(parent, pdata) \ - dbx500_add_msp_i2s(parent, 2, U8500_MSP2_BASE, IRQ_DB8500_MSP2, pdata) -#define db8500_add_msp3_i2s(parent, pdata) \ - dbx500_add_msp_i2s(parent, 3, U8500_MSP3_BASE, IRQ_DB8500_MSP1, pdata) - #define db8500_add_msp0_spi(parent, pdata) \ dbx500_add_msp_spi(parent, "msp0", U8500_MSP0_BASE, \ IRQ_DB8500_MSP0, pdata) diff --git a/arch/arm/mach-ux500/include/mach/msp.h b/arch/arm/mach-ux500/include/mach/msp.h new file mode 100644 index 0000000..798be19 --- /dev/null +++ b/arch/arm/mach-ux500/include/mach/msp.h @@ -0,0 +1,29 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * Author: Rabin Vincent <rabin.vincent(a)stericsson.com> for ST-Ericsson + * License terms: GNU General Public License (GPL), version 2. + */ + +#ifndef __MSP_H +#define __MSP_H + +#include <plat/ste_dma40.h> + +enum msp_i2s_id { + MSP_I2S_0 = 0, + MSP_I2S_1, + MSP_I2S_2, + MSP_I2S_3, +}; + +/* Platform data structure for a MSP I2S-device */ +struct msp_i2s_platform_data { + enum msp_i2s_id id; + struct stedma40_chan_cfg *msp_i2s_dma_rx; + struct stedma40_chan_cfg *msp_i2s_dma_tx; + int (*msp_i2s_init) (void); + int (*msp_i2s_exit) (void); +}; + +#endif -- 1.7.8.3

2 1

[alsa-devel] [PATCH 0/5] Alsa: snd-usb: Improve M-Audio Fast Track Ultra mixer - v3
by Felix Homann 24 Apr '12

24 Apr '12

This is the third try on a series of patches that improves the mixer for Fast Track Ultra (8R) devices by * adding TLV to existing controls * adding new controls for the effect section. In addition, it unifies the control creation for FTU and Ebox-44 devices. Mark, could you please look at mixer_quirks.c. Do we really need the range settings in snd_create_std_mono_ctl()? BTW, you might like to add some TLV callback to the Ebox-44 controls as well. I hope I really sorted Takashis issues out this time. Regards, Felix Felix Homann (5): Unify M-Audio Fast Track Ultra and Ebox-44 mixer quirks. Rename and export mixer_vol_tlv Add TLV to M-Audio Fast Track Ultra controls Rename Fast Track Ultra mixer quirk functions M-Audio Fast Track Ultra: Add effect controls sound/usb/mixer.c | 22 ++- sound/usb/mixer.h | 3 + sound/usb/mixer_quirks.c | 471 +++++++++++++++++++++++++++++++++++++++------- 3 files changed, 428 insertions(+), 68 deletions(-) -- 1.7.5.4

3 12

[alsa-devel] [PATCH 0/7] Alsa: snd-usb: Improve M-Audio Fast Track Ultra mixer - v2
by Felix Homann 24 Apr '12

24 Apr '12

This is the second try on a series of patches that improves the mixer for Fast Track Ultra (8R) devices by * adding TLV to existing controls * adding new controls for the effect section. It unifies the control creation for FTU and Ebox-44 devices. Mark, could you please look at mixer_quirks.c. Do we really need the range settings in snd_create_std_mono_ctl()? BTW, you might like to add some TLV callback to the Ebox-44 controls as well. I hope I sorted Takashis issues out this time. Regards, Felix Felix Homann (7): Unify M-Audio Fast Track Ultra and Ebox-44 mixer quirks. Rename and export mixer_vol_tlv Add TLV to M-Audio Fast Track Ultra controls Rename Fast Track Ultra mixer quirk functions Update comment for M-Audio FTU quirk M-Audio Fast Track Ultra: Add effect program switch M-Audio Fast Track Ultra: Add effect controls sound/usb/mixer.c | 22 ++- sound/usb/mixer.h | 3 + sound/usb/mixer_quirks.c | 472 +++++++++++++++++++++++++++++++++++++++------- 3 files changed, 429 insertions(+), 68 deletions(-) -- 1.7.5.4

3 15

[alsa-devel] [PATCH 2/8] ASoC: core: Add strobe control
by Ola Lilja 23 Apr '12

23 Apr '12

From: Kristoffer KARLSSON <kristoffer.karlsson(a)stericsson.com> Added support for a control that strobes a bit in a register to high then back to low (or the inverse). This is typically useful for hardware that requires strobing a singe bit to trigger some functionality and where exposing the bit in a normal single control would require the user to first manually set then again unset the bit again for the strobe to trigger. Added convenience macro. SOC_SINGLE_STROBE Added accessor implementations. snd_soc_get_strobe snd_soc_put_strobe Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson(a)stericsson.com> --- include/sound/soc.h | 8 ++++++ sound/soc/soc-core.c | 63 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 71 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8888adc..bda0cd2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -209,6 +209,10 @@ {.regbase = xregbase, .regcount = xregcount, .nbits = xnbits, \ .invert = xinvert, .min = xmin, .max = xmax} } +#define SOC_SINGLE_STROBE(xname, xreg, xshift, xinvert) \ + SOC_SINGLE_EXT(xname, xreg, xshift, 1, xinvert, \ + snd_soc_get_strobe, snd_soc_put_strobe) + /* * Simplified versions of above macros, declaring a struct and calculating * ARRAY_SIZE internally @@ -448,6 +452,10 @@ int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /** * struct snd_soc_reg_access - Describes whether a given register is diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7fc7294b..e9c5850 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2966,6 +2966,69 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); /** + * snd_soc_get_strobe - strobe get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback get the value of a strobe mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int val = snd_soc_read(codec, reg) & mask; + + if (shift != 0 && val != 0) + val = val >> shift; + ucontrol->value.enumerated.item[0] = val ^ invert; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_strobe); + +/** + * snd_soc_put_strobe - strobe put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback strobe a register bit to high then low (or the inverse) + * in one pass of a single mixer enum control. + * + * Returns 1 for success. + */ +int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int strobe = ucontrol->value.enumerated.item[0] != 0; + unsigned int val1 = (strobe ^ invert) ? mask : 0; + unsigned int val2 = (strobe ^ invert) ? 0 : mask; + int err; + + err = snd_soc_update_bits_locked(codec, reg, mask, val1); + if (err < 0) + return err; + + err = snd_soc_update_bits_locked(codec, reg, mask, val2); + return err; +} +EXPORT_SYMBOL_GPL(snd_soc_put_strobe); + +/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID -- 1.7.8.3

2 1

[alsa-devel] [PATCH 1/8] ASoC: core: Add signed multi register control
by Ola Lilja 23 Apr '12

23 Apr '12

From: Kristoffer KARLSSON <kristoffer.karlsson(a)stericsson.com> Added control type that can span multiple consecutive codec registers forming a single signed value in a MSB/LSB manner. The control dynamically adjusts to the register word size configured in driver. Added convenience macro. SOC_SINGLE_XR_SX Added accessor implementations. snd_soc_info_xr_sx snd_soc_get_xr_sx snd_soc_put_xr_sx Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson(a)stericsson.com> --- include/sound/soc.h | 22 +++++++++- sound/soc/soc-core.c | 118 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 139 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2ebf787..8888adc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -200,6 +200,15 @@ {.base = xbase, .num_regs = xregs, \ .mask = xmask }) } +#define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \ + xmin, xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = snd_soc_info_xr_sx, .get = snd_soc_get_xr_sx, \ + .put = snd_soc_put_xr_sx, \ + .private_value = (unsigned long)&(struct soc_mreg_control) \ + {.regbase = xregbase, .regcount = xregcount, .nbits = xnbits, \ + .invert = xinvert, .min = xmin, .max = xmax} } + /* * Simplified versions of above macros, declaring a struct and calculating * ARRAY_SIZE internally @@ -433,7 +442,12 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); - +int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /** * struct snd_soc_reg_access - Describes whether a given register is @@ -916,6 +930,12 @@ struct soc_bytes { u32 mask; }; +/* multi register control */ +struct soc_mreg_control { + long min, max; + unsigned int regbase, regcount, nbits, invert; +}; + /* enumerated kcontrol */ struct soc_enum { unsigned short reg; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e19c24a..7fc7294b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2848,6 +2848,124 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_bytes_put); /** + * snd_soc_info_xr_sx - signed multi register info callback + * @kcontrol: mreg control + * @uinfo: control element information + * + * Callback to provide information of a control that can + * span multiple codec registers which together + * forms a single signed value in a MSB/LSB manner. + * + * Returns 0 for success. + */ +int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = mc->min; + uinfo->value.integer.max = mc->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_xr_sx); + +/** + * snd_soc_get_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to get the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = codec->driver->reg_word_size * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long min = mc->min; + long max = mc->max; + long val = 0; + unsigned long regval; + unsigned int i; + + for (i = 0; i < regcount; i++) { + regval = snd_soc_read(codec, regbase+i) & regwmask; + val |= regval << (regwshift*(regcount-i-1)); + } + val &= mask; + if (min < 0 && val > max) + val |= ~mask; + if (invert) + val = max - val; + ucontrol->value.integer.value[0] = val; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_xr_sx); + +/** + * snd_soc_put_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to set the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = codec->driver->reg_word_size * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long min = mc->min; + long max = mc->max; + long val = ucontrol->value.integer.value[0]; + unsigned int i, regval, regmask; + int err; + + if (invert) + val = max - val; + val &= mask; + for (i = 0; i < regcount; i++) { + regval = (val >> (regwshift*(regcount-i-1))) & regwmask; + regmask = (mask >> (regwshift*(regcount-i-1))) & regwmask; + err = snd_soc_update_bits_locked(codec, regbase+i, + regmask, regval); + if (err < 0) + return err; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); + +/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID -- 1.7.8.3

2 1

[alsa-devel] [PATCH 3/8] ASoC: core: Add debug-print helper function
by Ola Lilja 23 Apr '12

23 Apr '12

Adds a function getting the stream-name as a string for a specific stream. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- include/sound/soc.h | 1 + sound/soc/soc-pcm.c | 9 +++++++++ 2 files changed, 10 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index bda0cd2..e89d309 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -345,6 +345,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +const char *snd_soc_stream_str(struct snd_pcm_substream *substream); /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 0ad8dca..6bec24f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -705,3 +705,12 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) cpu_dai->name); return ret; } + +/* Get a string naming the direction of a stream */ +const char *snd_soc_stream_str(struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} -- 1.7.8.3

2 1

[alsa-devel] ASoC updates for 3.4
by Mark Brown 23 Apr '12

23 Apr '12

The following changes since commit 8abe05c6eb358967f16bce8a02c88d57c82cfbd6: ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS (2012-04-06 12:28:49 +0100) 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 1a38336b8611a04f0a624330c1f815421f4bf5f4: ASoC: wm8994: Improve sequencing of AIF channel enables (2012-04-23 12:55:52 +0100) ---------------------------------------------------------------- ASoC: updates for 3.4 Slightly larger than normal - the DAPM fix is a "this should always have worked" type of thing which is very clear and should have no impact on systems that don't need it. The WM8994 fix is driver specific but pretty important for that driver. ---------------------------------------------------------------- Jesper Juhl (1): ASoC: cs42l73: don't use negative array index Liam Girdwood (1): ASoC: core: Fix card RTD count for deferred probe. Mark Brown (2): ASoC: dapm: Ensure power gets managed for line widgets ASoC: wm8994: Improve sequencing of AIF channel enables Paul Mundt (1): ASoC: fsi: update for dmaengine prep_slave_sg fallout. sound/soc/codecs/cs42l73.c | 2 + sound/soc/codecs/wm8994.c | 276 +++++++++++++++++++++++++++++++++++--------- sound/soc/sh/fsi.c | 7 +- sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 2 + 5 files changed, 230 insertions(+), 58 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: dapm: Fix x86_64 build warning.
by Liam Girdwood 23 Apr '12

23 Apr '12

Fixes the following build warning on x86_64. In file included from include/trace/ftrace.h:567:0, from include/trace/define_trace.h:86, from include/trace/events/asoc.h:410, from sound/soc/soc-core.c:45: include/trace/events/asoc.h: In function 'ftrace_raw_event_snd_soc_dapm_output_path': include/trace/events/asoc.h:246:1: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] include/trace/events/asoc.h: In function 'ftrace_raw_event_snd_soc_dapm_input_path': include/trace/events/asoc.h:275:1: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/trace/events/asoc.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 6d8efb1..5fc2dcd 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -263,7 +263,7 @@ TRACE_EVENT(snd_soc_dapm_output_path, __assign_str(pname, path->name ? path->name : DAPM_DIRECT); __assign_str(psname, path->sink->name); __entry->path_connect = path->connect; - __entry->path_sink = (int)path->sink; + __entry->path_sink = (long)path->sink; ), TP_printk("%c%s -> %s -> %s\n", @@ -292,7 +292,7 @@ TRACE_EVENT(snd_soc_dapm_input_path, __assign_str(pname, path->name ? path->name : DAPM_DIRECT); __assign_str(psname, path->source->name); __entry->path_connect = path->connect; - __entry->path_source = (int)path->source; + __entry->path_source = (long)path->source; ), TP_printk("%c%s <- %s <- %s\n", -- 1.7.5.4

4 5

[alsa-devel] wm9713 problem
by jarek 23 Apr '12

23 Apr '12

Hi! I've problem with ARM board with WM9713 codec. I have very simple test application, which reads samples with snd_pcm_readi and then writes them with snd_pcm_writei. I've tested this on few PC-s and it works fine. However on FirendlyARM tiny6410 it hangs after few seconds on call to snd_pcm_readi. What is more funny If I'm writing samples with 0 (silence), reading works fine. It hangs only if I'm writing non-silence. If I break program with Ctrl+C, I can see in dmesg: s3c64xx_dma_stop: channel still active. and audio driver is not working any more. I need to restart board to make it working. What it can be ? Where to look for problem ? The kernel is 2.6.38 PREEMPT best regards Jarek

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Improve sequencing of AIF channel enables
by Mark Brown 23 Apr '12

23 Apr '12

This ensures a clean startup of the channels, without this change some use cases could result in issues in a small proportion of cases. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 276 ++++++++++++++++++++++++++++++++++++--------- 1 file changed, 222 insertions(+), 54 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7c49642..6c1fe3a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1000,61 +1000,170 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) } } -static int late_enable_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif1clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; + int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; + int dac; + int adc; + int val; + + switch (control->type) { + case WM8994: + case WM8958: + mask |= WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA; + break; + default: + break; + } switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (wm8994->aif1clk_enable) { - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA_MASK, - WM8994_AIF1CLK_ENA); - wm8994->aif1clk_enable = 0; - } - if (wm8994->aif2clk_enable) { - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA_MASK, - WM8994_AIF2CLK_ENA); - wm8994->aif2clk_enable = 0; - } + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_1); + if ((val & WM8994_AIF1ADCL_SRC) && + (val & WM8994_AIF1ADCR_SRC)) + adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA; + else if (!(val & WM8994_AIF1ADCL_SRC) && + !(val & WM8994_AIF1ADCR_SRC)) + adc = WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA; + else + adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA | + WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA; + + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_2); + if ((val & WM8994_AIF1DACL_SRC) && + (val & WM8994_AIF1DACR_SRC)) + dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA; + else if (!(val & WM8994_AIF1DACL_SRC) && + !(val & WM8994_AIF1DACR_SRC)) + dac = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA; + else + dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA | + WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA; + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + mask, adc); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, dac); + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_AIF1DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA, + WM8994_AIF1DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, mask, + WM8994_AIF1ADC1R_ENA | + WM8994_AIF1ADC1L_ENA | + WM8994_AIF1ADC2R_ENA | + WM8994_AIF1ADC2L_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, mask, + WM8994_AIF1DAC1R_ENA | + WM8994_AIF1DAC1L_ENA | + WM8994_AIF1DAC2R_ENA | + WM8994_AIF1DAC2L_ENA); break; - } - /* We may also have postponed startup of DSP, handle that. */ - wm8958_aif_ev(w, kcontrol, event); + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, 0); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + mask, 0); + + val = snd_soc_read(codec, WM8994_CLOCKING_1); + if (val & WM8994_AIF2DSPCLK_ENA) + val = WM8994_SYSDSPCLK_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSDSPCLK_ENA | + WM8994_AIF1DSPCLK_ENA, val); + break; + } return 0; } -static int late_disable_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif2clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + int dac; + int adc; + int val; switch (event) { + case SND_SOC_DAPM_PRE_PMU: + val = snd_soc_read(codec, WM8994_AIF2_CONTROL_1); + if ((val & WM8994_AIF2ADCL_SRC) && + (val & WM8994_AIF2ADCR_SRC)) + adc = WM8994_AIF2ADCR_ENA; + else if (!(val & WM8994_AIF2ADCL_SRC) && + !(val & WM8994_AIF2ADCR_SRC)) + adc = WM8994_AIF2ADCL_ENA; + else + adc = WM8994_AIF2ADCL_ENA | WM8994_AIF2ADCR_ENA; + + + val = snd_soc_read(codec, WM8994_AIF2_CONTROL_2); + if ((val & WM8994_AIF2DACL_SRC) && + (val & WM8994_AIF2DACR_SRC)) + dac = WM8994_AIF2DACR_ENA; + else if (!(val & WM8994_AIF2DACL_SRC) && + !(val & WM8994_AIF2DACR_SRC)) + dac = WM8994_AIF2DACL_ENA; + else + dac = WM8994_AIF2DACL_ENA | WM8994_AIF2DACR_ENA; + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, adc); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, dac); + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_AIF2DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA, + WM8994_AIF2DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA); + break; + + case SND_SOC_DAPM_PRE_PMD: case SND_SOC_DAPM_POST_PMD: - if (wm8994->aif1clk_disable) { - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA_MASK, 0); - wm8994->aif1clk_disable = 0; - } - if (wm8994->aif2clk_disable) { - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA_MASK, 0); - wm8994->aif2clk_disable = 0; - } + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, 0); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, 0); + + val = snd_soc_read(codec, WM8994_CLOCKING_1); + if (val & WM8994_AIF1DSPCLK_ENA) + val = WM8994_SYSDSPCLK_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSDSPCLK_ENA | + WM8994_AIF2DSPCLK_ENA, val); break; } return 0; } -static int aif1clk_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif1clk_late_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -1071,8 +1180,8 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, return 0; } -static int aif2clk_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif2clk_late_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -1089,6 +1198,63 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int late_enable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wm8994->aif1clk_enable) { + aif1clk_ev(w, kcontrol, event); + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, + WM8994_AIF1CLK_ENA); + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { + aif2clk_ev(w, kcontrol, event); + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, + WM8994_AIF2CLK_ENA); + wm8994->aif2clk_enable = 0; + } + break; + } + + /* We may also have postponed startup of DSP, handle that. */ + wm8958_aif_ev(w, kcontrol, event); + + return 0; +} + +static int late_disable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + if (wm8994->aif1clk_disable) { + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, 0); + aif1clk_ev(w, kcontrol, event); + wm8994->aif1clk_disable = 0; + } + if (wm8994->aif2clk_disable) { + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, 0); + aif2clk_ev(w, kcontrol, event); + wm8994->aif2clk_disable = 0; + } + break; + } + + return 0; +} + static int adc_mux_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1385,9 +1551,9 @@ static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = { -SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev, +SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_late_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev, +SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_late_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, @@ -1416,8 +1582,10 @@ SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) }; static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { -SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, aif1clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, aif2clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), @@ -1470,30 +1638,30 @@ SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event, SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSP2CLK", SND_SOC_NOPM, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPINTCLK", SND_SOC_NOPM, 1, 0, NULL, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 9, 0), + 0, SND_SOC_NOPM, 9, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 8, 0), + 0, SND_SOC_NOPM, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, + SND_SOC_NOPM, 9, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, + SND_SOC_NOPM, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 11, 0), + 0, SND_SOC_NOPM, 11, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 10, 0), + 0, SND_SOC_NOPM, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, + SND_SOC_NOPM, 11, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2R", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 10, 0, wm8958_aif_ev, + SND_SOC_NOPM, 10, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0, @@ -1520,14 +1688,14 @@ SND_SOC_DAPM_MIXER("DAC1R Mixer", SND_SOC_NOPM, 0, 0, dac1r_mix, ARRAY_SIZE(dac1r_mix)), SND_SOC_DAPM_AIF_OUT("AIF2ADCL", NULL, 0, - WM8994_POWER_MANAGEMENT_4, 13, 0), + SND_SOC_NOPM, 13, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCR", NULL, 0, - WM8994_POWER_MANAGEMENT_4, 12, 0), + SND_SOC_NOPM, 12, 0), SND_SOC_DAPM_AIF_IN_E("AIF2DACL", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 13, 0, wm8958_aif_ev, + SND_SOC_NOPM, 13, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 12, 0, wm8958_aif_ev, + SND_SOC_NOPM, 12, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_AIF_IN("AIF1DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0), -- 1.7.10

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Re: [alsa-devel] [PATCH] ASoC: omap-pcm: Free dma buffers in case of error.
by Mark Brown 23 Apr '12

23 Apr '12

On Fri, Apr 20, 2012 at 05:59:25PM +0000, Matcovschi, Oleg wrote: > Should I resend patch again including maintainers? Please, I didn't see it so I'll need a copy if I'm going to review it. > Sorry, my first patch. Will do in future patches. No worries.

1 0

[alsa-devel] ALSA HDMI Audio Hot Plug Detect functionality
by Raj, Deepu 23 Apr '12

23 Apr '12

All, Can someone help me on how to implement the HPD function in ALSA soc driver? In my case the Video driver is already registered an IRQ for the HDP detect and doing all the configurations for the video part. How can I integrate a call or some other mechanism to the dummy codec driver where I am using the audio configurations? All the hdmi configuration functions are implemented in a hdmi_lib.c and accessed from the dummy codec driver. Thanks & Regards Deepu Raj

1 0

[alsa-devel] [PATCH 0/3] Alsa: USB: Improve M-Audio Fast Track Ultra mixer
by Felix Homann 23 Apr '12

23 Apr '12

This series of patches improves the mixer for Fast Track Ultra (8R) devices by * adding TLV to existing controls * adding new controls for the effect section. It unifies the control creation for FTU and Ebox-44 devices. Mark, could you please look at mixer_quirks.c. Do we really need the range settings in snd_create_std_mono_ctl()? BTW, you might like to add some TLV callback to the Ebox-44 controls as well. Regards, Felix Felix Homann (3): Unify M-Audio Fast Track Ultra and Ebox-44 mixer quirks. Add TLV to M-Audio volume controls M-Audio FTU: Add effect controls sound/usb/mixer.c | 23 +++ sound/usb/mixer_quirks.c | 493 +++++++++++++++++++++++++++++++++++++++------ 2 files changed, 450 insertions(+), 66 deletions(-) -- 1.7.5.4

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[alsa-devel] [PATCH] alsa-lib conf: Add two more USB devices to iec958 blacklist
by David Henningsson 23 Apr '12

23 Apr '12

Add "Logitech Wireless Headset" and "Logitech Speaker Lapdesk N700" to the iec958 blacklist. Also reorder the list in alphabetic order. BugLink: https://bugs.launchpad.net/bugs/987163 Reported-by: Jean-Baptiste Lallement <jean-baptiste(a)ubuntu.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index b9148e7..0a0e374 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -38,10 +38,12 @@ USB-Audio.pcm.iec958_device { # "NoiseBlaster 3000" 42 # The below don't have digital in/out, so prevent them from being opened. - "Sennheiser USB headset" 999 + "Blue Snowball" 999 + "Logitech Speaker Lapdesk N700" 999 "Logitech USB Headset" 999 + "Logitech Wireless Headset" 999 + "Sennheiser USB headset" 999 "USB Device 0x46d:0x992" 999 - "Blue Snowball" 999 } -- 1.7.5.4

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[alsa-devel] HDSPM: Wrong samplerate on PCI cards
by Andre Schramm 22 Apr '12

22 Apr '12

Hi, we're having issues obtaining the correct system sample rate from PCI cards, while it works correctly for PCIe cards. I tested with stock ALSA-driver 1.0.{24,25} as well as latest Repo-Version via alsa-compile.sh on 64b-Kernel 3.0.9-RT25. I added some printks, the culprit seems to be: hdspm.c:1988: period = hdspm_read(hdspm, HDSPM_RD_PLL_FREQ); Output for PCIe card: kernel: ALSA sound/pci/rme9652/hdspm.c:1379 HDSPM: hdspm_calc_dds_value: freq_const=110069313433624, period=2293110696, freq_const/period=48000 Output for PCI card: kernel: ALSA sound/pci/rme9652/hdspm.c:1379 HDSPM: hdspm_calc_dds_value: freq_const=110069313433624, period=41943044, freq_const/period=2624256 I tested with 3 different PCI cards (latest firmware) and the same card(s) works as expected on 32b-Kernel 2.6.24.7 with ALSA 1.0.16. The card's registers look exactly the same when applying the same card settings via amixer. Starting JACKd works on both cards and audio output is fine. Within the driver, I don't see a different code path for PCI and PCIe, so this seems weird. I took a look at the Mac driver as well, it defines RD_PLL_FREQ as 4*128, so this should be ok? Can anybody confirm this? And any ideas, what's going on here? I noticed a mismatch between "struct hdspm" (int system_sample_rate) and "struct hdspm_config" (unsigned int system_sample_rate), but this doesn't seem to be an issue? And the proc-output for the samplerate is wrong/misleading as it isn't read directly from the card as opposed to the RayDat-Code. Is there any reason for this? Detailed output below (card 0 is PCIe, card 1 is PCI). Note: amixer is 64b as well, so the 32b compat stuff shouldn't matter. Cheers, Andre # cat /proc/asound/cards 0 [HDSPMx1c9359 ]: HDSPM - RME MADI_1c9359 RME MADI S/N 0x1c9359 at 0xfe900000, irq 17 1 [Default ]: HDSPM - RME MADI_ffffff RME MADI S/N 0xffffff at 0xfeb00000, irq 17 # amixer -c0 cget numid=6 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=48000 # amixer -c1 cget numid=6 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=2624256 # cat /proc/asound/card0/hdspm RME MADI (Card #1) Rev.d2 Status2first3bits: 1 HW Serial: 0x0000011c9359 IRQ: 17 Registers bus: 0xfe900000 VM: 0xffffc900114e0000 --- System --- IRQ Pending: Audio=0, MIDI0=0, MIDI1=0, IRQcount=0 HW pointer: id = 0, rawptr = 0 (0->16320) estimated= 0 (bytes) MIDI FIFO: Out1=0x0, Out2=0x0, In1=0x80, In2=0x80 MIDIoverMADI FIFO: In=0x0, Out=0x0 Register: ctrl1=0x10044da, ctrl2=0x0, status1=0x4, status2=0xc0020701 No TCO module detected. --- Settings --- Size (Latency): 2048 samples (2 periods of 8192 bytes) Line out: on ClearTrackMarker = off, Transmit in 64 Channel Mode, Auto Input off AutoSync Reference: Master Preferred Sync Reference: Word Clock System Clock Frequency: 48000 --- Status: Inputs MADI=NoLock, WordClock=NoLock AutoSync: Reference= Input not valid, Freq=0 (MADI = 0, Word = 0) Input: Coax, Mode=56 channels # cat /proc/asound/card1/hdspm RME MADI (Card #2) Rev.cf Status2first3bits: 1 HW Serial: 0xffffffffffff IRQ: 17 Registers bus: 0xfeb00000 VM: 0xffffc90011420000 --- System --- IRQ Pending: Audio=0, MIDI0=0, MIDI1=0, IRQcount=0 HW pointer: id = 0, rawptr = 0 (0->16320) estimated= 0 (bytes) MIDI FIFO: Out1=0x0, Out2=0x0, In1=0x0, In2=0x0 MIDIoverMADI FIFO: In=0x0, Out=0x0 Register: ctrl1=0x10044da, ctrl2=0x0, status1=0x1000004, status2=0xffff0701 No TCO module detected. --- Settings --- Size (Latency): 2048 samples (2 periods of 8192 bytes) Line out: on ClearTrackMarker = off, Transmit in 64 Channel Mode, Auto Input off AutoSync Reference: Master Preferred Sync Reference: Word Clock System Clock Frequency: 48000 --- Status: Inputs MADI=NoLock, WordClock=NoLock AutoSync: Reference= Input not valid, Freq=0 (MADI = 4, Word = 0) Input: Coax, Mode=56 channels # cat /proc/asound/card0/debug 0x00000000: 00000004 00000004 00000004 00000004 0x00000010: 00000004 00000004 00000004 00000004 0x00000020: 00000004 00000004 00000004 00000004 0x00000030: 00000004 00000004 00000004 00000004 0x00000040: 00000004 00000004 00000004 00000004 0x00000050: 00000004 00000004 00000004 00000004 0x00000060: 00000004 00000004 00000004 00000004 0x00000070: 00000004 00000004 00000004 00000004 0x00000080: 00000004 00000004 00000004 00000004 0x00000090: 00000004 00000004 00000004 00000004 0x000000A0: 00000004 00000004 00000004 00000004 0x000000B0: 00000004 00000004 00000004 00000004 0x000000C0: C0020701 C0020701 C0020701 C0020701 0x000000D0: C0020701 C0020701 C0020701 C0020701 0x000000E0: C0020701 C0020701 C0020701 C0020701 0x000000F0: C0020701 C0020701 C0020701 C0020701 # cat /proc/asound/card1/debug 0x00000000: 02800004 02800004 02800004 02800004 0x00000010: 02800004 02800004 02800004 02800004 0x00000020: 02800004 02800004 02800004 02800004 0x00000030: 02800004 02800004 02800004 02800004 0x00000040: 02800004 02800004 02800004 02800004 0x00000050: 02800004 02800004 02800004 02800004 0x00000060: 02800004 02800004 02800004 02800004 0x00000070: 02800004 02800004 02800004 02800004 0x00000080: FFFFFFFF FFFFFFFF FFFFFFFF FFFFFFFF 0x00000090: FFFFFFFF FFFFFFFF FFFFFFFF FFFFFFFF 0x000000A0: FFFFFFFF FFFFFFFF FFFFFFFF FFFFFFFF 0x000000B0: FFFFFFFF FFFFFFFF FFFFFFFF FFFFFFFF 0x000000C0: FFFF0701 FFFF0701 FFFF0701 FFFF0701 0x000000D0: FFFF0701 FFFF0701 FFFF0701 FFFF0701 0x000000E0: FFFF0701 FFFF0701 FFFF0701 FFFF0701 0x000000F0: FFFF0701 FFFF0701 FFFF0701 FFFF0701

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[alsa-devel] [PATCH] ALSA: snd-usb: remove refactorization left-overs
by Daniel Mack 21 Apr '12

21 Apr '12

Drop some struct members and definitions that became obsolete during the refactorization of the driver. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/card.h | 13 ------------- sound/usb/proc.c | 6 +----- 2 files changed, 1 insertions(+), 18 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 77d2eec..0d37238 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -43,13 +43,6 @@ struct snd_urb_ctx { struct list_head ready_list; }; -struct snd_urb_ops { - int (*prepare)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*retire)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*prepare_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*retire_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); -}; - struct snd_usb_endpoint { struct snd_usb_audio *chip; @@ -125,11 +118,6 @@ struct snd_usb_substream { unsigned long active_mask; /* bitmask of active urbs */ unsigned long unlink_mask; /* bitmask of unlinked urbs */ - unsigned int nurbs; /* # urbs */ - struct snd_urb_ctx dataurb[MAX_URBS]; /* data urb table */ - struct snd_urb_ctx syncurb[SYNC_URBS]; /* sync urb table */ - char *syncbuf; /* sync buffer for all sync URBs */ - dma_addr_t sync_dma; /* DMA address of syncbuf */ /* data and sync endpoints for this stream */ struct snd_usb_endpoint *data_endpoint; struct snd_usb_endpoint *sync_endpoint; @@ -141,7 +129,6 @@ struct snd_usb_substream { struct snd_pcm_hw_constraint_list rate_list; /* limited rates */ spinlock_t lock; - struct snd_urb_ops ops; /* callbacks (must be filled at init) */ int last_frame_number; /* stored frame number */ int last_delay; /* stored delay */ }; diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 06e23d8..ebc1a5b 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -25,6 +25,7 @@ #include "usbaudio.h" #include "helper.h" #include "card.h" +#include "endpoint.h" #include "proc.h" /* convert our full speed USB rate into sampling rate in Hz */ @@ -137,14 +138,9 @@ static void proc_dump_ep_status(struct snd_usb_substream *subs, static void proc_dump_substream_status(struct snd_usb_substream *subs, struct snd_info_buffer *buffer) { if (subs->running) { - unsigned int i; snd_iprintf(buffer, " Status: Running\n"); snd_iprintf(buffer, " Interface = %d\n", subs->interface); snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx); - snd_iprintf(buffer, " URBs = %d [ ", subs->nurbs); - for (i = 0; i < subs->nurbs; i++) - snd_iprintf(buffer, "%d ", subs->dataurb[i].packets); - snd_iprintf(buffer, "]\n"); proc_dump_ep_status(subs, subs->data_endpoint, buffer); proc_dump_ep_status(subs, subs->sync_endpoint, buffer); } else { -- 1.7.5.4

2 1

[alsa-devel] Fast Track Ultra (8R): Sync source switch
by Felix Homann 20 Apr '12

20 Apr '12

Hi, I've taken a look again at the usbmon logs I posted some years ago. It seems that by switching the sync source you essentially tell the FTU to use a different altsetting for iface 2: Switching "Sync Source" from "Internal" to "External" (see last line): ffff8800662a60c0 3325542111 S Co:1:006:0 s 01 0b 0000 0001 0000 0 ffff8800662a60c0 3325543386 C Co:1:006:0 0 0 ffff88009159e740 3325594872 S Co:1:006:0 s 01 0b 0000 0002 0000 0 ffff88009159e740 3325596263 C Co:1:006:0 0 0 ffff8800a00d0900 3325633405 S Co:1:006:0 s 01 0b 0001 0001 0000 0 ffff8800a00d0900 3325633637 C Co:1:006:0 0 0 ffff88009159e440 3325644251 S Co:1:006:0 s 01 0b 0002 0002 0000 0 Switching "Sync Source" from "External" to "Internal" (see last line): ffff8800a00d0480 3349133460 S Co:1:006:0 s 01 0b 0000 0001 0000 0 ffff8800a00d0480 3349134696 C Co:1:006:0 0 0 ffff88009159e740 3349186489 S Co:1:006:0 s 01 0b 0000 0002 0000 0 ffff88009159e740 3349187696 C Co:1:006:0 0 0 ffff88009159e440 3349239544 S Co:1:006:0 s 01 0b 0001 0001 0000 0 ffff88009159e440 3349239823 C Co:1:006:0 0 0 ffff8800a00d0900 3349255385 S Co:1:006:0 s 01 0b 0001 0002 0000 0 If I haven't missed something we can achieve this by adding a quirk hook to either find_format() or set_format() in pcm.c. Daniel, can you think of a generic way to do this? (Note to all: can't test anything on this topic because I don't have a S/PDIF device here.) BTW, has anybody tested the mixer patch I posted a couple of days ago? Regards, Felix

2 1

Re: [alsa-devel] [PATCH] ASoC: omap-pcm: Free dma buffers in case of error.
by Ujfalusi, Peter 20 Apr '12

20 Apr '12

Hi, On Thu, Apr 19, 2012 at 3:37 AM, Oleg Matcovschi <oleg.matcovschi(a)ti.com> wrote: > Signed-off-by: Oleg Matcovschi <oleg.matcovschi(a)ti.com> > --- > sound/soc/omap/omap-pcm.c | 4 ++++ > 1 files changed, 4 insertions(+), 0 deletions(-) > > diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c > index a2a464f..380a5bd 100644 > --- a/sound/soc/omap/omap-pcm.c > +++ b/sound/soc/omap/omap-pcm.c > @@ -414,6 +414,10 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) > } > > out: > + /* free preallocated buffers in case of error */ > + if (ret) > + omap_pcm_free_dma_buffers(pcm); > + If the playback preallocation fails, or if only capture is supported on the dai link this is not needed. It only applies if we have both playback and capture streams and the capture preallocation fails. Luckily the omap_pcm_free_dma_buffers() has internal checks for these cases... Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com>

2 1

[alsa-devel] [PATCH 0/8] Third edition of Ux500 ASoC-driver
by Ola Lilja 20 Apr '12

20 Apr '12

This is the third version and since last version there has been alot of rewriting and improvments. All according to the comments of Mark. Highlights: - Moved all regulator and clock-code from machine into codec. - Using DMA-engine PCM-interface. - Removed all virtual enum-switches to pin-switches. - Cleaned up I2S-abstraction and rearranged it in a better way. - Moved device-initialization into mach-tree. Kristoffer KARLSSON (2): ASoC: core: Add signed multi register control ASoC: core: Add strobe control Ola Lilja (6): ASoC: core: Add debug-print helper function arm: ux500: Add support for MSP I2S-devices ASoC: Ux500: Add MSP I2S-driver ASoC: Ux500: Add platform-driver ASoC: codecs: Add AB8500 codec-driver ASoC: Ux500: Add machine-driver arch/arm/mach-ux500/Makefile | 3 +- arch/arm/mach-ux500/board-mop500-msp.c | 250 +++ arch/arm/mach-ux500/board-mop500-msp.h | 14 + arch/arm/mach-ux500/board-mop500.c | 4 + arch/arm/mach-ux500/clock.c | 8 +- arch/arm/mach-ux500/devices-db8500.h | 10 - arch/arm/mach-ux500/include/mach/msp.h | 29 + include/sound/soc.h | 31 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/ab8500-codec.c | 3028 ++++++++++++++++++++++++++++++++ sound/soc/codecs/ab8500-codec.h | 603 +++++++ sound/soc/soc-core.c | 181 ++ sound/soc/soc-pcm.c | 9 + sound/soc/ux500/Kconfig | 34 + sound/soc/ux500/Makefile | 11 + sound/soc/ux500/u8500.c | 118 ++ sound/soc/ux500/ux500_ab8500.c | 225 +++ sound/soc/ux500/ux500_ab8500.h | 21 + sound/soc/ux500/ux500_msp_dai.c | 876 +++++++++ sound/soc/ux500/ux500_msp_dai.h | 80 + sound/soc/ux500/ux500_msp_i2s.c | 742 ++++++++ sound/soc/ux500/ux500_msp_i2s.h | 553 ++++++ sound/soc/ux500/ux500_pcm.c | 332 ++++ sound/soc/ux500/ux500_pcm.h | 42 + 27 files changed, 7198 insertions(+), 15 deletions(-) create mode 100644 arch/arm/mach-ux500/board-mop500-msp.c create mode 100644 arch/arm/mach-ux500/board-mop500-msp.h create mode 100644 arch/arm/mach-ux500/include/mach/msp.h create mode 100644 sound/soc/codecs/ab8500-codec.c create mode 100644 sound/soc/codecs/ab8500-codec.h create mode 100644 sound/soc/ux500/Kconfig create mode 100644 sound/soc/ux500/Makefile create mode 100644 sound/soc/ux500/u8500.c create mode 100644 sound/soc/ux500/ux500_ab8500.c create mode 100644 sound/soc/ux500/ux500_ab8500.h create mode 100644 sound/soc/ux500/ux500_msp_dai.c create mode 100644 sound/soc/ux500/ux500_msp_dai.h create mode 100644 sound/soc/ux500/ux500_msp_i2s.c create mode 100644 sound/soc/ux500/ux500_msp_i2s.h create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h -- 1.7.8.3

1 0

[alsa-devel] [PATCH] ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E
by David Henningsson 20 Apr '12

20 Apr '12

According to the reporter, external mic starts to work if the laptop-dmic model is used. According to BIOS pin config, all pins are consistent with the alc269vb_laptop_dmic fixup, except for the external mic, which is not present. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/950490 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 e65e354..818f90b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6109,6 +6109,7 @@ static const struct alc_fixup alc269_fixups[] = { static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), -- 1.7.9.5

2 1

Re: [alsa-devel] M-Audio FastTrack C400
by Chris Cavey 19 Apr '12

19 Apr '12

Here we go - disconnect, reconnect, wait, disconnect, reconnect [ 81.626968] usb 1-1: USB disconnect, device number 2 [ 96.644352] usb 1-1: new high-speed USB device number 3 using ehci_hcd [ 96.854092] snd-usb-audio: probe of 1-1:1.1 failed with error -5 [ 96.860196] snd-usb-audio: probe of 1-1:1.2 failed with error -5 [ 96.869071] snd-usb-audio: probe of 1-1:1.3 failed with error -5 [ 96.886969] input: M-Audio Fast Track C400 as /devices/pci0000:00/0000:00:0b.0/usb1/1-1/1-1:1.5/input/input7 [ 96.887558] generic-usb 0003:0763:2030.0003: input,hidraw0: USB HID v1.10 Keyboard [M-Audio Fast Track C400] on usb-0000:00:0b.0-1/input5 [ 132.340302] usb 1-1: USB disconnect, device number 3 [ 140.444314] usb 1-1: new high-speed USB device number 4 using ehci_hcd [ 140.665648] snd-usb-audio: probe of 1-1:1.1 failed with error -5 [ 140.671506] snd-usb-audio: probe of 1-1:1.2 failed with error -5 [ 140.680480] snd-usb-audio: probe of 1-1:1.3 failed with error -5 [ 140.695583] input: M-Audio Fast Track C400 as /devices/pci0000:00/0000:00:0b.0/usb1/1-1/1-1:1.5/input/input8 [ 140.696324] generic-usb 0003:0763:2030.0004: input,hidraw0: USB HID v1.10 Keyboard [M-Audio Fast Track C400] on usb-0000:00:0b.0-1/input5 On 04/19/2012 07:20 AM, Grant Diffey wrote: > oh... turn the interface off and on and give us the last 10 lines. > > On Thu, Apr 19, 2012 at 9:19 PM, Chris Cavey <chris-alsa(a)rauros.net > <mailto:chris-alsa@rauros.net>> wrote: > > entire log > > [ 0.000000] Initializing cgroup subsys cpuset > [ 0.000000] Initializing cgroup subsys cpu > [ 0.000000] Linux version 3.2.0-23-generic (buildd@crested) > (gcc version 4.6.3 (Ubuntu/Linaro 4.6.3-1ubuntu4) ) #36-Ubuntu SMP > Tue Apr 10 20:39:51 UTC 2012 (Ubuntu 3.2.0-23.36-generic 3.2.14) > [ 0.000000] Command line: > BOOT_IMAGE=/boot/vmlinuz-3.2.0-23-generic > root=UUID=ec96128d-feb6-40e6-a5aa-b55aa2219935 ro > [ 0.000000] KERNEL supported cpus: > [ 0.000000] Intel GenuineIntel > [ 0.000000] AMD AuthenticAMD > [ 0.000000] Centaur CentaurHauls > [ 0.000000] BIOS-provided physical RAM map: > [ 0.000000] BIOS-e820: 0000000000000000 - 000000000009fc00 > (usable) > [ 0.000000] BIOS-e820: 000000000009fc00 - 00000000000a0000 > (reserved) > [ 0.000000] BIOS-e820: 00000000000f0000 - 0000000000100000 > (reserved) > [ 0.000000] BIOS-e820: 0000000000100000 - 00000000dfff0000 > (usable) > [ 0.000000] BIOS-e820: 00000000dfff0000 - 00000000e0000000 > (ACPI data) > [ 0.000000] BIOS-e820: 00000000fffc0000 - 0000000100000000 > (reserved) > [ 0.000000] BIOS-e820: 0000000100000000 - 0000000120000000 > (usable) > [ 0.000000] NX (Execute Disable) protection: active > [ 0.000000] DMI 2.5 present. > [ 0.000000] DMI: innotek GmbH VirtualBox, BIOS VirtualBox > 12/01/2006 > [ 0.000000] e820 update range: 0000000000000000 - > 0000000000010000 (usable) ==> (reserved) > [ 0.000000] e820 remove range: 00000000000a0000 - > 0000000000100000 (usable) > [ 0.000000] No AGP bridge found > [ 0.000000] last_pfn = 0x120000 max_arch_pfn = 0x400000000 > [ 0.000000] MTRR default type: uncachable > [ 0.000000] MTRR variable ranges disabled: > [ 0.000000] x86 PAT enabled: cpu 0, old 0x7040600070406, new > 0x7010600070106 > [ 0.000000] CPU MTRRs all blank - virtualized system. > [ 0.000000] last_pfn = 0xdfff0 max_arch_pfn = 0x400000000 > [ 0.000000] found SMP MP-table at [ffff88000009fff0] 9fff0 > [ 0.000000] initial memory mapped : 0 - 20000000 > [ 0.000000] Base memory trampoline at [ffff88000009a000] 9a000 > size 20480 > [ 0.000000] init_memory_mapping: 0000000000000000-00000000dfff0000 > [ 0.000000] 0000000000 - 00dfe00000 page 2M > [ 0.000000] 00dfe00000 - 00dfff0000 page 4k > [ 0.000000] kernel direct mapping tables up to dfff0000 @ > 1fffa000-20000000 > [ 0.000000] init_memory_mapping: 0000000100000000-0000000120000000 > [ 0.000000] 0100000000 - 0120000000 page 2M > [ 0.000000] kernel direct mapping tables up to 120000000 @ > dffea000-dfff0000 > [ 0.000000] RAMDISK: 364ec000 - 3726e000 > [ 0.000000] ACPI: RSDP 00000000000e0000 00024 (v02 VBOX ) > [ 0.000000] ACPI: XSDT 00000000dfff0030 0003C (v01 VBOX > VBOXXSDT 00000001 ASL 00000061) > [ 0.000000] ACPI: FACP 00000000dfff00f0 000F4 (v04 VBOX > VBOXFACP 00000001 ASL 00000061) > [ 0.000000] ACPI: DSDT 00000000dfff0480 01B96 (v01 VBOX > VBOXBIOS 00000002 INTL 20100528) > [ 0.000000] ACPI: FACS 00000000dfff0200 00040 > [ 0.000000] ACPI: APIC 00000000dfff0240 0006C (v02 VBOX > VBOXAPIC 00000001 ASL 00000061) > [ 0.000000] ACPI: SSDT 00000000dfff02b0 001CC (v01 VBOX > VBOXCPUT 00000002 INTL 20100528) > [ 0.000000] ACPI: Local APIC address 0xfee00000 > [ 0.000000] No NUMA configuration found > [ 0.000000] Faking a node at 0000000000000000-0000000120000000 > [ 0.000000] Initmem setup node 0 0000000000000000-0000000120000000 > [ 0.000000] NODE_DATA [000000011fffb000 - 000000011fffffff] > [ 0.000000] [ffffea0000000000-ffffea00047fffff] PMD -> > [ffff88011b600000-ffff88011f5fffff] on node 0 > [ 0.000000] Zone PFN ranges: > [ 0.000000] DMA 0x00000010 -> 0x00001000 > [ 0.000000] DMA32 0x00001000 -> 0x00100000 > [ 0.000000] Normal 0x00100000 -> 0x00120000 > [ 0.000000] Movable zone start PFN for each node > [ 0.000000] early_node_map[3] active PFN ranges > [ 0.000000] 0: 0x00000010 -> 0x0000009f > [ 0.000000] 0: 0x00000100 -> 0x000dfff0 > [ 0.000000] 0: 0x00100000 -> 0x00120000 > [ 0.000000] On node 0 totalpages: 1048447 > [ 0.000000] DMA zone: 64 pages used for memmap > [ 0.000000] DMA zone: 5 pages reserved > [ 0.000000] DMA zone: 3914 pages, LIFO batch:0 > [ 0.000000] DMA32 zone: 16320 pages used for memmap > [ 0.000000] DMA32 zone: 897072 pages, LIFO batch:31 > [ 0.000000] Normal zone: 2048 pages used for memmap > [ 0.000000] Normal zone: 129024 pages, LIFO batch:31 > [ 0.000000] ACPI: PM-Timer IO Port: 0x4008 > [ 0.000000] ACPI: Local APIC address 0xfee00000 > [ 0.000000] ACPI: LAPIC (acpi_id[0x00] lapic_id[0x00] enabled) > [ 0.000000] ACPI: LAPIC (acpi_id[0x01] lapic_id[0x01] enabled) > [ 0.000000] ACPI: LAPIC (acpi_id[0x02] lapic_id[0x02] enabled) > [ 0.000000] ACPI: LAPIC (acpi_id[0x03] lapic_id[0x03] enabled) > [ 0.000000] ACPI: IOAPIC (id[0x04] address[0xfec00000] gsi_base[0]) > [ 0.000000] IOAPIC[0]: apic_id 4, version 17, address > 0xfec00000, GSI 0-23 > [ 0.000000] ACPI: INT_SRC_OVR (bus 0 bus_irq 0 global_irq 2 dfl > dfl) > [ 0.000000] ACPI: INT_SRC_OVR (bus 0 bus_irq 9 global_irq 9 > high level) > [ 0.000000] ACPI: IRQ0 used by override. > [ 0.000000] ACPI: IRQ2 used by override. > [ 0.000000] ACPI: IRQ9 used by override. > [ 0.000000] Using ACPI (MADT) for SMP configuration information > [ 0.000000] SMP: Allowing 4 CPUs, 0 hotplug CPUs > [ 0.000000] nr_irqs_gsi: 40 > [ 0.000000] PM: Registered nosave memory: 000000000009f000 - > 00000000000a0000 > [ 0.000000] PM: Registered nosave memory: 00000000000a0000 - > 00000000000f0000 > [ 0.000000] PM: Registered nosave memory: 00000000000f0000 - > 0000000000100000 > [ 0.000000] PM: Registered nosave memory: 00000000dfff0000 - > 00000000e0000000 > [ 0.000000] PM: Registered nosave memory: 00000000e0000000 - > 00000000fffc0000 > [ 0.000000] PM: Registered nosave memory: 00000000fffc0000 - > 0000000100000000 > [ 0.000000] Allocating PCI resources starting at e0000000 (gap: > e0000000:1ffc0000) > [ 0.000000] Booting paravirtualized kernel on bare hardware > [ 0.000000] setup_percpu: NR_CPUS:256 nr_cpumask_bits:256 > nr_cpu_ids:4 nr_node_ids:1 > [ 0.000000] PERCPU: Embedded 28 pages/cpu @ffff88011fc00000 > s83072 r8192 d23424 u524288 > [ 0.000000] pcpu-alloc: s83072 r8192 d23424 u524288 alloc=1*2097152 > [ 0.000000] pcpu-alloc: [0] 0 1 2 3 > [ 0.000000] Built 1 zonelists in Node order, mobility grouping > on. Total pages: 1030010 > [ 0.000000] Policy zone: Normal > [ 0.000000] Kernel command line: > BOOT_IMAGE=/boot/vmlinuz-3.2.0-23-generic > root=UUID=ec96128d-feb6-40e6-a5aa-b55aa2219935 ro > [ 0.000000] PID hash table entries: 4096 (order: 3, 32768 bytes) > [ 0.000000] Checking aperture... > [ 0.000000] No AGP bridge found > [ 0.000000] Calgary: detecting Calgary via BIOS EBDA area > [ 0.000000] Calgary: Unable to locate Rio Grande table in EBDA > - bailing! > [ 0.000000] Memory: 4032084k/4718592k available (6566k kernel > code, 524804k absent, 161704k reserved, 6637k data, 920k init) > [ 0.000000] SLUB: Genslabs=15, HWalign=64, Order=0-3, > MinObjects=0, CPUs=4, Nodes=1 > [ 0.000000] Hierarchical RCU implementation. > [ 0.000000] RCU dyntick-idle grace-period acceleration is enabled. > [ 0.000000] NR_IRQS:16640 nr_irqs:712 16 > [ 0.000000] Console: colour dummy device 80x25 > [ 0.000000] console [tty0] enabled > [ 0.000000] allocated 33554432 bytes of page_cgroup > [ 0.000000] please try 'cgroup_disable=memory' option if you > don't want memory cgroups > [ 0.000000] Fast TSC calibration using PIT > [ 0.000000] Detected 3385.153 MHz processor. > [ 0.004003] Calibrating delay loop (skipped), value calculated > using timer frequency.. 6770.30 BogoMIPS (lpj=13540612) > [ 0.004007] pid_max: default: 32768 minimum: 301 > [ 0.004023] Security Framework initialized > [ 0.004033] AppArmor: AppArmor initialized > [ 0.004035] Yama: becoming mindful. > [ 0.004588] Dentry cache hash table entries: 524288 (order: 10, > 4194304 bytes) > [ 0.005520] Inode-cache hash table entries: 262144 (order: 9, > 2097152 bytes) > [ 0.005792] Mount-cache hash table entries: 256 > [ 0.005903] Initializing cgroup subsys cpuacct > [ 0.005908] Initializing cgroup subsys memory > [ 0.005913] Initializing cgroup subsys devices > [ 0.005916] Initializing cgroup subsys freezer > [ 0.005918] Initializing cgroup subsys blkio > [ 0.005922] Initializing cgroup subsys perf_event > [ 0.006124] CPU: Physical Processor ID: 0 > [ 0.006127] CPU: Processor Core ID: 0 > [ 0.006143] mce: CPU supports 0 MCE banks > [ 0.024772] ACPI: Core revision 20110623 > [ 0.025296] ftrace: allocating 27049 entries in 107 pages > [ 0.036357] ..TIMER: vector=0x30 apic1=0 pin1=2 apic2=-1 pin2=-1 > [ 0.076499] CPU0: Intel(R) Core(TM) i7-2600K CPU @ 3.40GHz > stepping 07 > [ 0.080004] Performance Events: unsupported p6 CPU model 42 no > PMU driver, software events only. > [ 0.080004] NMI watchdog disabled (cpu0): hardware events not > enabled > [ 0.080004] Booting Node 0, Processors #1 > [ 0.080004] smpboot cpu 1: start_ip = 9a000 > [ 0.008000] mce: CPU supports 0 MCE banks > [ 0.168009] TSC synchronization [CPU#0 -> CPU#1]: > [ 0.168009] Measured 195227 cycles TSC warp between CPUs, > turning off TSC clock. > [ 0.168009] Marking TSC unstable due to check_tsc_sync_source > failed > [ 0.168030] NMI watchdog disabled (cpu1): hardware events not > enabled > [ 0.168227] #2 > [ 0.168228] smpboot cpu 2: start_ip = 9a000 > [ 0.008000] mce: CPU supports 0 MCE banks > [ 0.256324] NMI watchdog disabled (cpu2): hardware events not > enabled > [ 0.256528] #3 Ok. > [ 0.256529] smpboot cpu 3: start_ip = 9a000 > [ 0.008000] mce: CPU supports 0 MCE banks > [ 0.344331] NMI watchdog disabled (cpu3): hardware events not > enabled > [ 0.344371] Brought up 4 CPUs > [ 0.344373] Total of 4 processors activated (27145.08 BogoMIPS). > [ 0.345354] devtmpfs: initialized > [ 0.345354] EVM: security.selinux > [ 0.345354] EVM: security.SMACK64 > [ 0.345354] EVM: security.capability > [ 0.345354] print_constraints: dummy: > [ 0.345354] RTC time: 10:28:42, date: 04/19/12 > [ 0.345354] NET: Registered protocol family 16 > [ 0.345354] ACPI: bus type pci registered > [ 0.345354] PCI: Using configuration type 1 for base access > [ 0.345354] Trying to unpack rootfs image as initramfs... > [ 0.348353] bio: create slab <bio-0> at 0 > [ 0.348353] ACPI: Added _OSI(Module Device) > [ 0.348353] ACPI: Added _OSI(Processor Device) > [ 0.348353] ACPI: Added _OSI(3.0 _SCP Extensions) > [ 0.348353] ACPI: Added _OSI(Processor Aggregator Device) > [ 0.348353] ACPI: EC: Look up EC in DSDT > [ 0.348547] ACPI: Executed 1 blocks of module-level executable > AML code > [ 0.350447] ACPI: Interpreter enabled > [ 0.350450] ACPI: (supports S0 S5) > [ 0.350459] ACPI: Using IOAPIC for interrupt routing > [ 0.352692] ACPI: No dock devices found. > [ 0.352695] HEST: Table not found. > [ 0.352697] PCI: Ignoring host bridge windows from ACPI; if > necessary, use "pci=use_crs" and report a bug > [ 0.352738] ACPI: PCI Root Bridge [PCI0] (domain 0000 [bus 00-ff]) > [ 0.352813] pci_root PNP0A03:00: host bridge window [io > 0x0000-0x0cf7] (ignored) > [ 0.352815] pci_root PNP0A03:00: host bridge window [io > 0x0d00-0xffff] (ignored) > [ 0.352816] pci_root PNP0A03:00: host bridge window [mem > 0x000a0000-0x000bffff] (ignored) > [ 0.352818] pci_root PNP0A03:00: host bridge window [mem > 0xe0000000-0xffdfffff] (ignored) > [ 0.352852] pci 0000:00:00.0: [8086:1237] type 0 class 0x000600 > [ 0.353150] pci 0000:00:01.0: [8086:7000] type 0 class 0x000601 > [ 0.353567] pci 0000:00:01.1: [8086:7111] type 0 class 0x000101 > [ 0.353885] pci 0000:00:01.1: reg 20: [io 0xd000-0xd00f] > [ 0.354120] pci 0000:00:02.0: [80ee:beef] type 0 class 0x000300 > [ 0.356449] pci 0000:00:02.0: reg 10: [mem > 0xe0000000-0xe0ffffff pref] > [ 0.372789] pci 0000:00:03.0: [1af4:1000] type 0 class 0x000200 > [ 0.372900] pci 0000:00:03.0: reg 10: [io 0xd020-0xd03f] > [ 0.373432] pci 0000:00:04.0: [80ee:cafe] type 0 class 0x000880 > [ 0.376229] pci 0000:00:04.0: reg 10: [io 0xd040-0xd05f] > [ 0.380229] pci 0000:00:04.0: reg 14: [mem 0xf0000000-0xf03fffff] > [ 0.382891] pci 0000:00:04.0: reg 18: [mem > 0xf0400000-0xf0403fff pref] > [ 0.392444] pci 0000:00:06.0: [106b:003f] type 0 class 0x000c03 > [ 0.396234] pci 0000:00:06.0: reg 10: [mem 0xf0404000-0xf0404fff] > [ 0.411054] pci 0000:00:07.0: [8086:7113] type 0 class 0x000680 > [ 0.412109] pci 0000:00:0b.0: [8086:265c] type 0 class 0x000c03 > [ 0.414940] pci 0000:00:0b.0: reg 10: [mem 0xf0405000-0xf0405fff] > [ 0.431155] pci 0000:00:0d.0: [8086:2829] type 0 class 0x000106 > [ 0.432285] pci 0000:00:0d.0: reg 10: [io 0xd060-0xd067] > [ 0.440252] pci 0000:00:0d.0: reg 18: [io 0xd070-0xd077] > [ 0.444431] pci 0000:00:0d.0: reg 20: [io 0xd080-0xd08f] > [ 0.448447] pci 0000:00:0d.0: reg 24: [mem 0xf0406000-0xf0407fff] > [ 0.453161] ACPI: PCI Interrupt Routing Table [\_SB_.PCI0._PRT] > [ 0.454249] pci0000:00: Unable to request _OSC control (_OSC > support mask: 0x1e) > [ 0.460028] ACPI: PCI Interrupt Link [LNKA] (IRQs *5 9 10 11) > [ 0.460448] ACPI: PCI Interrupt Link [LNKB] (IRQs 5 9 10 *11) > [ 0.460566] ACPI: PCI Interrupt Link [LNKC] (IRQs 5 9 *10 11) > [ 0.460681] ACPI: PCI Interrupt Link [LNKD] (IRQs 5 *9 10 11) > [ 0.461127] vgaarb: device added: > PCI:0000:00:02.0,decodes=io+mem,owns=io+mem,locks=none > [ 0.461127] vgaarb: loaded > [ 0.461127] vgaarb: bridge control possible 0000:00:02.0 > [ 0.461127] i2c-core: driver [aat2870] using legacy suspend method > [ 0.461127] i2c-core: driver [aat2870] using legacy resume method > [ 0.461127] SCSI subsystem initialized > [ 0.461127] libata version 3.00 loaded. > [ 0.461127] usbcore: registered new interface driver usbfs > [ 0.461127] usbcore: registered new interface driver hub > [ 0.461127] usbcore: registered new device driver usb > [ 0.461127] PCI: Using ACPI for IRQ routing > [ 0.461127] PCI: pci_cache_line_size set to 64 bytes > [ 0.461127] reserve RAM buffer: 000000000009fc00 - 000000000009ffff > [ 0.461127] reserve RAM buffer: 00000000dfff0000 - 00000000dfffffff > [ 0.461127] NetLabel: Initializing > [ 0.461127] NetLabel: domain hash size = 128 > [ 0.461127] NetLabel: protocols = UNLABELED CIPSOv4 > [ 0.461127] NetLabel: unlabeled traffic allowed by default > [ 0.469329] AppArmor: AppArmor Filesystem Enabled > [ 0.469352] pnp: PnP ACPI init > [ 0.469365] ACPI: bus type pnp registered > [ 0.469424] pnp 00:00: [bus 00-ff] > [ 0.469425] pnp 00:00: [io 0x0cf8-0x0cff] > [ 0.469427] pnp 00:00: [io 0x0000-0x0cf7 window] > [ 0.469428] pnp 00:00: [io 0x0d00-0xffff window] > [ 0.469431] pnp 00:00: [mem 0x000a0000-0x000bffff window] > [ 0.469432] pnp 00:00: [mem 0xe0000000-0xffdfffff window] > [ 0.469454] pnp 00:00: Plug and Play ACPI device, IDs PNP0a03 > (active) > [ 0.469465] pnp 00:01: [io 0x0060] > [ 0.469466] pnp 00:01: [io 0x0064] > [ 0.469492] pnp 00:01: [irq 1] > [ 0.469507] pnp 00:01: Plug and Play ACPI device, IDs PNP0303 > (active) > [ 0.469513] pnp 00:02: [io 0x0000-0x000f] > [ 0.469514] pnp 00:02: [io 0x0080-0x008f] > [ 0.469515] pnp 00:02: [io 0x00c0-0x00df] > [ 0.469516] pnp 00:02: [dma 4] > [ 0.469527] pnp 00:02: Plug and Play ACPI device, IDs PNP0200 > (active) > [ 0.469569] pnp 00:03: [irq 12] > [ 0.469584] pnp 00:03: Plug and Play ACPI device, IDs PNP0f03 > (active) > [ 0.469593] pnp 00:04: [io 0x0378-0x037f] > [ 0.469594] pnp 00:04: [io 0x0778-0x077f] > [ 0.469612] pnp 00:04: [irq 7] > [ 0.469626] pnp 00:04: Plug and Play ACPI device, IDs PNP0400 > (active) > [ 0.469954] pnp: PnP ACPI: found 5 devices > [ 0.469956] ACPI: ACPI bus type pnp unregistered > [ 0.479843] Switching to clocksource acpi_pm > [ 0.479942] PCI: max bus depth: 0 pci_try_num: 1 > [ 0.479960] pci_bus 0000:00: resource 0 [io 0x0000-0xffff] > [ 0.479964] pci_bus 0000:00: resource 1 [mem > 0x00000000-0xfffffffff] > [ 0.479989] NET: Registered protocol family 2 > [ 0.479989] IP route cache hash table entries: 131072 (order: > 8, 1048576 bytes) > [ 0.479989] TCP established hash table entries: 524288 (order: > 11, 8388608 bytes) > [ 0.479989] TCP bind hash table entries: 65536 (order: 8, > 1048576 bytes) > [ 0.479989] TCP: Hash tables configured (established 524288 > bind 65536) > [ 0.479989] TCP reno registered > [ 0.479989] UDP hash table entries: 2048 (order: 4, 65536 bytes) > [ 0.479989] UDP-Lite hash table entries: 2048 (order: 4, 65536 > bytes) > [ 0.479989] NET: Registered protocol family 1 > [ 0.479989] pci 0000:00:00.0: Limiting direct PCI/PCI transfers > [ 0.479989] pci 0000:00:01.0: Activating ISA DMA hang workarounds > [ 0.479989] pci 0000:00:02.0: Boot video device > [ 0.479989] pci 0000:00:06.0: PCI INT A -> GSI 22 (level, low) > -> IRQ 22 > [ 0.536224] pci 0000:00:06.0: PCI INT A disabled > [ 0.536281] pci 0000:00:0b.0: PCI INT A -> GSI 19 (level, low) > -> IRQ 19 > [ 0.536419] pci 0000:00:0b.0: PCI INT A disabled > [ 0.536438] PCI: CLS 64 bytes, default 64 > [ 0.536440] PCI-DMA: Using software bounce buffering for IO > (SWIOTLB) > [ 0.536442] Placing 64MB software IO TLB between > ffff8800dbfea000 - ffff8800dffea000 > [ 0.536445] software IO TLB at phys 0xdbfea000 - 0xdffea000 > [ 0.536534] platform rtc_cmos: registered platform RTC device > (no PNP device found) > [ 0.537218] audit: initializing netlink socket (disabled) > [ 0.537228] type=2000 audit(1334831321.536:1): initialized > [ 0.559691] HugeTLB registered 2 MB page size, pre-allocated 0 > pages > [ 0.569424] VFS: Disk quotas dquot_6.5.2 > [ 0.569466] Dquot-cache hash table entries: 512 (order 0, 4096 > bytes) > [ 0.570011] fuse init (API version 7.17) > [ 0.570131] msgmni has been set to 7875 > [ 0.571849] Block layer SCSI generic (bsg) driver version 0.4 > loaded (major 253) > [ 0.571976] io scheduler noop registered > [ 0.571979] io scheduler deadline registered > [ 0.572009] io scheduler cfq registered (default) > [ 0.572158] pci_hotplug: PCI Hot Plug PCI Core version: 0.5 > [ 0.572176] pciehp: PCI Express Hot Plug Controller Driver > version: 0.4 > [ 0.572258] ACPI: Deprecated procfs I/F for AC is loaded, > please retry with CONFIG_ACPI_PROCFS_POWER cleared > [ 0.572298] ACPI: AC Adapter [AC] (on-line) > [ 0.572375] input: Power Button as > /devices/LNXSYSTM:00/LNXPWRBN:00/input/input0 > [ 0.572380] ACPI: Power Button [PWRF] > [ 0.572433] input: Sleep Button as > /devices/LNXSYSTM:00/LNXSLPBN:00/input/input1 > [ 0.572437] ACPI: Sleep Button [SLPF] > [ 0.573452] ERST: Table is not found! > [ 0.573454] GHES: HEST is not enabled! > [ 0.573530] virtio-pci 0000:00:03.0: PCI INT A -> GSI 19 > (level, low) -> IRQ 19 > [ 0.573727] Serial: 8250/16550 driver, 32 ports, IRQ sharing > enabled > [ 0.605871] Freeing initrd memory: 13832k freed > [ 1.248951] Linux agpgart interface v0.103 > [ 1.254544] brd: module loaded > [ 1.256947] loop: module loaded > [ 1.257314] ahci 0000:00:0d.0: version 3.0 > [ 1.257415] ahci 0000:00:0d.0: PCI INT A -> GSI 21 (level, low) > -> IRQ 21 > [ 1.257633] ahci: SSS flag set, parallel bus scan disabled > [ 1.257942] ahci 0000:00:0d.0: AHCI 0001.0100 32 slots 1 ports > 3 Gbps 0x1 impl SATA mode > [ 1.257951] ahci 0000:00:0d.0: flags: 64bit ncq stag only ccc > [ 1.258674] scsi0 : ahci > [ 1.258767] ata1: SATA max UDMA/133 abar m8192@0xf0406000 port > 0xf0406100 irq 21 > [ 1.258844] ata_piix 0000:00:01.1: version 2.13 > [ 1.259824] scsi1 : ata_piix > [ 1.260317] scsi2 : ata_piix > [ 1.260374] ata2: PATA max UDMA/33 cmd 0x1f0 ctl 0x3f6 bmdma > 0xd000 irq 14 > [ 1.260380] ata3: PATA max UDMA/33 cmd 0x170 ctl 0x376 bmdma > 0xd008 irq 15 > [ 1.260804] Fixed MDIO Bus: probed > [ 1.260817] tun: Universal TUN/TAP device driver, 1.6 > [ 1.260818] tun: (C) 1999-2004 Max Krasnyansky > <maxk(a)qualcomm.com <mailto:maxk@qualcomm.com>> > [ 1.261638] PPP generic driver version 2.4.2 > [ 1.262079] ehci_hcd: USB 2.0 'Enhanced' Host Controller (EHCI) > Driver > [ 1.262122] ehci_hcd 0000:00:0b.0: PCI INT A -> GSI 19 (level, > low) -> IRQ 19 > [ 1.262195] ehci_hcd 0000:00:0b.0: EHCI Host Controller > [ 1.262430] ehci_hcd 0000:00:0b.0: new USB bus registered, > assigned bus number 1 > [ 1.262761] ehci_hcd 0000:00:0b.0: irq 19, io mem 0xf0405000 > [ 1.272188] ehci_hcd 0000:00:0b.0: USB 2.0 started, EHCI 1.00 > [ 1.272644] hub 1-0:1.0: USB hub found > [ 1.272654] hub 1-0:1.0: 8 ports detected > [ 1.272827] ohci_hcd: USB 1.1 'Open' Host Controller (OHCI) Driver > [ 1.272898] ohci_hcd 0000:00:06.0: PCI INT A -> GSI 22 (level, > low) -> IRQ 22 > [ 1.272984] ohci_hcd 0000:00:06.0: OHCI Host Controller > [ 1.273269] ohci_hcd 0000:00:06.0: new USB bus registered, > assigned bus number 2 > [ 1.273417] ohci_hcd 0000:00:06.0: irq 22, io mem 0xf0404000 > [ 1.329002] hub 2-0:1.0: USB hub found > [ 1.329032] hub 2-0:1.0: 8 ports detected > [ 1.329525] uhci_hcd: USB Universal Host Controller Interface > driver > [ 1.329601] usbcore: registered new interface driver libusual > [ 1.329655] i8042: PNP: PS/2 Controller > [PNP0303:PS2K,PNP0f03:PS2M] at 0x60,0x64 irq 1,12 > [ 1.330733] serio: i8042 KBD port at 0x60,0x64 irq 1 > [ 1.330748] serio: i8042 AUX port at 0x60,0x64 irq 12 > [ 1.331290] mousedev: PS/2 mouse device common for all mice > [ 1.332223] input: AT Translated Set 2 keyboard as > /devices/platform/i8042/serio0/input/input2 > [ 1.332913] rtc_cmos rtc_cmos: rtc core: registered rtc_cmos as > rtc0 > [ 1.333075] rtc0: alarms up to one day, 114 bytes nvram > [ 1.333222] device-mapper: uevent: version 1.0.3 > [ 1.333528] device-mapper: ioctl: 4.22.0-ioctl (2011-10-19) > initialised: dm-devel(a)redhat.com <mailto:dm-devel@redhat.com> > [ 1.333559] cpuidle: using governor ladder > [ 1.333563] cpuidle: using governor menu > [ 1.333567] EFI Variables Facility v0.08 2004-May-17 > [ 1.333859] TCP cubic registered > [ 1.334011] NET: Registered protocol family 10 > [ 1.334802] NET: Registered protocol family 17 > [ 1.334835] Registering the dns_resolver key type > [ 1.335954] PM: Hibernation image not present or could not be > loaded. > [ 1.335973] registered taskstats version 1 > [ 1.360349] Magic number: 8:400:477 > [ 1.360600] rtc_cmos rtc_cmos: setting system clock to > 2012-04-19 10:28:43 UTC (1334831323) > [ 1.360808] BIOS EDD facility v0.16 2004-Jun-25, 0 devices found > [ 1.360813] EDD information not available. > [ 1.417071] ata3.00: ATAPI: VBOX CD-ROM, 1.0, max UDMA/133 > [ 1.418236] ata3.00: configured for UDMA/33 > [ 1.576294] ata1: SATA link up 3.0 Gbps (SStatus 123 SControl 300) > [ 1.576442] ata1.00: ATA-6: VBOX HARDDISK, 1.0, max UDMA/133 > [ 1.576449] ata1.00: 16777216 sectors, multi 128: LBA48 NCQ > (depth 31/32) > [ 1.576637] ata1.00: configured for UDMA/133 > [ 1.577145] scsi 0:0:0:0: Direct-Access ATA VBOX > HARDDISK 1.0 PQ: 0 ANSI: 5 > [ 1.577596] sd 0:0:0:0: Attached scsi generic sg0 type 0 > [ 1.577628] sd 0:0:0:0: [sda] 16777216 512-byte logical blocks: > (8.58 GB/8.00 GiB) > [ 1.578323] sd 0:0:0:0: [sda] Write Protect is off > [ 1.578332] sd 0:0:0:0: [sda] Mode Sense: 00 3a 00 00 > [ 1.578580] sd 0:0:0:0: [sda] Write cache: enabled, read cache: > enabled, doesn't support DPO or FUA > [ 1.578765] scsi 2:0:0:0: CD-ROM VBOX CD-ROM > 1.0 PQ: 0 ANSI: 5 > [ 1.579836] sr0: scsi3-mmc drive: 32x/32x xa/form2 tray > [ 1.579853] cdrom: Uniform CD-ROM driver Revision: 3.20 > [ 1.580258] sr 2:0:0:0: Attached scsi CD-ROM sr0 > [ 1.580557] sr 2:0:0:0: Attached scsi generic sg1 type 5 > [ 1.581366] sda: sda1 sda2 < sda5 > > [ 1.583094] sd 0:0:0:0: [sda] Attached SCSI disk > [ 1.584231] usb 1-1: new high-speed USB device number 2 using > ehci_hcd > [ 1.585305] Freeing unused kernel memory: 920k freed > [ 1.585472] Write protecting the kernel read-only data: 12288k > [ 1.593051] Freeing unused kernel memory: 1608k freed > [ 1.599031] Freeing unused kernel memory: 1196k freed > [ 1.637534] udevd[101]: starting version 175 > [ 1.771768] EXT4-fs (sda1): mounted filesystem with ordered > data mode. Opts: (null) > [ 1.874977] hrtimer: interrupt took 107977688 ns > [ 2.224241] usb 2-1: new full-speed USB device number 2 using > ohci_hcd > [ 4.998829] ADDRCONF(NETDEV_UP): eth0: link is not ready > [ 5.015680] udevd[296]: starting version 175 > [ 5.046608] Adding 4192252k swap on /dev/sda5. Priority:-1 > extents:1 across:4192252k > [ 5.084794] piix4_smbus 0000:00:07.0: SMBus base address > uninitialized - upgrade BIOS or use force_addr=0xaddr > [ 5.090614] lp: driver loaded but no devices found > [ 5.096851] type=1400 audit(1334831327.232:2): > apparmor="STATUS" operation="profile_load" name="/sbin/dhclient" > pid=359 comm="apparmor_parser" > [ 5.097421] type=1400 audit(1334831327.232:3): > apparmor="STATUS" operation="profile_load" > name="/usr/lib/NetworkManager/nm-dhcp-client.action" pid=359 > comm="apparmor_parser" > [ 5.097656] type=1400 audit(1334831327.232:4): > apparmor="STATUS" operation="profile_load" > name="/usr/lib/connman/scripts/dhclient-script" pid=359 > comm="apparmor_parser" > [ 5.142805] pci 0000:00:04.0: PCI INT A -> GSI 20 (level, low) > -> IRQ 20 > [ 5.148416] input: M-Audio Fast Track C400 as > /devices/pci0000:00/0000:00:0b.0/usb1/1-1/1-1:1.5/input/input3 > [ 5.150294] input: Unspecified device as > /devices/virtual/input/input4 > [ 5.151404] vboxguest: major 0, IRQ 20, I/O port d040, MMIO at > 00000000f0000000 (size 0x400000) > [ 5.151407] vboxguest: Successfully loaded version 4.1.12 > (interface 0x00010004) > [ 5.153200] generic-usb 0003:0763:2030.0001: input,hidraw0: USB > HID v1.10 Keyboard [M-Audio Fast Track C400] on > usb-0000:00:0b.0-1/input5 > [ 5.154104] input: ImExPS/2 Generic Explorer Mouse as > /devices/platform/i8042/serio1/input/input5 > [ 5.173055] parport_pc 00:04: reported by Plug and Play ACPI > [ 5.176070] input: VirtualBox USB Tablet as > /devices/pci0000:00/0000:00:06.0/usb2/2-1/2-1:1.0/input/input6 > [ 5.176363] generic-usb 0003:80EE:0021.0002: input,hidraw1: USB > HID v1.10 Mouse [VirtualBox USB Tablet] on usb-0000:00:06.0-1/input0 > [ 5.176376] usbcore: registered new interface driver usbhid > [ 5.176377] usbhid: USB HID core driver > [ 5.193484] EXT4-fs (sda1): re-mounted. Opts: errors=remount-ro > [ 5.197201] ppdev: user-space parallel port driver > [ 5.284929] snd-usb-audio: probe of 1-1:1.1 failed with error -5 > [ 5.290898] snd-usb-audio: probe of 1-1:1.2 failed with error -5 > [ 5.296928] snd-usb-audio: probe of 1-1:1.3 failed with error -5 > [ 5.303749] usbcore: registered new interface driver snd-usb-audio > [ 5.568497] vesafb: mode is 640x480x32, linelength=2560, pages=0 > [ 5.568501] vesafb: scrolling: redraw > [ 5.568505] vesafb: Truecolor: size=8:8:8:8, shift=24:16:8:0 > [ 5.568573] mtrr: your processor doesn't support write-combining > [ 5.568906] vesafb: framebuffer at 0xe0000000, mapped to > 0xffffc90003f00000, using 1216k, total 1216k > [ 5.569237] Console: switching to colour frame buffer device 80x30 > [ 5.573612] fb0: VESA VGA frame buffer device > [ 7.628873] init: failsafe main process (676) killed by TERM signal > [ 7.686842] type=1400 audit(1334831329.820:5): > apparmor="STATUS" operation="profile_load" > name="/usr/sbin/tcpdump" pid=856 comm="apparmor_parser" > [ 7.686955] type=1400 audit(1334831329.820:6): > apparmor="STATUS" operation="profile_replace" > name="/sbin/dhclient" pid=854 comm="apparmor_parser" > [ 7.687523] type=1400 audit(1334831329.820:7): > apparmor="STATUS" operation="profile_replace" > name="/usr/lib/NetworkManager/nm-dhcp-client.action" pid=854 > comm="apparmor_parser" > [ 7.687836] type=1400 audit(1334831329.820:8): > apparmor="STATUS" operation="profile_replace" > name="/usr/lib/connman/scripts/dhclient-script" pid=854 > comm="apparmor_parser" > [ 7.881629] vboxsf: Successfully loaded version 4.1.12 > (interface 0x00010004) > [ 15.552201] eth0: no IPv6 routers present > > > > On 4/19/2012 7:09 AM, Grant Diffey wrote: > > On Thu, Apr 19, 2012 at 8:26 PM, Chris > Cavey<chris-alsa(a)rauros.net <mailto:chris-alsa@rauros.net>> > wrote: > > Complete snippet from the log > > [ 374.104138] usbcore: deregistering interface driver > snd-usb-audio > [ 374.875660] snd-usb-audio: probe of 1-1:1.1 failed with > error -5 > [ 374.881649] snd-usb-audio: probe of 1-1:1.2 failed with > error -5 > [ 374.887654] snd-usb-audio: probe of 1-1:1.3 failed with > error -5 > [ 374.895366] usbcore: registered new interface driver > snd-usb-audio > > Sidenote - While I don't have a midi device to test with, > it would > appear that the midi portion registers correctly and is > accessible (at > least through amidi and aplaymidi) > > On 04/19/2012 04:53 AM, Clemens Ladisch wrote: > > Chris Cavey wrote: > > snd-usb-audio: probe of 1-1:1.1 failed with error -5 > snd-usb-audio: probe of 1-1:1.2 failed with error -5 > snd-usb-audio: probe of 1-1:1.3 failed with error -5 > > Are there no other messages immediately before this? > > > that looks like a snippet on unload can I have another 10 > lines either side? > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org <mailto:Alsa-devel@alsa-project.org> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org <mailto:Alsa-devel@alsa-project.org> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

1 0

Re: [alsa-devel] [english 100%] Re: Steinberg CMC
by Clemens Ladisch 19 Apr '12

19 Apr '12

Patrick wrote: > Probably 1062 1063 1064 are the CMC- FD PD and QC but in which order? The important information is the ID. The name can be read from the device's descriptors; this parameter is there just to allow overriding it. > I understand the patch is trivial it is three lines to add. But I need to > get the development version to submit it. Otherwise if in the meantime the > 105f and the 1066 are entered, my patch will return an error. > > ... or can the maintainers do it? Such conflicts would be trivial to resolve. Regards, Clemens

1 0

[alsa-devel] Steinberg CMC
by Patrick 19 Apr '12

19 Apr '12

Hi I own 3 of the 6 Steinberg control surfaces CMC series To run them, I added some entries in the (usb) quirk-table. They are midi input and output of vendor ID Yahama and products as the following CMC-CH 0x1060 CMC-TP 0x1061 CMC-AI 0x1065 Where to notify that or how to add a patch? For the patch, should I paste the lines in the last Linux Kernel version and create the patch? Cheers Patrick

2 1

[alsa-devel] [PATCH] sound/isa/sc6000.c: add missing sc6000_setup_board
by Julia Lawall 19 Apr '12

19 Apr '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> There seems to be no interaction between *port and the requested region, so the clearing of *vport should take place if the request_region fails. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- Not tested. sc6000_setup_board seems rather complicated, so my reasoning could be completely wrong. sound/isa/sc6000.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index d97d0f3..57679cb 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -594,7 +594,7 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) snd_printk(KERN_ERR PFX "I/O port cannot be iomaped.\n"); err = -EBUSY; - goto err_unmap1; + goto err_unmap0; } /* to make it marked as used */ @@ -678,9 +678,10 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) return 0; err_unmap2: - sc6000_setup_board(*vport, 0); release_region(mss_port[dev], 4); err_unmap1: + sc6000_setup_board(*vport, 0); +err_unmap0: release_region(port[dev], 0x10); err_exit: snd_card_free(card);

2 3

[alsa-devel] [PATCH 0/5] ASoC: sh: fsi-codec
by Kuninori Morimoto 19 Apr '12

19 Apr '12

Hi Mark, Liam Cc: Paul, Rafael These patches adds fsi-codec support for SuperH board. Kuninori Morimoto (5): ASoC: fsi: add fsi-codec card support ASoC: fsi: remove fsi-ak4642 card driver ASoC: fsi: remove fsi-hdmi card driver ASoC: fsi: remove fsi-da7210 card driver ASoC: fsi: add FSI-WM8978 support Current FSI-xxx cards are using fsi-xxx.c driver. But these were very similar. To reduce useless code, #1 adds common card driver as "fsi-codec", and #2-#4 switch to use it. And #5 adds new card support Please note that these patches are based on latest linus/master. I guess current mark/for-3.4 will get conflict Best regards --- Kuninori Morimoto

4 41

[alsa-devel] ALSA: Updating Sample Rates in sound/core/pcm_native.c
by Jeremy Torres 19 Apr '12

19 Apr '12

Hello, I have updated pcm.h in the linux kernel to allow for new sample rates. For example, I have added 128000 and 144000 sample rates. Per the documentation in pcm.h, I also updated pcm_native.c (the rates table). However when I try and use the "default" device for the sample rates: 1. For 128000: plug conversion occurs for 128K, but the slave is configured to 96K; 2. For 144000: plug conversion is attempted at the 176400 rate; Note that: a) I have snd_pcm_hw_resample set to FALSE in my application; however, I've been using aplay/arecord to test as well with repeatable results. b) I do not have an asoundrc setup for plug conversion, etc. I only have the default /usr/share/alsa/alsa.conf; c) When I try the the device "hw:0,0", I get a (proper) warning that the sample rate requested is not being honored (i.e., 96K instead of 128K). I have placed debug in pcm_lib.c, and pcm_native.c and my sample rate constraints are set. My understanding is if I define a specific rate and it can be matched in the "rates" table, it should be honored. Environment Info: ALSA lib/driver version 1.0.20 SOC Driver Any help would be greatly appreciated! Regards, Jeremy

3 2

Re: [alsa-devel] No sound on Quanta KN1 with kernel 3.4
by Takashi Iwai 19 Apr '12

19 Apr '12

At Sat, 14 Apr 2012 22:52:09 +0200, Uros Vampl wrote: > > On 14.04.12 21:25, Takashi Iwai wrote: > > At Sat, 14 Apr 2012 19:10:30 +0200, > > Uros Vampl wrote: > > > > > > Hi, > > > > > > as I wrote on the -users list, I don't have sound anymore in kernel 3.4. > > > I was told to post the output of alsa-info.sh here, so here I am. The > > > requested output is attached. > > > > Could you give the alsa-info.sh output on the working kernel? > > In that way, it becomes easier to spot out what caused the problem. > > Attached. Could you try the patch below? Also, which output is missing? Is it a speaker output or a line-out jack? Basically it's a bug of BIOS that doesn't give the proper association number for the corresponding pin 0x0f. The patch is just to make less restrictive. But, if it's a laptop, the corresponding pin is likely a speaker output, so it'd be better to fix the pin-default value itself via pinfix table. thanks, Takashi --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7a8fcc4..83f0b13 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4957,8 +4957,6 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, if (!(wid_caps & AC_WCAP_STEREO)) if (!cfg->mono_out_pin) cfg->mono_out_pin = nid; - if (!assoc) - continue; if (!assoc_line_out) assoc_line_out = assoc; else if (assoc_line_out != assoc)

2 16

[alsa-devel] [PATCH v2] ASoC: dapm - Add API call to query valid DAPM paths.
by Liam Girdwood 18 Apr '12

18 Apr '12

In preparation for ASoC DSP support. Add a DAPM API call to determine whether a DAPM audio path is valid between source and sink widgets. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the DSP core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- Changes since V1:- Now uses new DAPM mutex. Tracepoint used for path debug. include/sound/soc-dapm.h | 5 ++ include/trace/events/asoc.h | 80 ++++++++++++++++++++++++++++ sound/soc/soc-dapm.c | 122 +++++++++++++++++++++++++++++++++++++++---- 3 files changed, 197 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index bea0c86..e3833d9 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -321,6 +321,7 @@ struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; struct regulator; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -403,6 +404,10 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); +/* dapm path query */ +int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, + struct snd_soc_dapm_widget_list **list); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index ab26f8a..6d8efb1 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -7,6 +7,8 @@ #include <linux/ktime.h> #include <linux/tracepoint.h> +#define DAPM_DIRECT "(direct)" + struct snd_soc_jack; struct snd_soc_codec; struct snd_soc_platform; @@ -241,6 +243,84 @@ TRACE_EVENT(snd_soc_dapm_walk_done, (int)__entry->path_checks, (int)__entry->neighbour_checks) ); +TRACE_EVENT(snd_soc_dapm_output_path, + + TP_PROTO(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_path *path), + + TP_ARGS(widget, path), + + TP_STRUCT__entry( + __string( wname, widget->name ) + __string( pname, path->name ? path->name : DAPM_DIRECT) + __string( psname, path->sink->name ) + __field( int, path_sink ) + __field( int, path_connect ) + ), + + TP_fast_assign( + __assign_str(wname, widget->name); + __assign_str(pname, path->name ? path->name : DAPM_DIRECT); + __assign_str(psname, path->sink->name); + __entry->path_connect = path->connect; + __entry->path_sink = (int)path->sink; + ), + + TP_printk("%c%s -> %s -> %s\n", + (int) __entry->path_sink && + (int) __entry->path_connect ? '*' : ' ', + __get_str(wname), __get_str(pname), __get_str(psname)) +); + +TRACE_EVENT(snd_soc_dapm_input_path, + + TP_PROTO(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_path *path), + + TP_ARGS(widget, path), + + TP_STRUCT__entry( + __string( wname, widget->name ) + __string( pname, path->name ? path->name : DAPM_DIRECT) + __string( psname, path->source->name ) + __field( int, path_source ) + __field( int, path_connect ) + ), + + TP_fast_assign( + __assign_str(wname, widget->name); + __assign_str(pname, path->name ? path->name : DAPM_DIRECT); + __assign_str(psname, path->source->name); + __entry->path_connect = path->connect; + __entry->path_source = (int)path->source; + ), + + TP_printk("%c%s <- %s <- %s\n", + (int) __entry->path_source && + (int) __entry->path_connect ? '*' : ' ', + __get_str(wname), __get_str(pname), __get_str(psname)) +); + +TRACE_EVENT(snd_soc_dapm_connected, + + TP_PROTO(int paths, int stream), + + TP_ARGS(paths, stream), + + TP_STRUCT__entry( + __field( int, paths ) + __field( int, stream ) + ), + + TP_fast_assign( + __entry->paths = paths; + __entry->stream = stream; + ), + + TP_printk("%s: found %d paths\n", + __entry->stream ? "capture" : "playback", __entry->paths) +); + TRACE_EVENT(snd_soc_jack_irq, TP_PROTO(const char *name), diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d7ee73a..214323f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -705,11 +705,51 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +/* add widget to list if it's not already in the list */ +static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + if (*list == NULL) + return -EINVAL; + + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + /* allocate some new space */ + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(w->dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist = *list; + + /* insert the widget */ + dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. */ -static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -765,9 +805,23 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) if (path->walked) continue; + trace_snd_soc_dapm_output_path(widget, path); + if (path->sink && path->connect) { path->walked = 1; - con += is_connected_output_ep(path->sink); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_output_ep(path->sink, list); } } @@ -780,7 +834,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. */ -static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -848,9 +903,23 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) if (path->walked) continue; + trace_snd_soc_dapm_input_path(widget, path); + if (path->source && path->connect) { path->walked = 1; - con += is_connected_input_ep(path->source); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_input_ep(path->source, list); } } @@ -859,6 +928,39 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return con; } +/** + * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * @dai: the soc DAI. + * @stream: stream direction. + * @list: list of active widgets for this stream. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial stream specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, + struct snd_soc_dapm_widget_list **list) +{ + struct snd_soc_card *card = dai->card; + int paths; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + dapm_reset(card); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = is_connected_output_ep(dai->playback_widget, list); + else + paths = is_connected_input_ep(dai->playback_widget, list); + + trace_snd_soc_dapm_connected(paths, stream); + dapm_clear_walk(&card->dapm); + mutex_unlock(&card->dapm_mutex); + + return paths; +} + /* * Handler for generic register modifier widget. */ @@ -915,9 +1017,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0 && in != 0; } @@ -940,7 +1042,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); if (w->active) { - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); return in != 0; } else { @@ -956,7 +1058,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); if (w->active) { - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0; } else { @@ -1558,9 +1660,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: samsung: Hook up AIF2 to the CODEC on Littlemill
by Mark Brown 18 Apr '12

18 Apr '12

Connect the WM1250-EV1 baseband simulator on Littlemill systems up to the CODEC AIF2 using the new CODEC<->CODEC link support, allowing a wider range of use cases to be represented. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/littlemill.c | 82 ++++++++++++++++++++++++++++++++++------ 1 file changed, 70 insertions(+), 12 deletions(-) diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index e741685..71bff63 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -23,10 +23,10 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; int ret; - if (dapm->dev != codec_dai->dev) + if (dapm->dev != aif1_dai->dev) return 0; switch (level) { @@ -36,7 +36,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, * then do so now, otherwise these are noops. */ if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK2, 32768, sample_rate * 512); if (ret < 0) { @@ -44,7 +44,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, return ret; } - ret = snd_soc_dai_set_sysclk(codec_dai, + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_FLL1, sample_rate * 512, SND_SOC_CLOCK_IN); @@ -66,25 +66,58 @@ static int littlemill_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; int ret; - if (dapm->dev != codec_dai->dev) + if (dapm->dev != aif1_dai->dev) return 0; switch (level) { case SND_SOC_BIAS_STANDBY: - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) { - pr_err("Failed to switch away from FLL: %d\n", ret); + pr_err("Failed to switch away from FLL2: %d\n", ret); return ret; } - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2, 0, 0, 0); if (ret < 0) { - pr_err("Failed to stop FLL: %d\n", ret); + pr_err("Failed to stop FLL2: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL1: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL1: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_PREPARE: + ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2, + WM8994_FLL_SRC_BCLK, 64 * 8000, + 8000 * 256); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_FLL2, + 8000 * 256, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); return ret; } break; @@ -131,6 +164,14 @@ static struct snd_soc_ops littlemill_ops = { .hw_params = littlemill_hw_params, }; +static const struct snd_soc_pcm_stream baseband_params = { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, + .channels_max = 2, +}; + static struct snd_soc_dai_link littlemill_dai[] = { { .name = "CPU", @@ -143,6 +184,17 @@ static struct snd_soc_dai_link littlemill_dai[] = { | SND_SOC_DAIFMT_CBM_CFM, .ops = &littlemill_ops, }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm8994-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &baseband_params, + }, }; static struct snd_soc_dapm_widget widgets[] = { @@ -169,10 +221,16 @@ static struct snd_soc_jack littlemill_headset; static int littlemill_late_probe(struct snd_soc_card *card) { struct snd_soc_codec *codec = card->rtd[0].codec; - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) return ret; -- 1.7.10

1 0

[alsa-devel] [PATCH] configure.in: look for ncurses pkg-config first
by Mike Frysinger 18 Apr '12

18 Apr '12

Newer ncurses installs pkg-config files, and these are much easier to manage when cross-compiling, so default to searching for those. If they can't be found, we fall back to the existing xxx-config. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- configure.in | 64 +++++++++++++++++++++++++++++++++++++-------------------- 1 files changed, 41 insertions(+), 23 deletions(-) diff --git a/configure.in b/configure.in index 2208a95..9404df0 100644 --- a/configure.in +++ b/configure.in @@ -29,6 +29,7 @@ AC_PROG_INSTALL AC_PROG_MKDIR_P AC_PROG_LN_S AC_PROG_SED +PKG_PROG_PKG_CONFIG AM_PATH_ALSA(1.0.24) AC_CHECK_FUNC([snd_ctl_elem_add_enumerated], , [AC_ERROR([No user enum control support in alsa-lib])]) @@ -130,34 +131,52 @@ if test x$alsamixer = xtrue; then NCURSESLIBSUFFIX="" CURSES_NLS="no" if test "$curseslib" = "ncursesw" -o $ "$curseslib" = "auto" -a "$USE_NLS" = "yes" $; then - AC_CHECK_PROG([ncursesw5_config], [ncursesw5-config], [yes]) - if test "$ncursesw5_config" = "yes"; then - CURSESINC="<ncurses.h>" - CURSESLIB=`ncursesw5-config --libs` - CURSESLIBDIR=`ncursesw5-config --libdir` - CURSES_CFLAGS=`ncursesw5-config --cflags` - curseslib="ncursesw" - else - AC_CHECK_LIB(ncursesw, initscr, - [ CURSESINC='<ncurses.h>'; CURSESLIB='-lncursesw'; curseslib="ncursesw"]) - fi + dnl First try out pkg-config, then fall back to old config scripts. + PKG_CHECK_MODULES([NCURSESW], [ncursesw], [ + CURSESINC="<ncurses.h>" + CURSESLIB="${NCURSESW_LIBS}" + CURSESLIBDIR= + CURSES_CFLAGS="${NCURSESW_CFLAGS}" + curseslib="ncursesw" + ], [ + AC_CHECK_PROG([ncursesw5_config], [ncursesw5-config], [yes]) + if test "$ncursesw5_config" = "yes"; then + CURSESINC="<ncurses.h>" + CURSESLIB=`ncursesw5-config --libs` + CURSESLIBDIR=`ncursesw5-config --libdir` + CURSES_CFLAGS=`ncursesw5-config --cflags` + curseslib="ncursesw" + else + AC_CHECK_LIB(ncursesw, initscr, + [ CURSESINC='<ncurses.h>'; CURSESLIB='-lncursesw'; curseslib="ncursesw"]) + fi + ]) if test -n "$CURSESINC"; then NCURSESLIBSUFFIX="w" CURSES_NLS="yes" fi fi if test "$curseslib" = "ncurses" -o "$curseslib" = "auto"; then - AC_CHECK_PROG([ncurses5_config], [ncurses5-config], [yes]) - if test "$ncurses5_config" = "yes"; then - CURSESINC="<ncurses.h>" - CURSESLIB=`ncurses5-config --libs` - CURSESLIBDIR=`ncurses5-config --libdir` - CURSES_CFLAGS=`ncurses5-config --cflags` - curseslib="ncurses" - else - AC_CHECK_LIB(ncurses, initscr, - [ CURSESINC='<ncurses.h>'; CURSESLIB='-lncurses'; curseslib="ncurses"]) - fi + dnl First try out pkg-config, then fall back to old config scripts. + PKG_CHECK_MODULES([NCURSES], [ncurses], [ + CURSESINC="<ncurses.h>" + CURSESLIB="${NCURSES_LIBS}" + CURSESLIBDIR= + CURSES_CFLAGS="${NCURSES_CFLAGS}" + curseslib="ncurses" + ], [ + AC_CHECK_PROG([ncurses5_config], [ncurses5-config], [yes]) + if test "$ncurses5_config" = "yes"; then + CURSESINC="<ncurses.h>" + CURSESLIB=`ncurses5-config --libs` + CURSESLIBDIR=`ncurses5-config --libdir` + CURSES_CFLAGS=`ncurses5-config --cflags` + curseslib="ncurses" + else + AC_CHECK_LIB(ncurses, initscr, + [ CURSESINC='<ncurses.h>'; CURSESLIB='-lncurses'; curseslib="ncurses"]) + fi + ]) fi if test "$curseslib" = "curses" -o "$curseslib" = "auto"; then AC_CHECK_LIB(curses, initscr, @@ -287,7 +306,6 @@ SAVE_UTIL_VERSION AC_SUBST(LIBRT) dnl Check for systemd -PKG_PROG_PKG_CONFIG AC_ARG_WITH([systemdsystemunitdir], AS_HELP_STRING([--with-systemdsystemunitdir=DIR], [Directory for systemd service files]), [], [with_systemdsystemunitdir=$($PKG_CONFIG --variable=systemdsystemunitdir systemd)]) -- 1.7.8.5

2 2

[alsa-devel] [PATCH] configure.in: use AS_HELP_STRING everywhere
by Mike Frysinger 18 Apr '12

18 Apr '12

This macro takes care of spacing for us. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- configure.in | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/configure.in b/configure.in index 9404df0..e7bb4ca 100644 --- a/configure.in +++ b/configure.in @@ -76,7 +76,7 @@ CURSES_CFLAGS="" alsamixer= if test "$have_mixer" = "yes"; then AC_ARG_ENABLE(alsamixer, - [ --disable-alsamixer Disable alsamixer compilation], + AS_HELP_STRING([--disable-alsamixer], [Disable alsamixer compilation]), [case "${enableval}" in yes) alsamixer=true ;; no) alsamixer=false ;; @@ -87,7 +87,7 @@ AM_CONDITIONAL(ALSAMIXER, test x$alsamixer = xtrue) dnl Disable alsaconf AC_ARG_ENABLE(alsaconf, - [ --disable-alsaconf Disable alsaconf packaging], + AS_HELP_STRING([--disable-alsaconf], [Disable alsaconf packaging]), [case "${enableval}" in yes) alsaconf=true ;; no) alsaconf=false ;; @@ -97,7 +97,7 @@ AM_CONDITIONAL(ALSACONF, test x$alsaconf = xtrue) dnl Disable alsaloop AC_ARG_ENABLE(alsaloop, - [ --disable-alsaloop Disable alsaloop packaging], + AS_HELP_STRING([--disable-alsaloop], [Disable alsaloop packaging]), [case "${enableval}" in yes) alsaloop=true ;; no) alsaloop=false ;; @@ -124,7 +124,7 @@ dnl Checks for header files. AC_HEADER_STDC if test x$alsamixer = xtrue; then AC_ARG_WITH(curses, - [ --with-curses libname Specify the curses library to use (default=auto)], + AS_HELP_STRING([--with-curses=libname], [Specify the curses library to use (default=auto)]), curseslib="$withval", curseslib="auto") CURSESLIBDIR="" @@ -283,7 +283,7 @@ mydatadir="$dir/alsa" AC_DEFINE_UNQUOTED(DATADIR, "$mydatadir", [directory containing alsa configuration]) AC_ARG_WITH(testsound, - [ --with-testsound=file give the path of test sound file for alsaconf], + AS_HELP_STRING([--with-testsound=file], [give the path of test sound file for alsaconf]), TESTSOUND="$withval", TESTSOUND="$dir/test.wav") AC_SUBST(TESTSOUND) -- 1.7.8.5

2 1

[alsa-devel] Fast Track Ultra mixer - opinions anyone?
by Felix Homann 18 Apr '12

18 Apr '12

Hi, over the weekend I tried to prepare a "set-everything-to-zero-by-default" mixer patch. And failed... I could get the controls to be defaulted to zero by quirking the volume range, but then there would not have been any sound at all, even after adjusting the levels. So without diving any deeper in this I decided to put some work in the mixer controls. You can find a patch to play with attached to this mail. I've left a couple of commented quirks in mixer.c so you can see, where simple changes can be made. But working on this patch raised a couple of questions I would like hear your opinions on: ** Effects ** Essentially, with the attached patch you can control everything but the effect program. The switch should not be too hard to implement. But before I put any more work on it I would like to ask wether it makes sense at all. IMO, the effect section on the FTUs really sucks. I can't believe that anybody can really use it in a reasonable way. So should we really add another 40 controls to the mixer which can cause a lot of pain when misadjusted (i.e. anything else but 0)? For those reasons I would prefer not to put effect support in the kernel. What do you think? ** dB values ** Currently alsamixer does not report any dB changes when you change a control value. I don't know how to fix it. Anyone else? ** Naming scheme again ** Once again, is everyone pleased with the current naming scheme? Looking forward for your suggestions. Regards, Felix

4 7

[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

5 28

[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

2 1

[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.

2 1

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

1 0

[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

1 0

[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

2 1

[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.

2 1

[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

2 3

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

17 Apr '12

Current DA7210 driver does support PLL mode fully. It uses fixed value of input master clock and PLL mode is enabled and disabled based on the sampling frequency being used for playback or recording. It also doesn't support Sample Rate Measurement feature of DA7210 hardware. This patch adds full support for PLL and SRM. Basically following three modes of operation are possible for DA7210 hardware, (1) I2S SLAVE mode with PLL bypassed (2) I2S SLAVE mode with PLL enabled (3) I2S Master mode with PLL enabled This patch adds support for all three modes. Also, in case of SLAVE mode with PLL, it supports SRM (Sample Rate Measurement) feature of the chip. Actually this patch was submitted earlier and received some review comments, but after that the driver got update by other patches. Because of that, I am considering this as new patch and not versioning it based of previous patches. This version tries to take care of all review comments received for earlier submissions. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Revamped logic to find correct pll divisors. Now it doesn't reply on magic numbers. This is done with the intension to make code more readable and easily maintainable. - Also updated data structure for pll divisors to accommodate above logic. --- sound/soc/codecs/da7210.c | 225 +++++++++++++++++++++++++++++++++++++-------- 1 files changed, 187 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 1e3716d..cf4063b 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -148,6 +148,7 @@ #define DA7210_DAI_EN (1 << 7) /*PLL_DIV3 bit fields */ +#define DA7210_PLL_DIV_L_MASK (0xF << 0) #define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) #define DA7210_PLL_BYP (1 << 6) @@ -164,12 +165,16 @@ #define DA7210_PLL_FS_48000 (0xB << 0) #define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) +#define DA7210_MCLK_DET_EN (0x1 << 5) +#define DA7210_MCLK_SRM_EN (0x1 << 6) #define DA7210_PLL_EN (0x1 << 7) /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) /* CONTROL bit fields */ +#define DA7210_REG_EN (1 << 0) +#define DA7210_BIAS_EN (1 << 2) #define DA7210_NOISE_SUP_EN (1 << 3) /* IN_GAIN bit fields */ @@ -208,6 +213,47 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +struct pll_div { + int fref; + int fout; + u8 div1; + u8 div2; + u8 div3; + u8 mode; /* 0 = slave, 1 = master */ +}; + +/* PLL dividers table */ +static const struct pll_div da7210_pll_div[] = { + /* for MASTER mode, fs = 44.1Khz */ + { 12000000, 2822400, 0xE8, 0x6C, 0x2, 1}, /* MCLK=12Mhz */ + { 13000000, 2822400, 0xDF, 0x28, 0xC, 1}, /* MCLK=13Mhz */ + { 13500000, 2822400, 0xDB, 0x0A, 0xD, 1}, /* MCLK=13.5Mhz */ + { 14400000, 2822400, 0xD4, 0x5A, 0x2, 1}, /* MCLK=14.4Mhz */ + { 19200000, 2822400, 0xBB, 0x43, 0x9, 1}, /* MCLK=19.2Mhz */ + { 19680000, 2822400, 0xB9, 0x6D, 0xA, 1}, /* MCLK=19.68Mhz */ + { 19800000, 2822400, 0xB8, 0xFB, 0xB, 1}, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz */ + { 12000000, 3072000, 0xF3, 0x12, 0x7, 1}, /* MCLK=12Mhz */ + { 13000000, 3072000, 0xE8, 0xFD, 0x5, 1}, /* MCLK=13Mhz */ + { 13500000, 3072000, 0xE4, 0x82, 0x3, 1}, /* MCLK=13.5Mhz */ + { 14400000, 3072000, 0xDD, 0x3A, 0x0, 1}, /* MCLK=14.4Mhz */ + { 19200000, 3072000, 0xC1, 0xEB, 0x8, 1}, /* MCLK=19.2Mhz */ + { 19680000, 3072000, 0xBF, 0xEC, 0x0, 1}, /* MCLK=19.68Mhz */ + { 19800000, 3072000, 0xBF, 0x70, 0x0, 1}, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + { 12000000, 2822400, 0xED, 0xBF, 0x5, 0}, /* MCLK=12Mhz */ + { 13000000, 2822400, 0xE4, 0x13, 0x0, 0}, /* MCLK=13Mhz */ + { 13500000, 2822400, 0xDF, 0xC6, 0x8, 0}, /* MCLK=13.5Mhz */ + { 14400000, 2822400, 0xD8, 0xCA, 0x1, 0}, /* MCLK=14.4Mhz */ + { 19200000, 2822400, 0xBE, 0x97, 0x9, 0}, /* MCLK=19.2Mhz */ + { 19680000, 2822400, 0xBC, 0xAC, 0xD, 0}, /* MCLK=19.68Mhz */ + { 19800000, 2822400, 0xBC, 0x35, 0xE, 0}, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA7210_CLKSRC_MCLK +}; + #define DA7210_VERSION "0.0.1" /* @@ -630,6 +676,8 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Codec private data */ struct da7210_priv { struct regmap *regmap; + unsigned int mclk_rate; + int master; }; static struct reg_default da7210_reg_defaults[] = { @@ -717,8 +765,9 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; - u32 fs, bypass; + u32 fs, sysclk; /* set DAI source to Left and Right ADC */ snd_soc_write(codec, DA7210_DAI_SRC_SEL, @@ -751,43 +800,43 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 11025: fs = DA7210_PLL_FS_11025; - bypass = 0; + sysclk = 2822400; break; case 12000: fs = DA7210_PLL_FS_12000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 16000: fs = DA7210_PLL_FS_16000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 22050: fs = DA7210_PLL_FS_22050; - bypass = 0; + sysclk = 2822400; break; case 32000: fs = DA7210_PLL_FS_32000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 44100: fs = DA7210_PLL_FS_44100; - bypass = 0; + sysclk = 2822400; break; case 48000: fs = DA7210_PLL_FS_48000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 88200: fs = DA7210_PLL_FS_88200; - bypass = 0; + sysclk = 2822400; break; case 96000: fs = DA7210_PLL_FS_96000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; default: return -EINVAL; @@ -797,8 +846,15 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); - snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + if (da7210->mclk_rate && (da7210->mclk_rate != sysclk)) { + /* PLL mode, disable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0); + } else { + /* PLL bypass mode, enable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, + DA7210_PLL_BYP); + } /* Enable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, DA7210_SC_MST_EN); @@ -812,17 +868,24 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; u32 dai_cfg3; dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); + if ((snd_soc_read(codec, DA7210_PLL) & DA7210_PLL_EN) && + (!(snd_soc_read(codec, DA7210_PLL_DIV3) & DA7210_PLL_BYP))) + return -EINVAL; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: + da7210->master = 1; dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: + da7210->master = 0; dai_cfg1 |= DA7210_DAI_MODE_SLAVE; break; default: @@ -874,10 +937,114 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static int da7210_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 da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA7210_CLKSRC_MCLK: + switch (freq) { + case 12000000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da7210->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/** + * da7210_set_dai_pll :Configure the codec PLL + * @param codec_dai : pointer to codec DAI + * @param pll_id : da7210 has only one pll, so pll_id is always zero + * @param fref : MCLK frequency, should be < 20MHz + * @param fout : FsDM value, Refer page 44 & 45 of datasheet + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz, + * 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + u8 pll_div1, pll_div2, pll_div3, cnt; + + /* In slave mode, there is only one set of divisors */ + if (!da7210->master) + fout = 2822400; + + /* 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; + } + } + } + } + if (cnt >= ARRAY_SIZE(da7210_pll_div)) + goto err; + + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + /* Write PLL dividers */ + snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1); + snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, + DA7210_PLL_DIV_L_MASK, pll_div3); + + if (da7210->master) { + /* In master mode, no need to enable SRM */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, + DA7210_PLL_EN); + } else { + /* In slave mode, enable SRM and PLL */ + snd_soc_update_bits(codec, DA7210_PLL, + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN), + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN)); + } + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, + DA7210_SC_MST_EN); + return 0; +err: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", fref); + return -EINVAL; +} + /* DAI operations */ static const struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .set_sysclk = da7210_set_dai_sysclk, + .set_pll = da7210_set_dai_pll, .digital_mute = da7210_mute, }; @@ -917,17 +1084,11 @@ static int da7210_probe(struct snd_soc_codec *codec) return ret; } - /* FIXME - * - * This driver use fixed value here - * And below settings expects MCLK = 12.288MHz - * - * When you select different MCLK, please check... - * DA7210_PLL_DIV1 val - * DA7210_PLL_DIV2 val - * DA7210_PLL_DIV3 val - * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx - */ + da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ + da7210->master = 0; /* This will be set from set_fmt() */ + + /* Enable internal regulator & bias current */ + snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN); /* * ADC settings @@ -1002,24 +1163,12 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Set PLL Master clock range 10-20 MHz */ + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* - * If 48kHz sound came, it use bypass mode, - * and when it is 44.1kHz, it use PLL. - * - * This time, this driver sets PLL always ON - * and controls bypass/PLL mode by switching - * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. - * see da7210_hw_params - */ - snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); - snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); - snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); - /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); -- 1.7.1

2 3

[alsa-devel] [PATCH 0/6] snd-usb endpoint rework, version 7
by Daniel Mack 17 Apr '12

17 Apr '12

Here's the 7th iteration of patches that reimplement the streaming logic of the ALSA USB audio driver. There were some bugs reportes by Felix and Grant, which have been resolved. More testing is definitely important, but maybe the code will get a wider test coverage when merged to the development tree? Daniel Daniel Mack (6): ALSA: snd-usb: add snd_usb_audio-wide mutex ALSA: snd-usb: implement new endpoint streaming model ALSA: snd-usb: switch over to new endpoint streaming logic ALSA: snd-usb: remove old streaming logic ALSA: snd-usb: add support for implicit feedback ALSA: snd-usb: add some documentation sound/usb/card.c | 10 +- sound/usb/card.h | 62 ++ sound/usb/endpoint.c | 1601 ++++++++++++++++++++++++++++---------------------- sound/usb/endpoint.h | 32 +- sound/usb/pcm.c | 441 ++++++++++++--- sound/usb/stream.c | 31 +- sound/usb/usbaudio.h | 2 + 7 files changed, 1381 insertions(+), 798 deletions(-) -- 1.7.7.6

5 24

[alsa-devel] M-Audio FastTrack C400
by Chris Cavey 17 Apr '12

17 Apr '12

I'd love to be able to help ALSA support this USB sound device. I've currently got the device working in a VM on my Ubuntu 12.04 workstation. I believe I can capture USB packets. Would this be helpful to anyone? Any pointers on where to get started?

1 0

[alsa-devel] system suspend and live audio streams
by Grazvydas Ignotas 17 Apr '12

17 Apr '12

Hi, Currently on OMAP the system fails to suspend properly if there are audio streams active (PER and CORE don't enter low power states). I wonder if this is expected to be handled by the driver and can be considered a bug, or is it audio daemon's/system's responsibility to stop all audio streams before kernel to is asked to start suspending? -- Gražvydas

4 5

[alsa-devel] alsamonitor on SunOS
by james pruett 17 Apr '12

17 Apr '12

I love the alsamonitor look and feel. I want to port it to my SunOS box. Q: Is there a Hello World I might try first to see if the colors, etc work? Q: what visual toolkit/gizmo is alsamonitor based upon? Any leads appreciated! -cellurl

2 1

[alsa-devel] [PATCH -next] soc: fix printk format in soc-dapm.c
by Randy Dunlap 17 Apr '12

17 Apr '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix printk format warning for 'formats': sound/soc/soc-dapm.c:2913:5: warning: format '%lx' expects type 'long unsigned int', but argument 3 has type 'u64' Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20120416.orig/sound/soc/soc-dapm.c +++ linux-next-20120416/sound/soc/soc-dapm.c @@ -2909,7 +2909,7 @@ static int snd_soc_dai_link_event(struct if (config->formats) { fmt = ffs(config->formats) - 1; } else { - dev_warn(w->dapm->dev, "Invalid format %lx specified\n", + dev_warn(w->dapm->dev, "Invalid format %llx specified\n", config->formats); fmt = 0; }

2 1

[alsa-devel] USB audio devices on Audio 4DJ and similar
by Alan Horstmann 16 Apr '12

16 Apr '12

Hi Daniel and Alsa folks, At present I am involved with resolving some issues that have arisen from people using the Audio 4DJ interface with Portaudio, and I hoped it would be OK to check a few Alsa details with you? Unfortunately I have limited experience of USB audio. Apparently the 4 audio inputs/outputs appear as 2 stereo sub-devices on the first device, rather than a single 4-channel device - is this correct? Assuming that is correct, does that arise as a result of the intrinsic nature of the interface, or is it by developer choice in the driver (for example so that it can be opened as 2 stereo streams)? Is there (or could there be) a way to change this, eg by module option or udev rule? I notice in the source #define CHANNELS_PER_STREAM 2 Is there a way at present that the device can be accessed as a single 4-channel stream other than by creating a custom ~asoundrc with a pcm definition using 'multi' to join the 2 stereo streams? I don't see any USB audio units having files in /usr/share/alsa/cards; is it not relevant, not normally necessary or something no-one has got round to writing? I appreciate any clarification you can bring. Regards Alan

4 8

Re: [alsa-devel] [PATCH] ASoC: soc-dapm: Use '%llx' with 'u64' type.
by Mark Brown 16 Apr '12

16 Apr '12

On Mon, Apr 16, 2012 at 10:53:09AM -0300, Fabio Estevam wrote: > Fix the following build warning: > > sound/soc/soc-dapm.c: In function 'snd_soc_dai_link_event': > sound/soc/soc-dapm.c:2913: warning: format '%lx' expects type 'long unsigned int', but argument 3 has type 'u64' > > '%llx' should be used with 'u64' type. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: core: Support transparent CODEC<->CODEC DAI links
by Mark Brown 16 Apr '12

16 Apr '12

Rather than having the user half start a stream but avoid any DMA to trigger data flow on links which don't pass through the CPU create a DAPM route between the two DAI widgets using a hw_params configuration provided by the machine driver with the new 'params' member of the dai_link struct. If no configuration is provided in the dai_link then use the old style even for CODEC<->CODEC links to avoid breaking systems. This greatly simplifies the userspace usage of such links, making them as simple as analogue connections with the stream configuration being completely transparent to them. This is achieved by defining a new dai_link widget type which is created when CODECs are linked and triggering the configuration of the link via the normal PCM operations from there. It is expected that the bias level callbacks will be used for clock configuration. Currently only the DAI format, rate and channel count can be configured and currently the only DAI operations which can be called are hw_params and digital_mute(). This corresponds well to the majority of CODEC drivers which only use other callbacks for constraint setting but there is obviously much room for extension here. We can't simply call hw_params() on startup as things like the system clocking configuration may change at runtime and in future it will be desirable to offer some configurability of the link parameters. At present we are also restricted to a single DAPM link for the entire DAI. Once we have better support for channel mapping it would also be desirable to extend this feature so that we can propagate per-channel power state over the link. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 6 ++ include/sound/soc.h | 2 + sound/soc/soc-core.c | 43 +++++++++++-- sound/soc/soc-dapm.c | 155 +++++++++++++++++++++++++++++++++++++++++++++- 4 files changed, 196 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 01e7ad1..bea0c86 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -356,6 +356,10 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, struct snd_soc_dai *dai); int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink); /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm); @@ -427,6 +431,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ snd_soc_dapm_dai, /* link to DAI structure */ + snd_soc_dapm_dai_link, /* link between two DAI structures */ }; enum snd_soc_dapm_subclass { @@ -485,6 +490,7 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ + const struct snd_soc_pcm_stream *params; /* params for dai links */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index acb57b8..afc3204 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -761,6 +761,8 @@ struct snd_soc_dai_link { const struct device_node *cpu_dai_of_node; const char *codec_dai_name; + const struct snd_soc_pcm_stream *params; + unsigned int dai_fmt; /* format to set on init */ /* Keep DAI active over suspend */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 20a15a3..6953ea3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1191,7 +1191,9 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dapm_widget *play_w, *capture_w; int ret; dev_dbg(card->dev, "probe %s dai link %d late %d\n", @@ -1272,12 +1274,39 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (ret < 0) pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret); - /* create the pcm */ - ret = soc_new_pcm(rtd, num); - if (ret < 0) { - pr_err("asoc: can't create pcm %s :%d\n", - dai_link->stream_name, ret); - return ret; + if (!dai_link->params) { + /* create the pcm */ + ret = soc_new_pcm(rtd, num); + if (ret < 0) { + pr_err("asoc: can't create pcm %s :%d\n", + dai_link->stream_name, ret); + return ret; + } + } else { + /* link the DAI widgets */ + play_w = codec_dai->playback_widget; + capture_w = cpu_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } + + play_w = cpu_dai->playback_widget; + capture_w = codec_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } } /* add platform data for AC97 devices */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7c8ff90..69886b6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -52,6 +52,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, [snd_soc_dapm_micbias] = 2, + [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai] = 3, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, @@ -90,9 +91,10 @@ static int dapm_down_seq[] = { [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, [snd_soc_dapm_dai] = 10, - [snd_soc_dapm_regulator_supply] = 11, - [snd_soc_dapm_supply] = 11, - [snd_soc_dapm_post] = 12, + [snd_soc_dapm_dai_link] = 11, + [snd_soc_dapm_regulator_supply] = 12, + [snd_soc_dapm_supply] = 12, + [snd_soc_dapm_post] = 13, }; static void pop_wait(u32 pop_time) @@ -396,6 +398,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_mic: case snd_soc_dapm_spk: case snd_soc_dapm_line: + case snd_soc_dapm_dai_link: p->connect = 1; break; /* does affect routing - dynamically connected */ @@ -2081,6 +2084,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: + case snd_soc_dapm_dai_link: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -2809,6 +2813,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_line: + case snd_soc_dapm_dai_link: w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_supply: @@ -2871,6 +2876,150 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); +static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_dapm_path *source_p, *sink_p; + struct snd_soc_dai *source, *sink; + const struct snd_soc_pcm_stream *config = w->params; + struct snd_pcm_substream substream; + struct snd_pcm_hw_params params; + u64 fmt; + int ret; + + BUG_ON(!config); + BUG_ON(list_empty(&w->sources) || list_empty(&w->sinks)); + + /* We only support a single source and sink, pick the first */ + source_p = list_first_entry(&w->sources, struct snd_soc_dapm_path, + list_sink); + sink_p = list_first_entry(&w->sinks, struct snd_soc_dapm_path, + list_source); + + BUG_ON(!source_p || !sink_p); + BUG_ON(!sink_p->source || !source_p->sink); + BUG_ON(!source_p->source || !sink_p->sink); + + source = source_p->source->priv; + sink = sink_p->sink->priv; + + /* Be a little careful as we don't want to overflow the mask array */ + if (config->formats) { + fmt = ffs(config->formats) - 1; + } else { + dev_warn(w->dapm->dev, "Invalid format %lx specified\n", + config->formats); + fmt = 0; + } + + /* Currently very limited parameter selection */ + memset(&params, 0, sizeof(params)); + snd_mask_set(hw_param_mask(&params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); + + hw_param_interval(&params, SNDRV_PCM_HW_PARAM_RATE)->min = + config->rate_min; + hw_param_interval(&params, SNDRV_PCM_HW_PARAM_RATE)->max = + config->rate_max; + + hw_param_interval(&params, SNDRV_PCM_HW_PARAM_CHANNELS)->min + = config->channels_min; + hw_param_interval(&params, SNDRV_PCM_HW_PARAM_CHANNELS)->max + = config->channels_max; + + memset(&substream, 0, sizeof(substream)); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (source->driver->ops && source->driver->ops->hw_params) { + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + ret = source->driver->ops->hw_params(&substream, + &params, source); + if (ret != 0) { + dev_err(source->dev, + "hw_params() failed: %d\n", ret); + return ret; + } + } + + if (sink->driver->ops && sink->driver->ops->hw_params) { + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + ret = sink->driver->ops->hw_params(&substream, &params, + sink); + if (ret != 0) { + dev_err(sink->dev, + "hw_params() failed: %d\n", ret); + return ret; + } + } + break; + + case SND_SOC_DAPM_POST_PMU: + ret = snd_soc_dai_digital_mute(sink, 0); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, "Failed to unmute: %d\n", ret); + break; + + case SND_SOC_DAPM_PRE_PMD: + ret = snd_soc_dai_digital_mute(sink, 1); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, "Failed to mute: %d\n", ret); + break; + + default: + BUG(); + return -EINVAL; + } + + return 0; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_route routes[2]; + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + size_t len; + char *link_name; + + len = strlen(source->name) + strlen(sink->name) + 2; + link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!link_name) + return -ENOMEM; + snprintf(link_name, len, "%s-%s", source->name, sink->name); + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + + dev_dbg(card->dev, "adding %s widget\n", link_name); + + w = snd_soc_dapm_new_control(&card->dapm, &template); + if (!w) { + dev_err(card->dev, "Failed to create %s widget\n", + link_name); + return -ENOMEM; + } + + w->params = params; + + memset(&routes, 0, sizeof(routes)); + + routes[0].source = source->name; + routes[0].sink = link_name; + routes[1].source = link_name; + routes[1].sink = sink->name; + + return snd_soc_dapm_add_routes(&card->dapm, routes, + ARRAY_SIZE(routes)); +} + int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, struct snd_soc_dai *dai) { -- 1.7.9.1

2 2

[alsa-devel] [PATCH] ASoC: dapm: Ensure power gets managed for line widgets
by Mark Brown 16 Apr '12

16 Apr '12

Line widgets had not been included in either the power up or power down sequences so if a widget had an event associated with it that event would never be run. Fix this minimally by adding them to the sequences, we should probably be doing away with the specific widget types as they all have the same priority anyway. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 922e693..1cde549 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -68,6 +68,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_out_drv] = 10, [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, + [snd_soc_dapm_line] = 10, [snd_soc_dapm_post] = 11, }; @@ -76,6 +77,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_adc] = 1, [snd_soc_dapm_hp] = 2, [snd_soc_dapm_spk] = 2, + [snd_soc_dapm_line] = 2, [snd_soc_dapm_out_drv] = 2, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_mixer_named_ctl] = 5, -- 1.7.9.1

2 1

[alsa-devel] [PATCH] ASoC: core: Return -ENOTSUPP instead of -EINVAL if mute is not supported
by Mark Brown 16 Apr '12

16 Apr '12

This helps us ignore errors in callers if the operation failed due to not being available as opposed to an error. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1ff0161..b957e68 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3090,7 +3090,7 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) if (dai->driver && dai->driver->ops->digital_mute) return dai->driver->ops->digital_mute(dai, mute); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); -- 1.7.9.1

2 1

Re: [alsa-devel] [PATCH for 3.4] MFD: twl6040: Convert to i2c driver, and separate it from twl core
by Peter Ujfalusi 16 Apr '12

16 Apr '12

Hi Samuel, On 04/16/2012 05:46 PM, Samuel Ortiz wrote: > Applied to my for-linus branch. I'll send a pull request this week, so I > expect it to make it to rc4. Thank you! -- Péter

1 0

[alsa-devel] [PATCH] ASoC: da7210: Add support for PLL and SRM
by Ashish Chavan 16 Apr '12

16 Apr '12

Current DA7210 driver does support PLL mode fully. It uses fixed value of input master clock and PLL mode is enabled and disabled based on the sampling frequency being used for playback or recording. It also doesn't support Sample Rate Measurement feature of DA7210 hardware. This patch adds full support for PLL and SRM. Basically following three modes of operation are possible for DA7210 hardware, (1) I2S SLAVE mode with PLL bypassed (2) I2S SLAVE mode with PLL enabled (3) I2S Master mode with PLL enabled This patch adds support for all three modes. Also, in case of SLAVE mode with PLL, it supports SRM (Sample Rate Measurement) feature of the chip. Actually this patch was submitted earlier and received some review comments, but after that the driver got update by other patches. Because of that, I am considering this as new patch and not versioning it based of previous patches. This version tries to take care of all review comments received for earlier submissions. 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 | 237 +++++++++++++++++++++++++++++++++++++------- 1 files changed, 199 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 5dfdf6e..3864ccc 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -148,6 +148,7 @@ #define DA7210_DAI_EN (1 << 7) /*PLL_DIV3 bit fields */ +#define DA7210_PLL_DIV_L_MASK (0xF << 0) #define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) #define DA7210_PLL_BYP (1 << 6) @@ -164,12 +165,16 @@ #define DA7210_PLL_FS_48000 (0xB << 0) #define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) +#define DA7210_MCLK_DET_EN (0x1 << 5) +#define DA7210_MCLK_SRM_EN (0x1 << 6) #define DA7210_PLL_EN (0x1 << 7) /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) /* CONTROL bit fields */ +#define DA7210_REG_EN (1 << 0) +#define DA7210_BIAS_EN (1 << 2) #define DA7210_NOISE_SUP_EN (1 << 3) /* IN_GAIN bit fields */ @@ -208,6 +213,53 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +/* Number of PLL input frequencies supported */ +#define FREF_CNT (7 << 0) + +/* Index of first relevant row in PLL divider table */ +#define MASTER_2822400_DIV_OFFSET 0 +#define MASTER_3072000_DIV_OFFSET (MASTER_2822400_DIV_OFFSET + FREF_CNT) +#define SLAVE_SRM_DIV_OFFSET (MASTER_3072000_DIV_OFFSET + FREF_CNT) + +struct pll_div { + int fref; + u8 div1; + u8 div2; + u8 div3; +}; + +/* PLL dividers table */ +static const struct pll_div da7210_pll_div[] = { + /* for MASTER mode, fs = 44.1Khz */ + { 12000000, 0xE8, 0x6C, 0x2, }, /* MCLK=12Mhz */ + { 13000000, 0xDF, 0x28, 0xC, }, /* MCLK=13Mhz */ + { 13500000, 0xDB, 0x0A, 0xD, }, /* MCLK=13.5Mhz */ + { 14400000, 0xD4, 0x5A, 0x2, }, /* MCLK=14.4Mhz */ + { 19200000, 0xBB, 0x43, 0x9, }, /* MCLK=19.2Mhz */ + { 19680000, 0xB9, 0x6D, 0xA, }, /* MCLK=19.68Mhz */ + { 19800000, 0xB8, 0xFB, 0xB, }, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz */ + { 12000000, 0xF3, 0x12, 0x7, }, /* MCLK=12Mhz */ + { 13000000, 0xE8, 0xFD, 0x5, }, /* MCLK=13Mhz */ + { 13500000, 0xE4, 0x82, 0x3, }, /* MCLK=13.5Mhz */ + { 14400000, 0xDD, 0x3A, 0x0, }, /* MCLK=14.4Mhz */ + { 19200000, 0xC1, 0xEB, 0x8, }, /* MCLK=19.2Mhz */ + { 19680000, 0xBF, 0xEC, 0x0, }, /* MCLK=19.68Mhz */ + { 19800000, 0xBF, 0x70, 0x0, }, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + { 12000000, 0xED, 0xBF, 0x5, }, /* MCLK=12Mhz */ + { 13000000, 0xE4, 0x13, 0x0, }, /* MCLK=13Mhz */ + { 13500000, 0xDF, 0xC6, 0x8, }, /* MCLK=13.5Mhz */ + { 14400000, 0xD8, 0xCA, 0x1, }, /* MCLK=14.4Mhz */ + { 19200000, 0xBE, 0x97, 0x9, }, /* MCLK=19.2Mhz */ + { 19680000, 0xBC, 0xAC, 0xD, }, /* MCLK=19.68Mhz */ + { 19800000, 0xBC, 0x35, 0xE, }, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA7210_CLKSRC_MCLK +}; + #define DA7210_VERSION "0.0.1" /* @@ -630,6 +682,8 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Codec private data */ struct da7210_priv { struct regmap *regmap; + unsigned int mclk_rate; + int master; }; static struct reg_default da7210_reg_defaults[] = { @@ -717,8 +771,9 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; - u32 fs, bypass; + u32 fs, sysclk; /* set DAI source to Left and Right ADC */ snd_soc_write(codec, DA7210_DAI_SRC_SEL, @@ -751,43 +806,43 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 11025: fs = DA7210_PLL_FS_11025; - bypass = 0; + sysclk = 2822400; break; case 12000: fs = DA7210_PLL_FS_12000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 16000: fs = DA7210_PLL_FS_16000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 22050: fs = DA7210_PLL_FS_22050; - bypass = 0; + sysclk = 2822400; break; case 32000: fs = DA7210_PLL_FS_32000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 44100: fs = DA7210_PLL_FS_44100; - bypass = 0; + sysclk = 2822400; break; case 48000: fs = DA7210_PLL_FS_48000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 88200: fs = DA7210_PLL_FS_88200; - bypass = 0; + sysclk = 2822400; break; case 96000: fs = DA7210_PLL_FS_96000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; default: return -EINVAL; @@ -797,8 +852,15 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); - snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + if (da7210->mclk_rate && (da7210->mclk_rate != sysclk)) { + /* PLL mode, disable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0); + } else { + /* PLL bypass mode, enable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, + DA7210_PLL_BYP); + } /* Enable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, DA7210_SC_MST_EN); @@ -812,17 +874,24 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; u32 dai_cfg3; dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); + if ((snd_soc_read(codec, DA7210_PLL) & DA7210_PLL_EN) && + (!(snd_soc_read(codec, DA7210_PLL_DIV3) & DA7210_PLL_BYP))) + return -EINVAL; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: + da7210->master = 1; dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: + da7210->master = 0; dai_cfg1 |= DA7210_DAI_MODE_SLAVE; break; default: @@ -874,10 +943,120 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static int da7210_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 da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA7210_CLKSRC_MCLK: + switch (freq) { + case 12000000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da7210->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/** + * da7210_set_dai_pll :Configure the codec PLL + * @param codec_dai : pointer to codec DAI + * @param pll_id : da7210 has only one pll, so pll_id is always zero + * @param fref : MCLK frequency, should be < 20MHz + * @param fout : FsDM value, Refer page 44 & 45 of datasheet + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz, + * 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + u8 pll_div1, pll_div2, pll_div3, row_idx, cnt; + + if (da7210->master) { + /* In PLL master mode, use master mode PLL dividers */ + switch (fout) { + case 2822400: + row_idx = MASTER_2822400_DIV_OFFSET; + break; + case 3072000: + row_idx = MASTER_3072000_DIV_OFFSET; + break; + default: + dev_err(codec_dai->dev, + "Unsupported PLL output frequency %d\n", fout); + return -EINVAL; + } + } else { + /* In PLL slave mode, use SRM mode PLL dividers */ + row_idx = SLAVE_SRM_DIV_OFFSET; + } + + for (cnt = 0; cnt < FREF_CNT; cnt++) { + if (fref == da7210_pll_div[row_idx + cnt].fref) { + pll_div1 = da7210_pll_div[row_idx + cnt].div1; + pll_div2 = da7210_pll_div[row_idx + cnt].div2; + pll_div3 = da7210_pll_div[row_idx + cnt].div3; + break; + } + } + if (cnt >= FREF_CNT) + goto err; + + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + /* Write PLL dividers */ + snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1); + snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, + DA7210_PLL_DIV_L_MASK, pll_div3); + + if (da7210->master) { + /* In master mode, no need to enable SRM */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, + DA7210_PLL_EN); + } else { + /* In slave mode, enable SRM and PLL */ + snd_soc_update_bits(codec, DA7210_PLL, + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN), + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN)); + } + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, + DA7210_SC_MST_EN); + return 0; +err: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", fref); + return -EINVAL; +} + /* DAI operations */ static const struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .set_sysclk = da7210_set_dai_sysclk, + .set_pll = da7210_set_dai_pll, .digital_mute = da7210_mute, }; @@ -917,17 +1096,11 @@ static int da7210_probe(struct snd_soc_codec *codec) return ret; } - /* FIXME - * - * This driver use fixed value here - * And below settings expects MCLK = 12.288MHz - * - * When you select different MCLK, please check... - * DA7210_PLL_DIV1 val - * DA7210_PLL_DIV2 val - * DA7210_PLL_DIV3 val - * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx - */ + da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ + da7210->master = 0; /* This will be set from set_fmt() */ + + /* Enable internal regulator & bias current */ + snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN); /* * ADC settings @@ -1002,24 +1175,12 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Set PLL Master clock range 10-20 MHz */ + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* - * If 48kHz sound came, it use bypass mode, - * and when it is 44.1kHz, it use PLL. - * - * This time, this driver sets PLL always ON - * and controls bypass/PLL mode by switching - * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. - * see da7210_hw_params - */ - snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); - snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); - snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); - /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); -- 1.7.1

2 5

[alsa-devel] [PATCH v2] control_external: Add ability to specify TLV data.
by Dylan Reid 16 Apr '12

16 Apr '12

Allow external control plugins to provide TLV data. This allows user-space pcms to specify dB ranges for controls. This follows the same model as the ALSA drivers for accessing the data. The code is based on that implementation. The control can provide static data or a callback. The data is accessed or modified in the new snd_ctl_ext_elem_tlv callback. Rev bump the protocol version to enable checking if an external control supports TLV. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- include/control_external.h | 17 +++++++++++++++- src/control/control_ext.c | 46 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 62 insertions(+), 1 deletions(-) diff --git a/include/control_external.h b/include/control_external.h index 7c066cf..5392ea6 100644 --- a/include/control_external.h +++ b/include/control_external.h @@ -60,13 +60,16 @@ typedef struct snd_ctl_ext snd_ctl_ext_t; typedef struct snd_ctl_ext_callback snd_ctl_ext_callback_t; /** Key to access a control pointer */ typedef unsigned long snd_ctl_ext_key_t; +/** Callback to handle TLV commands. */ +typedef int (snd_ctl_ext_tlv_rw_t)(snd_ctl_ext_t *ext, snd_ctl_ext_key_t key, int op_flag, unsigned int numid, + unsigned int *tlv, unsigned int tlv_size); /* * Protocol version */ #define SND_CTL_EXT_VERSION_MAJOR 1 /**< Protocol major version */ #define SND_CTL_EXT_VERSION_MINOR 0 /**< Protocol minor version */ -#define SND_CTL_EXT_VERSION_TINY 0 /**< Protocol tiny version */ +#define SND_CTL_EXT_VERSION_TINY 1 /**< Protocol tiny version */ /** * external plugin protocol version */ @@ -122,6 +125,13 @@ struct snd_ctl_ext { * control handle filled by #snd_ctl_ext_create() */ snd_ctl_t *handle; + /** + * optional TLV data for the control. + */ + union { + snd_ctl_ext_tlv_rw_t *c; + const unsigned int *p; + } tlv; int nonblock; /**< non-block mode; read-only */ int subscribed; /**< events subscribed; read-only */ @@ -245,7 +255,12 @@ typedef enum snd_ctl_ext_access { SND_CTL_EXT_ACCESS_WRITE = (1<<1), SND_CTL_EXT_ACCESS_READWRITE = (3<<0), SND_CTL_EXT_ACCESS_VOLATILE = (1<<2), + SND_CTL_EXT_ACCESS_TLV_READ = (1<<4), + SND_CTL_EXT_ACCESS_TLV_WRITE = (1<<5), + SND_CTL_EXT_ACCESS_TLV_READWRITE = (3<<4), + SND_CTL_EXT_ACCESS_TLV_COMMAND = (1<<6), SND_CTL_EXT_ACCESS_INACTIVE = (1<<8), + SND_CTL_EXT_ACCESS_TLV_CALLBACK = (1<<28), } snd_ctl_ext_access_t; /** diff --git a/src/control/control_ext.c b/src/control/control_ext.c index e20d4f3..0a0b397 100644 --- a/src/control/control_ext.c +++ b/src/control/control_ext.c @@ -324,6 +324,51 @@ static int snd_ctl_ext_elem_unlock(snd_ctl_t *handle ATTRIBUTE_UNUSED, return -ENXIO; } +static int snd_ctl_ext_elem_tlv(snd_ctl_t *handle, int op_flag, + unsigned int numid, + unsigned int *tlv, unsigned int tlv_size) +{ + snd_ctl_ext_t *ext = handle->private_data; + snd_ctl_ext_key_t key; + int type, ret; + unsigned int access, count, len; + snd_ctl_elem_id_t id; + + /* we don't support TLV on protocol ver 1.0.0 or earlier */ + if (ext->version <= SNDRV_PROTOCOL_VERSION(1, 0, 0)) + return -ENXIO; + + snd_ctl_elem_id_clear(&id); + if (numid > 0) { + ext->callback->elem_list(ext, numid - 1, &id); + id.numid = numid; + } else + id.numid = 0; + key = ext->callback->find_elem(ext, &id); + + if (key == SND_CTL_EXT_KEY_NOT_FOUND) + return -ENOENT; + ret = ext->callback->get_attribute(ext, key, &type, &access, &count); + if (ret < 0) + return ret; + + if ((op_flag == 0 && (access & SND_CTL_EXT_ACCESS_TLV_READ) == 0) || + (op_flag > 0 && (access & SND_CTL_EXT_ACCESS_TLV_WRITE) == 0) || + (op_flag < 0 && (access & SND_CTL_EXT_ACCESS_TLV_COMMAND) == 0)) + return -ENXIO; + if (access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + return ext->tlv.c(ext, key, op_flag, numid, tlv, tlv_size); + } else { + if (op_flag) + return -ENXIO; + len = ext->tlv.p[1] + 2 * sizeof(unsigned int); + if (tlv_size < len) + return -ENOMEM; + memcpy(tlv, ext->tlv.p, len); + return 0; + } +} + static int snd_ctl_ext_next_device(snd_ctl_t *handle ATTRIBUTE_UNUSED, int *device ATTRIBUTE_UNUSED) { @@ -429,6 +474,7 @@ static const snd_ctl_ops_t snd_ctl_ext_ops = { .element_write = snd_ctl_ext_elem_write, .element_lock = snd_ctl_ext_elem_lock, .element_unlock = snd_ctl_ext_elem_unlock, + .element_tlv = snd_ctl_ext_elem_tlv, .hwdep_next_device = snd_ctl_ext_next_device, .hwdep_info = snd_ctl_ext_hwdep_info, .pcm_next_device = snd_ctl_ext_next_device, -- 1.7.9.rc0

2 1

[alsa-devel] [PATCH 1/2] snd-usb-audio: Skip un-parseable mixer units instead of erroring
by Mark Hills 16 Apr '12

16 Apr '12

Some interfaces reference endpoints which do not exists. To accomodate these, do not fail completely, but skip over them. This allows the Electrix Ebox-44 with earlier firmware to be detected and used for audio. Signed-off-by: Mark Hills <mark(a)pogo.org.uk> --- sound/usb/mixer.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ab23869..c374c72 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1388,7 +1388,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *r for (pin = 0; pin < input_pins; pin++) { err = parse_audio_unit(state, desc->baSourceID[pin]); if (err < 0) - return err; + continue; err = check_input_term(state, desc->baSourceID[pin], &iterm); if (err < 0) return err; -- 1.7.4.4

3 5

[alsa-devel] [RFC/PATCH] control_external: Add ability to specify TLV data.
by Dylan Reid 16 Apr '12

16 Apr '12

Allow external control plugins to provide TLV data. This allows user-space pcms to specify dB ranges for controls. This follows the same model as the ALSA drivers for accessing the data. The code is based on that implementation. The control can provide static data or a callback. The data is accessed or modified in the new snd_ctl_ext_elem_tlv callback. Rev bump the protocol version to enable checking if an external control supports TLV. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- include/control_external.h | 17 +++++++++++++++- src/control/control_ext.c | 46 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 62 insertions(+), 1 deletions(-) diff --git a/include/control_external.h b/include/control_external.h index 7c066cf..5392ea6 100644 --- a/include/control_external.h +++ b/include/control_external.h @@ -60,13 +60,16 @@ typedef struct snd_ctl_ext snd_ctl_ext_t; typedef struct snd_ctl_ext_callback snd_ctl_ext_callback_t; /** Key to access a control pointer */ typedef unsigned long snd_ctl_ext_key_t; +/** Callback to handle TLV commands. */ +typedef int (snd_ctl_ext_tlv_rw_t)(snd_ctl_ext_t *ext, snd_ctl_ext_key_t key, int op_flag, unsigned int numid, + unsigned int *tlv, unsigned int tlv_size); /* * Protocol version */ #define SND_CTL_EXT_VERSION_MAJOR 1 /**< Protocol major version */ #define SND_CTL_EXT_VERSION_MINOR 0 /**< Protocol minor version */ -#define SND_CTL_EXT_VERSION_TINY 0 /**< Protocol tiny version */ +#define SND_CTL_EXT_VERSION_TINY 1 /**< Protocol tiny version */ /** * external plugin protocol version */ @@ -122,6 +125,13 @@ struct snd_ctl_ext { * control handle filled by #snd_ctl_ext_create() */ snd_ctl_t *handle; + /** + * optional TLV data for the control. + */ + union { + snd_ctl_ext_tlv_rw_t *c; + const unsigned int *p; + } tlv; int nonblock; /**< non-block mode; read-only */ int subscribed; /**< events subscribed; read-only */ @@ -245,7 +255,12 @@ typedef enum snd_ctl_ext_access { SND_CTL_EXT_ACCESS_WRITE = (1<<1), SND_CTL_EXT_ACCESS_READWRITE = (3<<0), SND_CTL_EXT_ACCESS_VOLATILE = (1<<2), + SND_CTL_EXT_ACCESS_TLV_READ = (1<<4), + SND_CTL_EXT_ACCESS_TLV_WRITE = (1<<5), + SND_CTL_EXT_ACCESS_TLV_READWRITE = (3<<4), + SND_CTL_EXT_ACCESS_TLV_COMMAND = (1<<6), SND_CTL_EXT_ACCESS_INACTIVE = (1<<8), + SND_CTL_EXT_ACCESS_TLV_CALLBACK = (1<<28), } snd_ctl_ext_access_t; /** diff --git a/src/control/control_ext.c b/src/control/control_ext.c index e20d4f3..0a0b397 100644 --- a/src/control/control_ext.c +++ b/src/control/control_ext.c @@ -324,6 +324,51 @@ static int snd_ctl_ext_elem_unlock(snd_ctl_t *handle ATTRIBUTE_UNUSED, return -ENXIO; } +static int snd_ctl_ext_elem_tlv(snd_ctl_t *handle, int op_flag, + unsigned int numid, + unsigned int *tlv, unsigned int tlv_size) +{ + snd_ctl_ext_t *ext = handle->private_data; + snd_ctl_ext_key_t key; + int type, ret; + unsigned int access, count, len; + snd_ctl_elem_id_t id; + + /* we don't support TLV on protocol ver 1.0.0 or earlier */ + if (ext->version < SNDRV_PROTOCOL_VERSION(1, 0, 0)) + return -ENXIO; + + snd_ctl_elem_id_clear(&id); + if (numid > 0) { + ext->callback->elem_list(ext, numid - 1, &id); + id.numid = numid; + } else + id.numid = 0; + key = ext->callback->find_elem(ext, &id); + + if (key == SND_CTL_EXT_KEY_NOT_FOUND) + return -ENOENT; + ret = ext->callback->get_attribute(ext, key, &type, &access, &count); + if (ret < 0) + return ret; + + if ((op_flag == 0 && (access & SND_CTL_EXT_ACCESS_TLV_READ) == 0) || + (op_flag > 0 && (access & SND_CTL_EXT_ACCESS_TLV_WRITE) == 0) || + (op_flag < 0 && (access & SND_CTL_EXT_ACCESS_TLV_COMMAND) == 0)) + return -ENXIO; + if (access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + return ext->tlv.c(ext, key, op_flag, numid, tlv, tlv_size); + } else { + if (op_flag) + return -ENXIO; + len = ext->tlv.p[1] + 2 * sizeof(unsigned int); + if (tlv_size < len) + return -ENOMEM; + memcpy(tlv, ext->tlv.p, len); + return 0; + } +} + static int snd_ctl_ext_next_device(snd_ctl_t *handle ATTRIBUTE_UNUSED, int *device ATTRIBUTE_UNUSED) { @@ -429,6 +474,7 @@ static const snd_ctl_ops_t snd_ctl_ext_ops = { .element_write = snd_ctl_ext_elem_write, .element_lock = snd_ctl_ext_elem_lock, .element_unlock = snd_ctl_ext_elem_unlock, + .element_tlv = snd_ctl_ext_elem_tlv, .hwdep_next_device = snd_ctl_ext_next_device, .hwdep_info = snd_ctl_ext_hwdep_info, .pcm_next_device = snd_ctl_ext_next_device, -- 1.7.9.rc0

2 2

[alsa-devel] [patch] ASoC: dapm: release lock on error paths
by Dan Carpenter 15 Apr '12

15 Apr '12

We added locking here but there were a couple error paths where we forgot to drop the lock before returning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 82e99b6..3191d01 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2133,7 +2133,7 @@ err: int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num) { - int i, ret; + int i, ret = 0; mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { @@ -2141,13 +2141,13 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, if (ret < 0) { dev_err(dapm->dev, "Failed to add route %s->%s\n", route->source, route->sink); - return ret; + break; } route++; } mutex_unlock(&dapm->card->dapm_mutex); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); @@ -2851,6 +2851,7 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w; int i; + int ret = 0; mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { @@ -2859,12 +2860,13 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, dev_err(dapm->dev, "ASoC: Failed to create DAPM control %s\n", widget->name); - return -ENOMEM; + ret = -ENOMEM; + break; } widget++; } mutex_unlock(&dapm->card->dapm_mutex); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);

2 1

[alsa-devel] No sound on Quanta KN1 with kernel 3.4
by Uros Vampl 14 Apr '12

14 Apr '12

Hi, as I wrote on the -users list, I don't have sound anymore in kernel 3.4. I was told to post the output of alsa-info.sh here, so here I am. The requested output is attached. Regards, Uroš

2 1

[alsa-devel] [PATCH] alsa-lib conf: block S/PDIF access for Sennheiser USB headset
by David Henningsson 14 Apr '12

14 Apr '12

The Sennheiser USB headset has no S/PDIF input or output, so make sure we can't open it that way. 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 966172f..b9148e7 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -38,6 +38,7 @@ USB-Audio.pcm.iec958_device { # "NoiseBlaster 3000" 42 # The below don't have digital in/out, so prevent them from being opened. + "Sennheiser USB headset" 999 "Logitech USB Headset" 999 "USB Device 0x46d:0x992" 999 "Blue Snowball" 999 -- 1.7.9.5

2 1

[alsa-devel] [PATCH V2 1/5] ARM: tegra: provide clock aliases for AHUB configlink
by Stephen Warren 13 Apr '12

13 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> The Tegra30 AHUB driver must call tegra_periph_reset_deassert() for all devices on the AHUB's configlink bus. The AHUB driver must be able to call clk_get_sys() to retrieve the clock parameter for this function. Add the necessary clock aliases to allow this. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- arch/arm/mach-tegra/tegra30_clocks.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-tegra/tegra30_clocks.c b/arch/arm/mach-tegra/tegra30_clocks.c index 6d08b53..e33fe4b 100644 --- a/arch/arm/mach-tegra/tegra30_clocks.c +++ b/arch/arm/mach-tegra/tegra30_clocks.c @@ -3015,6 +3015,15 @@ struct clk_duplicate tegra_clk_duplicates[] = { CLK_DUPLICATE("sbc6", "spi_slave_tegra.5", NULL), CLK_DUPLICATE("twd", "smp_twd", NULL), CLK_DUPLICATE("vcp", "nvavp", "vcp"), + CLK_DUPLICATE("i2s0", NULL, "i2s0"), + CLK_DUPLICATE("i2s1", NULL, "i2s1"), + CLK_DUPLICATE("i2s2", NULL, "i2s2"), + CLK_DUPLICATE("i2s3", NULL, "i2s3"), + CLK_DUPLICATE("i2s4", NULL, "i2s4"), + CLK_DUPLICATE("dam0", NULL, "dam0"), + CLK_DUPLICATE("dam1", NULL, "dam1"), + CLK_DUPLICATE("dam2", NULL, "dam2"), + CLK_DUPLICATE("spdif_in", NULL, "spdif_in"), }; struct clk *tegra_ptr_clks[] = { -- 1.7.0.4

2 7

[alsa-devel] [PATCH 1/2] ASoC: wm1250-ev1: Support stereo
by Mark Brown 13 Apr '12

13 Apr '12

Springbank can support stereo, though it is primarily intended for mono use cases. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm1250-ev1.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index e7a599a..0766b5f 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -84,14 +84,14 @@ static struct snd_soc_dai_driver wm1250_ev1_dai = { .playback = { .stream_name = "Playback", .channels_min = 1, - .channels_max = 1, + .channels_max = 2, .rates = SNDRV_PCM_RATE_8000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, - .channels_max = 1, + .channels_max = 2, .rates = SNDRV_PCM_RATE_8000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, -- 1.7.9.1

1 1

[alsa-devel] A few more codecs for hda-emu
by Claudio Matsuoka 13 Apr '12

13 Apr '12

Hi Takashi, I imported a few AD, IDT and VIA codec descriptions from codecgraph into the hda-emu codec collection. These bare bare codec info files instead of alsa-info output and can be replaced when better information becomes available. I can provide individual patches but you'll probably find it more convenient just to pull from git://github.com/cmatsuoka/hda-emu.git. Claudio

2 1

Re: [alsa-devel] Thinkpad hda regression in 3.3
by Takashi Iwai 13 Apr '12

13 Apr '12

At Fri, 13 Apr 2012 14:08:14 -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? Right. This must match with the actual values. Or the recent driver accepts 0 as the the second value. > 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. In that case, check whether the reported machines have the corresponding PCI SSIDs. thanks, Takashi

1 0

[alsa-devel] [PATCH V2 REBASE 1/4] ASoC: tegra: select REGMAP_MMIO
by Stephen Warren 13 Apr '12

13 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> All Tegra ASoC drivers will be reworked to use MMIO regmaps. Select this in Kconfig. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Split into separate patch --- sound/soc/tegra/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 441c317..5331c61 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,6 +1,7 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA + select REGMAP_MMIO help Say Y or M here if you want support for SoC audio on Tegra. -- 1.7.0.4

2 4

[alsa-devel] [PATCH V2 1/4] ASoC: tegra: select REGMAP_MMIO
by Stephen Warren 13 Apr '12

13 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> All Tegra ASoC drivers will be reworked to use MMIO regmaps. Select this in Kconfig. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Split into separate patch --- sound/soc/tegra/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 441c317..5331c61 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,6 +1,7 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA + select REGMAP_MMIO help Say Y or M here if you want support for SoC audio on Tegra. -- 1.7.0.4

2 5

[alsa-devel] [PATCH V2 1/3] ASoC: tegra: add tegra30-ahub driver
by Stephen Warren 13 Apr '12

13 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> The AHUB (Audio Hub) is a mux/crossbar which links all audio-related devices except the HDA controller on Tegra30. The devices include the DMA FIFOs, DAM (Digital Audio Mixers), I2S controllers, and SPDIF controller. Audio data may be routed between these devices in various combinations as required by board design/application. Includes a squashed bugfix from Nikesh Oswal <noswal(a)nvidia.com> Includes squashed bugfixes from Sumit Bhattacharya <sumitb(a)nvidia.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This depends on: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap.git topic/stride This addresses all the concerns previously raised, except: * The AHUB driver is still not a CODEC; unless I'm missing something, I don't believe the core ASoC infra-structure is in place to do that yet. * The AHUB device is not explicitly mentioned in the nvidia,tegra-audio-wm8903 device tree node. You'd argued it would be a good idea to represent the AHUB there to avoid future churn to the machine drivers. However, I think there will be churn for other reasons in the machine driver and possibly a different top-level dt binding required to start representing things generically, so adding the AHUB now wouldn't avoid any future churn. v2: * s/EXPORT_SYMBOL/EXPORT_SYMBOL_GPL/. Fix comment that stated direct I2S<->I2S connections were unlikely. * Explicitly treat the AHUB as a bus. Invoke of_platform_populate() to create child devices. * Remove tegra30_ahub_is_probed() function and related logic. * Correct APBIF reg size in DT binding example * Added runtime PM support * Document CIF IDs for use in child device bindings .../bindings/sound/nvidia,tegra30-ahub.txt | 32 + sound/soc/tegra/tegra30_ahub.c | 631 ++++++++++++++++++++ sound/soc/tegra/tegra30_ahub.h | 483 +++++++++++++++ 3 files changed, 1146 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt create mode 100644 sound/soc/tegra/tegra30_ahub.c create mode 100644 sound/soc/tegra/tegra30_ahub.h diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt new file mode 100644 index 0000000..1ac7b16 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt @@ -0,0 +1,32 @@ +NVIDIA Tegra30 AHUB (Audio Hub) + +Required properties: +- compatible : "nvidia,tegra30-ahub" +- reg : Should contain the register physical address and length for each of + the AHUB's APBIF registers and the AHUB's own registers. +- interrupts : Should contain AHUB interrupt +- nvidia,dma-request-selector : The Tegra DMA controller's phandle and + request selector for the first APBIF channel. +- ranges : The bus address mapping for the configlink register bus. + Can be empty since the mapping is 1:1. +- #address-cells : For the configlink bus. Should be <1>; +- #size-cells : For the configlink bus. Should be <1>. + +AHUB client modules need to specify the IDs of their CIFs (Client InterFaces). +For RX CIFs, the numbers indicate the register number within AHUB routing +register space (APBIF 0..3 RX, I2S 0..5 RX, DAM 0..2 RX 0..1, SPDIF RX 0..1). +For TX CIFs, the numbers indicate the bit position within the AHUB routing +registers (APBIF 0..3 TX, I2S 0..5 TX, DAM 0..2 TX, SPDIF TX 0..1). + +Example: + +ahub@70080000 { + compatible = "nvidia,tegra30-ahub"; + reg = <0x70080000 0x200 0x70080200 0x100>; + interrupts = < 0 103 0x04 >; + nvidia,dma-request-selector = <&apbdma 1>; + + ranges; + #address-cells = <1>; + #size-cells = <1>; +}; diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c new file mode 100644 index 0000000..57cd419 --- /dev/null +++ b/sound/soc/tegra/tegra30_ahub.c @@ -0,0 +1,631 @@ +/* + * tegra30_ahub.c - Tegra30 AHUB driver + * + * Copyright (c) 2011,2012, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, 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 General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <mach/clk.h> +#include <mach/dma.h> +#include <sound/soc.h> +#include "tegra30_ahub.h" + +#define DRV_NAME "tegra30-ahub" + +static struct tegra30_ahub *ahub; + +static inline void tegra30_apbif_write(u32 reg, u32 val) +{ + regmap_write(ahub->regmap_apbif, reg, val); +} + +static inline u32 tegra30_apbif_read(u32 reg) +{ + u32 val; + regmap_read(ahub->regmap_apbif, reg, &val); + return val; +} + +static inline void tegra30_audio_write(u32 reg, u32 val) +{ + regmap_write(ahub->regmap_ahub, reg, val); +} + +static int tegra30_ahub_runtime_suspend(struct device *dev) +{ + regcache_cache_only(ahub->regmap_apbif, true); + regcache_cache_only(ahub->regmap_ahub, true); + + clk_disable(ahub->clk_apbif); + clk_disable(ahub->clk_d_audio); + + return 0; +} + +/* + * clk_apbif isn't required for an I2S<->I2S configuration where no PCM data + * is read from or sent to memory. However, that's not something the rest of + * the driver supports right now, so we'll just treat the two clocks as one + * for now. + * + * These functions should not be a plain ref-count. Instead, each active stream + * contributes some requirement to the minimum clock rate, so starting or + * stopping streams should dynamically adjust the clock as required. However, + * this is not yet implemented. + */ +static int tegra30_ahub_runtime_resume(struct device *dev) +{ + int ret; + + ret = clk_enable(ahub->clk_d_audio); + if (ret) { + dev_err(dev, "clk_enable d_audio failed: %d\n", ret); + return ret; + } + ret = clk_enable(ahub->clk_apbif); + if (ret) { + dev_err(dev, "clk_enable apbif failed: %d\n", ret); + clk_disable(ahub->clk_d_audio); + return ret; + } + + regcache_cache_only(ahub->regmap_apbif, false); + regcache_cache_only(ahub->regmap_ahub, false); + + return 0; +} + +int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, + unsigned long *fiforeg, + unsigned long *reqsel) +{ + int channel; + u32 reg, val; + + channel = find_first_zero_bit(ahub->rx_usage, + TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + if (channel >= TEGRA30_AHUB_CHANNEL_CTRL_COUNT) + return -EBUSY; + + __set_bit(channel, ahub->rx_usage); + + *rxcif = TEGRA30_AHUB_RXCIF_APBIF_RX0 + channel; + *fiforeg = ahub->apbif_addr + TEGRA30_AHUB_CHANNEL_RXFIFO + + (channel * TEGRA30_AHUB_CHANNEL_RXFIFO_STRIDE); + *reqsel = ahub->dma_sel + channel; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~(TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK); + val |= (7 << TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT) | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_EN | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16; + tegra30_apbif_write(reg, val); + + reg = TEGRA30_AHUB_CIF_RX_CTRL + + (channel * TEGRA30_AHUB_CIF_RX_CTRL_STRIDE); + val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_allocate_rx_fifo); + +int tegra30_ahub_enable_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val |= TEGRA30_AHUB_CHANNEL_CTRL_RX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_enable_rx_fifo); + +int tegra30_ahub_disable_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~TEGRA30_AHUB_CHANNEL_CTRL_RX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_disable_rx_fifo); + +int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + + __clear_bit(channel, ahub->rx_usage); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_free_rx_fifo); + +int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, + unsigned long *fiforeg, + unsigned long *reqsel) +{ + int channel; + u32 reg, val; + + channel = find_first_zero_bit(ahub->tx_usage, + TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + if (channel >= TEGRA30_AHUB_CHANNEL_CTRL_COUNT) + return -EBUSY; + + __set_bit(channel, ahub->tx_usage); + + *txcif = TEGRA30_AHUB_TXCIF_APBIF_TX0 + channel; + *fiforeg = ahub->apbif_addr + TEGRA30_AHUB_CHANNEL_TXFIFO + + (channel * TEGRA30_AHUB_CHANNEL_TXFIFO_STRIDE); + *reqsel = ahub->dma_sel + channel; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~(TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK); + val |= (7 << TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT) | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_EN | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16; + tegra30_apbif_write(reg, val); + + reg = TEGRA30_AHUB_CIF_TX_CTRL + + (channel * TEGRA30_AHUB_CIF_TX_CTRL_STRIDE); + val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_allocate_tx_fifo); + +int tegra30_ahub_enable_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val |= TEGRA30_AHUB_CHANNEL_CTRL_TX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_enable_tx_fifo); + +int tegra30_ahub_disable_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~TEGRA30_AHUB_CHANNEL_CTRL_TX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_disable_tx_fifo); + +int tegra30_ahub_free_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + + __clear_bit(channel, ahub->tx_usage); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_free_tx_fifo); + +int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, + enum tegra30_ahub_txcif txcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg; + + reg = TEGRA30_AHUB_AUDIO_RX + + (channel * TEGRA30_AHUB_AUDIO_RX_STRIDE); + tegra30_audio_write(reg, 1 << txcif); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_set_rx_cif_source); + +int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg; + + reg = TEGRA30_AHUB_AUDIO_RX + + (channel * TEGRA30_AHUB_AUDIO_RX_STRIDE); + tegra30_audio_write(reg, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source); + +static const char * const configlink_clocks[] __devinitconst = { + "i2s0", + "i2s1", + "i2s2", + "i2s3", + "i2s4", + "dam0", + "dam1", + "dam2", + "spdif_in", +}; + +struct of_dev_auxdata ahub_auxdata[] __devinitdata = { + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080300, "tegra30-i2s.0", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080400, "tegra30-i2s.1", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080500, "tegra30-i2s.2", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080600, "tegra30-i2s.3", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080700, "tegra30-i2s.4", NULL), + {} +}; + +#define LAST_REG(name) \ + (TEGRA30_AHUB_##name + \ + (TEGRA30_AHUB_##name##_STRIDE * TEGRA30_AHUB_##name##_COUNT) - 4) + +#define REG_IN_ARRAY(reg, name) \ + ((reg >= TEGRA30_AHUB_##name) && \ + (reg <= LAST_REG(name) && \ + (!((reg - TEGRA30_AHUB_##name) % TEGRA30_AHUB_##name##_STRIDE)))) + +static bool tegra30_ahub_apbif_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA30_AHUB_CONFIG_LINK_CTRL: + case TEGRA30_AHUB_MISC_CTRL: + case TEGRA30_AHUB_APBDMA_LIVE_STATUS: + case TEGRA30_AHUB_I2S_LIVE_STATUS: + case TEGRA30_AHUB_SPDIF_LIVE_STATUS: + case TEGRA30_AHUB_I2S_INT_MASK: + case TEGRA30_AHUB_DAM_INT_MASK: + case TEGRA30_AHUB_SPDIF_INT_MASK: + case TEGRA30_AHUB_APBIF_INT_MASK: + case TEGRA30_AHUB_I2S_INT_STATUS: + case TEGRA30_AHUB_DAM_INT_STATUS: + case TEGRA30_AHUB_SPDIF_INT_STATUS: + case TEGRA30_AHUB_APBIF_INT_STATUS: + case TEGRA30_AHUB_I2S_INT_SOURCE: + case TEGRA30_AHUB_DAM_INT_SOURCE: + case TEGRA30_AHUB_SPDIF_INT_SOURCE: + case TEGRA30_AHUB_APBIF_INT_SOURCE: + case TEGRA30_AHUB_I2S_INT_SET: + case TEGRA30_AHUB_DAM_INT_SET: + case TEGRA30_AHUB_SPDIF_INT_SET: + case TEGRA30_AHUB_APBIF_INT_SET: + return true; + default: + break; + }; + + if (REG_IN_ARRAY(reg, CHANNEL_CTRL) || + REG_IN_ARRAY(reg, CHANNEL_CLEAR) || + REG_IN_ARRAY(reg, CHANNEL_STATUS) || + REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO) || + REG_IN_ARRAY(reg, CIF_TX_CTRL) || + REG_IN_ARRAY(reg, CIF_RX_CTRL) || + REG_IN_ARRAY(reg, DAM_LIVE_STATUS)) + return true; + + return false; +} + +static bool tegra30_ahub_apbif_volatile_reg(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case TEGRA30_AHUB_CONFIG_LINK_CTRL: + case TEGRA30_AHUB_MISC_CTRL: + case TEGRA30_AHUB_APBDMA_LIVE_STATUS: + case TEGRA30_AHUB_I2S_LIVE_STATUS: + case TEGRA30_AHUB_SPDIF_LIVE_STATUS: + case TEGRA30_AHUB_I2S_INT_STATUS: + case TEGRA30_AHUB_DAM_INT_STATUS: + case TEGRA30_AHUB_SPDIF_INT_STATUS: + case TEGRA30_AHUB_APBIF_INT_STATUS: + case TEGRA30_AHUB_I2S_INT_SET: + case TEGRA30_AHUB_DAM_INT_SET: + case TEGRA30_AHUB_SPDIF_INT_SET: + case TEGRA30_AHUB_APBIF_INT_SET: + return true; + default: + break; + }; + + if (REG_IN_ARRAY(reg, CHANNEL_CLEAR) || + REG_IN_ARRAY(reg, CHANNEL_STATUS) || + REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO) || + REG_IN_ARRAY(reg, DAM_LIVE_STATUS)) + return true; + + return false; +} + +static bool tegra30_ahub_apbif_precious_reg(struct device *dev, + unsigned int reg) +{ + if (REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO)) + return true; + + return false; +} + +static const struct regmap_config tegra30_ahub_apbif_regmap_config = { + .name = "apbif", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TEGRA30_AHUB_APBIF_INT_SET, + .writeable_reg = tegra30_ahub_apbif_wr_rd_reg, + .readable_reg = tegra30_ahub_apbif_wr_rd_reg, + .volatile_reg = tegra30_ahub_apbif_volatile_reg, + .precious_reg = tegra30_ahub_apbif_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg) +{ + if (REG_IN_ARRAY(reg, AUDIO_RX)) + return true; + + return false; +} + +static const struct regmap_config tegra30_ahub_ahub_regmap_config = { + .name = "ahub", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = LAST_REG(AUDIO_RX), + .writeable_reg = tegra30_ahub_ahub_wr_rd_reg, + .readable_reg = tegra30_ahub_ahub_wr_rd_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit tegra30_ahub_probe(struct platform_device *pdev) +{ + struct clk *clk; + int i; + struct resource *res0, *res1, *region; + u32 of_dma[2]; + void __iomem *regs_apbif, *regs_ahub; + int ret = 0; + + if (ahub) + return -ENODEV; + + /* + * The AHUB hosts a register bus: the "configlink". For this to + * operate correctly, all devices on this bus must be out of reset. + * Ensure that here. + */ + for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) { + clk = clk_get_sys(NULL, configlink_clocks[i]); + if (IS_ERR(clk)) { + dev_err(&pdev->dev, "Can't get clock %s\n", + configlink_clocks[i]); + ret = PTR_ERR(clk); + goto err; + } + tegra_periph_reset_deassert(clk); + clk_put(clk); + } + + ahub = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_ahub), + GFP_KERNEL); + if (!ahub) { + dev_err(&pdev->dev, "Can't allocate tegra30_ahub\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, ahub); + + ahub->dev = &pdev->dev; + + ahub->clk_d_audio = clk_get(&pdev->dev, "d_audio"); + if (IS_ERR(ahub->clk_d_audio)) { + dev_err(&pdev->dev, "Can't retrieve ahub d_audio clock\n"); + ret = PTR_ERR(ahub->clk_d_audio); + goto err; + } + + ahub->clk_apbif = clk_get(&pdev->dev, "apbif"); + if (IS_ERR(ahub->clk_apbif)) { + dev_err(&pdev->dev, "Can't retrieve ahub apbif clock\n"); + ret = PTR_ERR(ahub->clk_apbif); + goto err_clk_put_d_audio; + } + + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + of_dma, 2) < 0) { + dev_err(&pdev->dev, + "Missing property nvidia,dma-request-selector\n"); + ret = -ENODEV; + goto err_clk_put_d_audio; + } + ahub->dma_sel = of_dma[1]; + + res0 = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res0) { + dev_err(&pdev->dev, "No apbif memory resource\n"); + ret = -ENODEV; + goto err_clk_put_apbif; + } + + region = devm_request_mem_region(&pdev->dev, res0->start, + resource_size(res0), DRV_NAME); + if (!region) { + dev_err(&pdev->dev, "request region apbif failed\n"); + ret = -EBUSY; + goto err_clk_put_apbif; + } + ahub->apbif_addr = res0->start; + + regs_apbif = devm_ioremap(&pdev->dev, res0->start, + resource_size(res0)); + if (!regs_apbif) { + dev_err(&pdev->dev, "ioremap apbif failed\n"); + ret = -ENOMEM; + goto err_clk_put_apbif; + } + + ahub->regmap_apbif = devm_regmap_init_mmio(&pdev->dev, regs_apbif, + &tegra30_ahub_apbif_regmap_config); + if (IS_ERR(ahub->regmap_apbif)) { + dev_err(&pdev->dev, "apbif regmap init failed\n"); + ret = PTR_ERR(ahub->regmap_apbif); + goto err_clk_put_apbif; + } + regcache_cache_only(ahub->regmap_apbif, true); + + res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!res1) { + dev_err(&pdev->dev, "No ahub memory resource\n"); + ret = -ENODEV; + goto err_clk_put_apbif; + } + + region = devm_request_mem_region(&pdev->dev, res1->start, + resource_size(res1), DRV_NAME); + if (!region) { + dev_err(&pdev->dev, "request region ahub failed\n"); + ret = -EBUSY; + goto err_clk_put_apbif; + } + + regs_ahub = devm_ioremap(&pdev->dev, res1->start, + resource_size(res1)); + if (!regs_ahub) { + dev_err(&pdev->dev, "ioremap ahub failed\n"); + ret = -ENOMEM; + goto err_clk_put_apbif; + } + + ahub->regmap_ahub = devm_regmap_init_mmio(&pdev->dev, regs_ahub, + &tegra30_ahub_ahub_regmap_config); + if (IS_ERR(ahub->regmap_ahub)) { + dev_err(&pdev->dev, "ahub regmap init failed\n"); + ret = PTR_ERR(ahub->regmap_ahub); + goto err_clk_put_apbif; + } + regcache_cache_only(ahub->regmap_ahub, true); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra30_ahub_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + of_platform_populate(pdev->dev.of_node, NULL, ahub_auxdata, + &pdev->dev); + + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_clk_put_apbif: + clk_put(ahub->clk_apbif); +err_clk_put_d_audio: + clk_put(ahub->clk_d_audio); + ahub = 0; +err: + return ret; +} + +static int __devexit tegra30_ahub_remove(struct platform_device *pdev) +{ + if (!ahub) + return -ENODEV; + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra30_ahub_runtime_suspend(&pdev->dev); + + clk_put(ahub->clk_apbif); + clk_put(ahub->clk_d_audio); + + ahub = 0; + + return 0; +} + +static const struct of_device_id tegra30_ahub_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra30-ahub", }, + {}, +}; + +static const struct dev_pm_ops tegra30_ahub_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra30_ahub_runtime_suspend, + tegra30_ahub_runtime_resume, NULL) +}; + +static struct platform_driver tegra30_ahub_driver = { + .probe = tegra30_ahub_probe, + .remove = __devexit_p(tegra30_ahub_remove), + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra30_ahub_of_match, + .pm = &tegra30_ahub_pm_ops, + }, +}; +module_platform_driver(tegra30_ahub_driver); + +MODULE_AUTHOR("Stephen Warren <swarren(a)nvidia.com>"); +MODULE_DESCRIPTION("Tegra30 AHUB driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h new file mode 100644 index 0000000..e690e2e --- /dev/null +++ b/sound/soc/tegra/tegra30_ahub.h @@ -0,0 +1,483 @@ +/* + * tegra30_ahub.h - Definitions for Tegra30 AHUB driver + * + * Copyright (c) 2011,2012, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, 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 General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#ifndef __TEGRA30_AHUB_H__ +#define __TEGRA30_AHUB_H__ + +/* Fields in *_CIF_RX/TX_CTRL; used by AHUB FIFOs, and all other audio modules */ + +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT 28 +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US 0xf +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK (TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) + +/* Channel count minus 1 */ +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT 24 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US 7 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) + +/* Channel count minus 1 */ +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT 16 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US 7 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) + +#define TEGRA30_AUDIOCIF_BITS_4 0 +#define TEGRA30_AUDIOCIF_BITS_8 1 +#define TEGRA30_AUDIOCIF_BITS_12 2 +#define TEGRA30_AUDIOCIF_BITS_16 3 +#define TEGRA30_AUDIOCIF_BITS_20 4 +#define TEGRA30_AUDIOCIF_BITS_24 5 +#define TEGRA30_AUDIOCIF_BITS_28 6 +#define TEGRA30_AUDIOCIF_BITS_32 7 + +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT 12 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_MASK (7 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_4 (TEGRA30_AUDIOCIF_BITS_4 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_8 (TEGRA30_AUDIOCIF_BITS_8 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_12 (TEGRA30_AUDIOCIF_BITS_12 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 (TEGRA30_AUDIOCIF_BITS_16 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_20 (TEGRA30_AUDIOCIF_BITS_20 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_24 (TEGRA30_AUDIOCIF_BITS_24 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_28 (TEGRA30_AUDIOCIF_BITS_28 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_32 (TEGRA30_AUDIOCIF_BITS_32 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) + +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT 8 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_MASK (7 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_4 (TEGRA30_AUDIOCIF_BITS_4 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_8 (TEGRA30_AUDIOCIF_BITS_8 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_12 (TEGRA30_AUDIOCIF_BITS_12 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 (TEGRA30_AUDIOCIF_BITS_16 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_20 (TEGRA30_AUDIOCIF_BITS_20 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_24 (TEGRA30_AUDIOCIF_BITS_24 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_28 (TEGRA30_AUDIOCIF_BITS_28 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_32 (TEGRA30_AUDIOCIF_BITS_32 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) + +#define TEGRA30_AUDIOCIF_EXPAND_ZERO 0 +#define TEGRA30_AUDIOCIF_EXPAND_ONE 1 +#define TEGRA30_AUDIOCIF_EXPAND_LFSR 2 + +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT 6 +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_MASK (3 << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_ZERO (TEGRA30_AUDIOCIF_EXPAND_ZERO << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_ONE (TEGRA30_AUDIOCIF_EXPAND_ONE << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_LFSR (TEGRA30_AUDIOCIF_EXPAND_LFSR << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) + +#define TEGRA30_AUDIOCIF_STEREO_CONV_CH0 0 +#define TEGRA30_AUDIOCIF_STEREO_CONV_CH1 1 +#define TEGRA30_AUDIOCIF_STEREO_CONV_AVG 2 + +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT 4 +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_MASK (3 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH0 (TEGRA30_AUDIOCIF_STEREO_CONV_CH0 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH1 (TEGRA30_AUDIOCIF_STEREO_CONV_CH1 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_AVG (TEGRA30_AUDIOCIF_STEREO_CONV_AVG << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) + +#define TEGRA30_AUDIOCIF_CTRL_REPLICATE 3 + +#define TEGRA30_AUDIOCIF_DIRECTION_TX 0 +#define TEGRA30_AUDIOCIF_DIRECTION_RX 1 + +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT 2 +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_MASK (1 << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX (TEGRA30_AUDIOCIF_DIRECTION_TX << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX (TEGRA30_AUDIOCIF_DIRECTION_RX << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) + +#define TEGRA30_AUDIOCIF_TRUNCATE_ROUND 0 +#define TEGRA30_AUDIOCIF_TRUNCATE_CHOP 1 + +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT 1 +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_MASK (1 << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_ROUND (TEGRA30_AUDIOCIF_TRUNCATE_ROUND << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_CHOP (TEGRA30_AUDIOCIF_TRUNCATE_CHOP << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) + +#define TEGRA30_AUDIOCIF_MONO_CONV_ZERO 0 +#define TEGRA30_AUDIOCIF_MONO_CONV_COPY 1 + +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT 0 +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_MASK (1 << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_ZERO (TEGRA30_AUDIOCIF_MONO_CONV_ZERO << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_COPY (TEGRA30_AUDIOCIF_MONO_CONV_COPY << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) + +/* Registers within TEGRA30_AUDIO_CLUSTER_BASE */ + +/* TEGRA30_AHUB_CHANNEL_CTRL */ + +#define TEGRA30_AHUB_CHANNEL_CTRL 0x0 +#define TEGRA30_AHUB_CHANNEL_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_CTRL_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_EN (1 << 31) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_EN (1 << 30) +#define TEGRA30_AHUB_CHANNEL_CTRL_LOOPBACK (1 << 29) + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT 16 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK (TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT 8 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK (TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_EN (1 << 6) + +#define TEGRA30_PACK_8_4 2 +#define TEGRA30_PACK_16 3 + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT 4 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK_US 3 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK (TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_8_4 (TEGRA30_PACK_8_4 << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16 (TEGRA30_PACK_16 << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_EN (1 << 2) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT 0 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK_US 3 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK (TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_8_4 (TEGRA30_PACK_8_4 << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16 (TEGRA30_PACK_16 << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) + +/* TEGRA30_AHUB_CHANNEL_CLEAR */ + +#define TEGRA30_AHUB_CHANNEL_CLEAR 0x4 +#define TEGRA30_AHUB_CHANNEL_CLEAR_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_CLEAR_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_CLEAR_TX_SOFT_RESET (1 << 31) +#define TEGRA30_AHUB_CHANNEL_CLEAR_RX_SOFT_RESET (1 << 30) + +/* TEGRA30_AHUB_CHANNEL_STATUS */ + +#define TEGRA30_AHUB_CHANNEL_STATUS 0x8 +#define TEGRA30_AHUB_CHANNEL_STATUS_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_STATUS_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_SHIFT 24 +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK (TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK_US << TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_SHIFT) +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_SHIFT 16 +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK (TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK_US << TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_SHIFT) +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_TRIG (1 << 1) +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_TRIG (1 << 0) + +/* TEGRA30_AHUB_CHANNEL_TXFIFO */ + +#define TEGRA30_AHUB_CHANNEL_TXFIFO 0xc +#define TEGRA30_AHUB_CHANNEL_TXFIFO_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_TXFIFO_COUNT 4 + +/* TEGRA30_AHUB_CHANNEL_RXFIFO */ + +#define TEGRA30_AHUB_CHANNEL_RXFIFO 0x10 +#define TEGRA30_AHUB_CHANNEL_RXFIFO_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_RXFIFO_COUNT 4 + +/* TEGRA30_AHUB_CIF_TX_CTRL */ + +#define TEGRA30_AHUB_CIF_TX_CTRL 0x14 +#define TEGRA30_AHUB_CIF_TX_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CIF_TX_CTRL_COUNT 4 +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* */ + +/* TEGRA30_AHUB_CIF_RX_CTRL */ + +#define TEGRA30_AHUB_CIF_RX_CTRL 0x18 +#define TEGRA30_AHUB_CIF_RX_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CIF_RX_CTRL_COUNT 4 +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* */ + +/* TEGRA30_AHUB_CONFIG_LINK_CTRL */ + +#define TEGRA30_AHUB_CONFIG_LINK_CTRL 0x80 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_SHIFT 28 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK_US 0xf +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_SHIFT 16 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK_US 0xfff +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_SHIFT 4 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK_US 0xfff +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_CG_EN (1 << 2) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_CLEAR_TIMEOUT_CNTR (1 << 1) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_SOFT_RESET (1 << 0) + +/* TEGRA30_AHUB_MISC_CTRL */ + +#define TEGRA30_AHUB_MISC_CTRL 0x84 +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_ACTIVE (1 << 31) +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_CG_EN (1 << 8) +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_SHIFT 0 +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_MASK (0x1f << TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_SHIFT) + +/* TEGRA30_AHUB_APBDMA_LIVE_STATUS */ + +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS 0x88 +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_CIF_FIFO_FULL (1 << 31) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_CIF_FIFO_FULL (1 << 30) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_CIF_FIFO_FULL (1 << 29) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_CIF_FIFO_FULL (1 << 28) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_CIF_FIFO_FULL (1 << 27) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_CIF_FIFO_FULL (1 << 26) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_CIF_FIFO_FULL (1 << 25) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_CIF_FIFO_FULL (1 << 24) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_CIF_FIFO_EMPTY (1 << 23) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_CIF_FIFO_EMPTY (1 << 22) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_CIF_FIFO_EMPTY (1 << 21) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_CIF_FIFO_EMPTY (1 << 20) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_CIF_FIFO_EMPTY (1 << 19) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_CIF_FIFO_EMPTY (1 << 18) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_CIF_FIFO_EMPTY (1 << 17) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_CIF_FIFO_EMPTY (1 << 16) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_DMA_FIFO_FULL (1 << 15) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_DMA_FIFO_FULL (1 << 14) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_DMA_FIFO_FULL (1 << 13) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_DMA_FIFO_FULL (1 << 12) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_DMA_FIFO_FULL (1 << 11) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_DMA_FIFO_FULL (1 << 10) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_DMA_FIFO_FULL (1 << 9) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_DMA_FIFO_FULL (1 << 8) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_DMA_FIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_DMA_FIFO_EMPTY (1 << 6) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_DMA_FIFO_EMPTY (1 << 5) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_DMA_FIFO_EMPTY (1 << 4) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_DMA_FIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_DMA_FIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_DMA_FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_DMA_FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_I2S_LIVE_STATUS */ + +#define TEGRA30_AHUB_I2S_LIVE_STATUS 0x8c +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_FULL (1 << 29) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_FULL (1 << 28) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_FULL (1 << 27) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_FULL (1 << 26) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_FULL (1 << 25) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_FULL (1 << 24) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_FULL (1 << 23) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_FULL (1 << 22) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_FULL (1 << 21) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_FULL (1 << 20) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_ENABLED (1 << 19) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_ENABLED (1 << 18) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_ENABLED (1 << 17) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_ENABLED (1 << 16) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_ENABLED (1 << 15) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_ENABLED (1 << 14) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_ENABLED (1 << 13) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_ENABLED (1 << 12) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_ENABLED (1 << 11) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_ENABLED (1 << 10) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_EMPTY (1 << 9) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_EMPTY (1 << 8) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_EMPTY (1 << 6) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_EMPTY (1 << 5) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_EMPTY (1 << 4) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_DAM0_LIVE_STATUS */ + +#define TEGRA30_AHUB_DAM_LIVE_STATUS 0x90 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_STRIDE 0x8 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_COUNT 3 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TX_ENABLED (1 << 26) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1_ENABLED (1 << 25) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0_ENABLED (1 << 24) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TXFIFO_FULL (1 << 15) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1FIFO_FULL (1 << 9) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0FIFO_FULL (1 << 8) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TXFIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_SPDIF_LIVE_STATUS */ + +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS 0xa8 +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TX_ENABLED (1 << 11) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RX_ENABLED (1 << 10) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TX_ENABLED (1 << 9) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RX_ENABLED (1 << 8) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TXFIFO_FULL (1 << 7) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RXFIFO_FULL (1 << 6) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TXFIFO_FULL (1 << 5) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RXFIFO_FULL (1 << 4) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TXFIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RXFIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TXFIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RXFIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_I2S_INT_MASK */ + +#define TEGRA30_AHUB_I2S_INT_MASK 0xb0 + +/* TEGRA30_AHUB_DAM_INT_MASK */ + +#define TEGRA30_AHUB_DAM_INT_MASK 0xb4 + +/* TEGRA30_AHUB_SPDIF_INT_MASK */ + +#define TEGRA30_AHUB_SPDIF_INT_MASK 0xbc + +/* TEGRA30_AHUB_APBIF_INT_MASK */ + +#define TEGRA30_AHUB_APBIF_INT_MASK 0xc0 + +/* TEGRA30_AHUB_I2S_INT_STATUS */ + +#define TEGRA30_AHUB_I2S_INT_STATUS 0xc8 + +/* TEGRA30_AHUB_DAM_INT_STATUS */ + +#define TEGRA30_AHUB_DAM_INT_STATUS 0xcc + +/* TEGRA30_AHUB_SPDIF_INT_STATUS */ + +#define TEGRA30_AHUB_SPDIF_INT_STATUS 0xd4 + +/* TEGRA30_AHUB_APBIF_INT_STATUS */ + +#define TEGRA30_AHUB_APBIF_INT_STATUS 0xd8 + +/* TEGRA30_AHUB_I2S_INT_SOURCE */ + +#define TEGRA30_AHUB_I2S_INT_SOURCE 0xe0 + +/* TEGRA30_AHUB_DAM_INT_SOURCE */ + +#define TEGRA30_AHUB_DAM_INT_SOURCE 0xe4 + +/* TEGRA30_AHUB_SPDIF_INT_SOURCE */ + +#define TEGRA30_AHUB_SPDIF_INT_SOURCE 0xec + +/* TEGRA30_AHUB_APBIF_INT_SOURCE */ + +#define TEGRA30_AHUB_APBIF_INT_SOURCE 0xf0 + +/* TEGRA30_AHUB_I2S_INT_SET */ + +#define TEGRA30_AHUB_I2S_INT_SET 0xf8 + +/* TEGRA30_AHUB_DAM_INT_SET */ + +#define TEGRA30_AHUB_DAM_INT_SET 0xfc + +/* TEGRA30_AHUB_SPDIF_INT_SET */ + +#define TEGRA30_AHUB_SPDIF_INT_SET 0x100 + +/* TEGRA30_AHUB_APBIF_INT_SET */ + +#define TEGRA30_AHUB_APBIF_INT_SET 0x104 + +/* Registers within TEGRA30_AHUB_BASE */ + +#define TEGRA30_AHUB_AUDIO_RX 0x0 +#define TEGRA30_AHUB_AUDIO_RX_STRIDE 0x4 +#define TEGRA30_AHUB_AUDIO_RX_COUNT 17 +/* This register repeats once for each entry in enum tegra30_ahub_rxcif */ +/* The fields in this register are 1 bit per entry in tegra30_ahub_txcif */ + +/* + * Terminology: + * AHUB: Audio Hub; a cross-bar switch between the audio devices: DMA FIFOs, + * I2S controllers, SPDIF controllers, and DAMs. + * XBAR: The core cross-bar component of the AHUB. + * CIF: Client Interface; the HW module connecting an audio device to the + * XBAR. + * DAM: Digital Audio Mixer: A HW module that mixes multiple audio streams, + * possibly including sample-rate conversion. + * + * Each TX CIF transmits data into the XBAR. Each RX CIF can receive audio + * transmitted by a particular TX CIF. + * + * This driver is currently very simplistic; many HW features are not + * exposed; DAMs are not supported, only 16-bit stereo audio is supported, + * etc. + */ + +enum tegra30_ahub_txcif { + TEGRA30_AHUB_TXCIF_APBIF_TX0, + TEGRA30_AHUB_TXCIF_APBIF_TX1, + TEGRA30_AHUB_TXCIF_APBIF_TX2, + TEGRA30_AHUB_TXCIF_APBIF_TX3, + TEGRA30_AHUB_TXCIF_I2S0_TX0, + TEGRA30_AHUB_TXCIF_I2S1_TX0, + TEGRA30_AHUB_TXCIF_I2S2_TX0, + TEGRA30_AHUB_TXCIF_I2S3_TX0, + TEGRA30_AHUB_TXCIF_I2S4_TX0, + TEGRA30_AHUB_TXCIF_DAM0_TX0, + TEGRA30_AHUB_TXCIF_DAM1_TX0, + TEGRA30_AHUB_TXCIF_DAM2_TX0, + TEGRA30_AHUB_TXCIF_SPDIF_TX0, + TEGRA30_AHUB_TXCIF_SPDIF_TX1, +}; + +enum tegra30_ahub_rxcif { + TEGRA30_AHUB_RXCIF_APBIF_RX0, + TEGRA30_AHUB_RXCIF_APBIF_RX1, + TEGRA30_AHUB_RXcIF_APBIF_RX2, + TEGRA30_AHUB_RXCIF_APBIF_RX3, + TEGRA30_AHUB_RXCIF_I2S0_RX0, + TEGRA30_AHUB_RXCIF_I2S1_RX0, + TEGRA30_AHUB_RXCIF_I2S2_RX0, + TEGRA30_AHUB_RXCIF_I2S3_RX0, + TEGRA30_AHUB_RXCIF_I2S4_RX0, + TEGRA30_AHUB_RXCIF_DAM0_RX0, + TEGRA30_AHUB_RXCIF_DAM0_RX1, + TEGRA30_AHUB_RXCIF_DAM1_RX0, + TEGRA30_AHUB_RXCIF_DAM2_RX1, + TEGRA30_AHUB_RXCIF_DAM3_RX0, + TEGRA30_AHUB_RXCIF_DAM3_RX1, + TEGRA30_AHUB_RXCIF_SPDIF_RX0, + TEGRA30_AHUB_RXCIF_SPDIF_RX1, +}; + +extern int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, + unsigned long *fiforeg, + unsigned long *reqsel); +extern int tegra30_ahub_enable_rx_fifo(enum tegra30_ahub_rxcif rxcif); +extern int tegra30_ahub_disable_rx_fifo(enum tegra30_ahub_rxcif rxcif); +extern int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif); + +extern int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, + unsigned long *fiforeg, + unsigned long *reqsel); +extern int tegra30_ahub_enable_tx_fifo(enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_disable_tx_fifo(enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_free_tx_fifo(enum tegra30_ahub_txcif txcif); + +extern int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, + enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif); + +struct tegra30_ahub { + struct device *dev; + struct clk *clk_d_audio; + struct clk *clk_apbif; + int dma_sel; + resource_size_t apbif_addr; + struct regmap *regmap_apbif; + struct regmap *regmap_ahub; + DECLARE_BITMAP(rx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + DECLARE_BITMAP(tx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); +}; + +#endif -- 1.7.0.4

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Re: [alsa-devel] Thinkpad hda regression in 3.3
by Takashi Iwai 13 Apr '12

13 Apr '12

At Fri, 13 Apr 2012 08:43:06 -0400, Josh Boyer wrote: > > Hi Takashi and Jaroslav, > > We've had a number of bugs reported about sound not working correctly on > various models of Thinkpad machines in the 3.3 kernel. I've listed them > below. > > I looked through the commits and found: > > commit 29c5fbbcfefba5225a6783683c46c39e10877703: > > ALSA: HDA: Use model=auto for Thinkpad T510 > > The user reports that model=auto works fine for him. Using > model=auto bring in new features such as jack detection notification > to userspace. > > I did a local build here with that comment reverted and so far most of > the people reported their sound to be working again. The trivial revert > patch is below in case you want it. Could you get alsa-info.sh output on the non-working T510? If it's really about the built-in speaker or headphone jack, I doubt it's rahter the muted output in mixer. I see only "me too" in the bug report, which doesn't help much to diagnose the problem. I've got a few positive feedbacks of T410 and T510, so basically the driver should work. But I know the problem of docking-station (see below). > It seems there are multiple models that have PCI ID 17aa:215e and using > model=auto does not work for them all. It particularly messes with > machines using a docking station for some reason. > > Aside from just reverting the above commit, do you have other thoughts > on how to fix this? The problem regarding the docking station is basically a BIOS issue. ThinkPad BIOS doesn't expose the pins for docking stations at all. We have already a workaround for X200, and similar others would be needed for T410, T420, etc. For testing, you don't need even patch the driver. When the driver is built with CONFIG_SND_HDA_PATCH_LOADER=y, create a file containing the below as /lib/firmware/alsa-tp-dock: ================================================================ [codec] 0x14f15069 0x17aa21a4 0 [pincfg] 0x1a 0x21a11000 0x1c 0x23014250 ================================================================ then pass the module option patch="alsa-tp-dock" to snd-hda-intel, e.g. create a file containing below in /etc/modprobe.d/* options snd-hda-intel patch="alsa-tp-dock" This will override the pin-default values of 0x1a and 0x1c, which corresponds to the docking-station mic and docking-station output. This will set dock-out as the line-out so that the built-in headphone is preferred over it. Note that after applying this, the new mixer elements will appear, so you'd need to adjust the values manually once. Similarly, for T420, create a file with the contents below instead: ================================================================ [codec] 0x14f1506e 0x17aa21d2 0 [pincfg] 0x19 0x0421104f 0x1a 0x21a19050 0x1c 0x2121401f ================================================================ The rest procedure is same. 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. thanks, Takashi

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Re: [alsa-devel] Thinkpad hda regression in 3.3
by Takashi Iwai 13 Apr '12

13 Apr '12

At Fri, 13 Apr 2012 10:05:49 -0400, Josh Boyer wrote: > > On Fri, Apr 13, 2012 at 03:41:53PM +0200, Takashi Iwai wrote: > > At Fri, 13 Apr 2012 09:37:21 -0400, > > Josh Boyer wrote: > > Thanks. As mentionted, the difference between two setups (sent in two > > posts) is seen in the mixer setup. I personally don't mind which > > setup is taken; it's rather a choice of users. So, let me know which > > setup is preferred by testers. > > You're referring to the inclusion of pin 0x19 or not here, right? Yes. > > The reason behind such multiple setups is that the codec chip has only > > two DACs while you want to access for three or more outputs (two > > headphones, one speaker, etc). Thus some outputs must share the same > > DAC. > > I'm guessing when someone undocks the pins are reconfigured back to the > jacks on the laptop itself somehow? No, the configuration must be determined at the driver initialization time. So, the mixer elements stay unchanged with or without dock. (For example, if the mixer elements were to change upon the dock/undock, this would be really confusing both for users and apps. In theory, it's possible to implement like that, but no one wants it.) The jack-detection on the docking station works, yes, i.e. the auto-muting or the mic changing works. But it's different from changing the configuration on the hotplug of docking station itself. Takashi

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[alsa-devel] [GIT PULL] hda-jack-retask
by David Henningsson 13 Apr '12

13 Apr '12

It was a while since Takashi suggested I would try to upstream my hda-jack-retask program into alsa-tools, but with Ubuntu 12.04 into final freeze I finally got some headroom to do so. The program requires gtk3 to compile, and uses policykit (pkexec) as a run-time dependency to ask for root privileges where needed. I wasn't entirely sure where to document this in the best way, as the README file is primarily used for end-user documentation. Also, for the integration stuff with the rest of the programs (gitcompile, autotools etc), I have mostly copy-pasted what I found in the echomixer program. I also tested by modfying a debian package to do a test build with this new program included, and it seemed to install files in the right places, so I hope all is correct in that regard as well. Enjoy! The following changes since commit a680841d4a2a142981d2038e9af0a07abc74c5d7: alsa-tools: hwmixvolume - update README for au88x0 (2012-03-13 10:49:20 +0100) are available in the git repository at: git://kernel.ubuntu.com/diwic/alsa-tools.git hda-jack-retask David Henningsson (2): hdajackretask: Add new program for easy retask of hda pin nodes hdajackretask: Use autotools Makefile | 2 +- hdajackretask/AUTHORS | 1 + hdajackretask/ChangeLog | 1 + hdajackretask/Makefile.am | 20 ++ hdajackretask/NEWS | 1 + hdajackretask/README | 52 +++ hdajackretask/apply-changes.c | 285 +++++++++++++++++ hdajackretask/apply-changes.h | 15 + hdajackretask/configure.in | 10 + hdajackretask/gitcompile | 30 ++ hdajackretask/main-gtk.c | 523 +++++++++++++++++++++++++++++++ hdajackretask/sysfs-pin-configs.c | 626 +++++++++++++++++++++++++++++++++++++ hdajackretask/sysfs-pin-configs.h | 63 ++++ 13 files changed, 1628 insertions(+), 1 deletions(-) create mode 100644 hdajackretask/AUTHORS create mode 100644 hdajackretask/ChangeLog create mode 100644 hdajackretask/Makefile.am create mode 100644 hdajackretask/NEWS create mode 100644 hdajackretask/README create mode 100644 hdajackretask/apply-changes.c create mode 100644 hdajackretask/apply-changes.h create mode 100644 hdajackretask/configure.in create mode 100755 hdajackretask/gitcompile create mode 100644 hdajackretask/main-gtk.c create mode 100644 hdajackretask/sysfs-pin-configs.c create mode 100644 hdajackretask/sysfs-pin-configs.h -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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Re: [alsa-devel] Thinkpad hda regression in 3.3
by Takashi Iwai 13 Apr '12

13 Apr '12

At Fri, 13 Apr 2012 09:37:21 -0400, Josh Boyer wrote: > > On Fri, Apr 13, 2012 at 03:29:41PM +0200, Takashi Iwai wrote: > > At Fri, 13 Apr 2012 09:23:52 -0400, > > Josh Boyer wrote: > > > > The problem regarding the docking station is basically a BIOS issue. > > > > ThinkPad BIOS doesn't expose the pins for docking stations at all. > > > > We have already a workaround for X200, and similar others would be > > > > needed for T410, T420, etc. > > > > > > > > For testing, you don't need even patch the driver. When the driver is > > > > built with CONFIG_SND_HDA_PATCH_LOADER=y, create a file containing the > > > > below as /lib/firmware/alsa-tp-dock: > > > > > > > > ================================================================ > > > > [codec] > > > > 0x14f15069 0x17aa21a4 0 > > > > > > > > [pincfg] > > > > 0x1a 0x21a11000 > > > > 0x1c 0x23014250 > > > > ================================================================ > > > > > > The above is for the X200 docking station, correct? So it should work > > > with an x201 thinkpad in an x200 dock. > > > > It's for T410/T510 dock. But X201 uses the same codec chip, so this > > could be used, too. Just give it a try. > > OK. I've asked a reporter to test this out. We'll get back to you. Thanks. As mentionted, the difference between two setups (sent in two posts) is seen in the mixer setup. I personally don't mind which setup is taken; it's rather a choice of users. So, let me know which setup is preferred by testers. The reason behind such multiple setups is that the codec chip has only two DACs while you want to access for three or more outputs (two headphones, one speaker, etc). Thus some outputs must share the same DAC. Takashi

1 0

Re: [alsa-devel] Thinkpad hda regression in 3.3
by Takashi Iwai 13 Apr '12

13 Apr '12

At Fri, 13 Apr 2012 09:23:52 -0400, Josh Boyer wrote: > > On Fri, Apr 13, 2012 at 03:07:07PM +0200, Takashi Iwai wrote: > > At Fri, 13 Apr 2012 08:43:06 -0400, > > Josh Boyer wrote: > > > > > > Hi Takashi and Jaroslav, > > > > > > We've had a number of bugs reported about sound not working correctly on > > > various models of Thinkpad machines in the 3.3 kernel. I've listed them > > > below. > > > > > > I looked through the commits and found: > > > > > > commit 29c5fbbcfefba5225a6783683c46c39e10877703: > > > > > > ALSA: HDA: Use model=auto for Thinkpad T510 > > > > > > The user reports that model=auto works fine for him. Using > > > model=auto bring in new features such as jack detection notification > > > to userspace. > > > > > > I did a local build here with that comment reverted and so far most of > > > the people reported their sound to be working again. The trivial revert > > > patch is below in case you want it. > > > > Could you get alsa-info.sh output on the non-working T510? > > If it's really about the built-in speaker or headphone jack, I doubt > > it's rahter the muted output in mixer. I see only "me too" in the bug > > report, which doesn't help much to diagnose the problem. > > OK, I've asked for that. > > > I've got a few positive feedbacks of T410 and T510, so basically the > > driver should work. But I know the problem of docking-station (see > > below). > > > > > > > It seems there are multiple models that have PCI ID 17aa:215e and using > > > model=auto does not work for them all. It particularly messes with > > > machines using a docking station for some reason. > > > > > > Aside from just reverting the above commit, do you have other thoughts > > > on how to fix this? > > > > The problem regarding the docking station is basically a BIOS issue. > > ThinkPad BIOS doesn't expose the pins for docking stations at all. > > We have already a workaround for X200, and similar others would be > > needed for T410, T420, etc. > > > > For testing, you don't need even patch the driver. When the driver is > > built with CONFIG_SND_HDA_PATCH_LOADER=y, create a file containing the > > below as /lib/firmware/alsa-tp-dock: > > > > ================================================================ > > [codec] > > 0x14f15069 0x17aa21a4 0 > > > > [pincfg] > > 0x1a 0x21a11000 > > 0x1c 0x23014250 > > ================================================================ > > The above is for the X200 docking station, correct? So it should work > with an x201 thinkpad in an x200 dock. It's for T410/T510 dock. But X201 uses the same codec chip, so this could be used, too. Just give it a try. Takashi

1 0

[alsa-devel] [PATCH] ASoC: cs42l73: don't use negative array index
by Jesper Juhl 13 Apr '12

13 Apr '12

If cs42l73_get_mclkx_coeff() returns < 0 (which it can) in sound/soc/codecs/cs42l73.c::cs42l73_set_mclk(), then we'll be using the (negative) return value as array index on the very next line of code - that's bad. Catch the negative return value and propagate it to the caller (which checks for it) and things are a bit more sane :-) Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/soc/codecs/cs42l73.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 78979b3..07c44b7 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -929,6 +929,8 @@ static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) /* MCLKX -> MCLK */ mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / cs42l73_mclkx_coeffs[mclkx_coeff].ratio; -- 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] ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace).
by Jesper Juhl 13 Apr '12

13 Apr '12

In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the 'for (;;)' loop, if the 'badness' value returned from fill_and_eval_dacs() is negative, then we'll return from the function without freeing the memory we allocated for 'best_cfg', thus leaking. Fix the leak by kfree()'ing the memory when badness is negative. While I was there I also noticed some trailing whitespace in the function that I removed (along with all other trailing whitespace in the file) - it didn't seem worth-while to do that as two patches, so I hope it's OK that I just did it all as one patch. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/pci/hda/patch_realtek.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9917e55..c5e1a1a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3398,8 +3398,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) for (;;) { badness = fill_and_eval_dacs(codec, fill_hardwired, fill_mio_first); - if (badness < 0) + if (badness < 0) { + kfree(best_cfg); return badness; + } debug_badness("==> lo_type=%d, wired=%d, mio=%d, badness=0x%x\n", cfg->line_out_type, fill_hardwired, fill_mio_first, badness); @@ -3434,7 +3436,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; fill_hardwired = true; continue; - } + } if (cfg->hp_outs > 0 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { cfg->speaker_outs = cfg->line_outs; @@ -3448,7 +3450,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->line_out_type = AUTO_PIN_HP_OUT; fill_hardwired = true; continue; - } + } break; } @@ -4423,7 +4425,7 @@ static int alc_parse_auto_config(struct hda_codec *codec, static int alc880_parse_auto_config(struct hda_codec *codec) { static const hda_nid_t alc880_ignore[] = { 0x1d, 0 }; - static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; + static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids); } @@ -6079,7 +6081,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding * fixup entry. - */ + */ SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC), @@ -6296,7 +6298,7 @@ static void alc_fixup_no_jack_detect(struct hda_codec *codec, { if (action == ALC_FIXUP_ACT_PRE_PROBE) codec->no_jack_detect = 1; -} +} static const struct alc_fixup alc861_fixups[] = { [ALC861_FIXUP_FSC_AMILO_PI1505] = { @@ -6714,7 +6716,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding * fixup entry. - */ + */ SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC662_FIXUP_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC662_FIXUP_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC662_FIXUP_ASUS_MODE1), -- 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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Re: [alsa-devel] [PATCH] ALSA: hda/realtek - Add quirk for Mac Pro 5, 1 machines
by Takashi Iwai 12 Apr '12

12 Apr '12

At Thu, 12 Apr 2012 13:55:36 -0400, Josh Boyer wrote: > > A user reported that setting model=imac24 used to allow sound to work on their > Mac Pro 5,1 machine. Commit 5671087ffa "Move ALC885 macpro and imac24 models > to auto-parser" removed this model option. All Mac machines are now explicitly > handled with a quirk and the auto-parser. This adds a quirk for the device > found on the Mac Pro 5,1 machines. > > This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559 > > Reported-by: Gabriel Somlo <somlo(a)cmu.edu> > Signed-off-by: Josh Boyer <jwboyer(a)redhat.com> Applied now (with sorting in the id number order). Thanks. Takashi > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 9917e55..1e0eea8 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -5371,6 +5371,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { > SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC889_FIXUP_MBP_VREF), > SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), > SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_FIXUP_MACPRO_GPIO), > + SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), > SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_FIXUP_MACPRO_GPIO), > SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_FIXUP_MACPRO_GPIO), > SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF), > -- > 1.7.7.6 >

1 0

Re: [alsa-devel] [PATCH 0/6] snd-usb endpoint rework, version 6
by Felix Homann 12 Apr '12

12 Apr '12

Happy New Year! That being said, I've tested a kernel without Daniel's patch set. It doesn't work either. The error messages are only slightly different: Jan 1 11:52:31 weasel mtp-probe: checking bus 1, device 4: "/sys/devices/pci0000:00/0000:00:1a.7/usb1/1-1" Jan 1 11:52:31 weasel mtp-probe: bus: 1, device: 4 was not an MTP device Jan 1 11:52:31 weasel kernel: [ 161.880773] usbcore: registered new interface driver snd-usb-audio Jan 1 11:52:34 weasel kernel: [ 164.284980] cannot submit datapipe for urb 0, error -28: not enough bandwidth So, It's not related to Daniel's rework. But nevertheless, something's broken...

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[alsa-devel] High CPU usage when recording on TI DM6446 using ASoC AIC33
by Max Klein 12 Apr '12

12 Apr '12

I am having a problem with high CPU usage (>90%) on TI DM6446 (ARM926 core) with TI TLV320AIC33 audio codec. When using the command "gst-launch -v alsasrc ! mulawenc ! rtppcmupay ! fakesink silent=true" the CPU usage raises to 97% after a while (e.g. after 38s) and drops to 0% again later (e.g. after next 78s). To check if the alsasrc element of GStreamer is the problem, I used "arecord -r 8000 -c 1 -f S16_LE --period-time=10000 --buffer-time=200000 > /dev/null" for comparison, and I also could see a similar problem: after some (more) seconds the CPU usage raises to >90% and drops to 0% afterwards again (but it stays at >90% for only ca. 33s ). I just monitored the problem with top and vmstat - vmstat log for arecord example can be seen at http://pastebin.com/H3LRtyPg When using the gstreamer command (especially alsasrc), a different latency-time and buffer-time has influence to the cpu usage (e.g. when using "gst-launch -v alsasrc latency-time=20000 buffer-time=400000 ! mulawenc ! rtppcmupay ! fakesink silent=true" CPU usage raised to 49% after a while, when using "gst-launch -v alsasrc latency-time=40000 buffer-time=800000 ! mulawenc ! rtppcmupay ! fakesink silent=true" CPU usage raised to 25% after a while...) I guess it could be a problem somewhere in the ALSA driver, but before trying to patch the driver to the latest version I would go for sure that this would help...I have seen a lot of commits in AIC33 git tree, but I guess its not that easy to update my kernel to the latest alsa-kernel version??? 1. Can someone confirm such a behaviour on DM6446 EVM? 2. Does someone have a clue how to identify and solve the problem? 3. Would it be possible to update "just" the ALSA part in my Kernel do test the latest ASoC driver for DaVinci/AIC33? I am looking forward for any help! -- Information about used versions: > I am using DM6446 with TI AIC33 Audio Codec running at vendor-specific 2.6.27 kernel (based on http://gitorious.org/linux-davinci/linux-davinci/trees/v2.6.27-davinci1 with ALSA driver patched to cb6e2063697e91ca6983f9fe6958d20469b43641 from alsa-kernel [2008-11-18] tree) > Advanced Linux Sound Architecture Driver Version 1.0.17. > ASoC version 0.13.2 > AIC3X Audio Codec 0.2 > asoc: aic3x <-> davinci-i2s mapping ok > ALSA device list: > #0: DaVinci EVM (aic3x) > Version of GStreamer 0.10.25 (also tested with 0.10.35 - the same problem) > ALSA library in user space is 1.0.19 (extracted from libasound.so) -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

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[alsa-devel] [PATCH for 3.4] MFD: twl6040: Convert to i2c driver, and separate it from twl core
by Peter Ujfalusi 12 Apr '12

12 Apr '12

Complete the separation of the twl6040 from the twl core since it is a separate chip, not part of the twl6030 PMIC. Make the needed Kconfig changes for the depending drivers at the same time to avoid breaking the kernel build (vibra, ASoC components). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reviewed-by: Mark Brown <broonie(a)opensource.wolfsonicro.com> Acked-by: Tony Lindgren <tony(a)atomide.com> Acked-by: Samuel Ortiz <sameo(a)linux.intel.com> Acked-by: Dmitry Torokhov <dtor(a)mail.ru> --- Hi Samuel, Tony, Liam, Due to some cross tree issues 3.4-rc1 have big regression in OMAP4 audio: The twl6040 mfd driver (and codec, vibra) is no longer probe since the twl-core now did not create the platform device for it. This means we do not have audio working in rc1. This patch meant to follow the twl-core change (to not probe the twll6040 as platform device), but there were some glitch in the logistics so this patch is not in 3.4-rc1. I have forward ported it on top of 3.4-rc1, and you can also find the patch in: git@gitorious.org:omap-audio/linux-audio.git peter/upstream/for-3.4-rc2 Can someone take this patch further to be included in 3.4-rc2? Thank you, Peter arch/arm/mach-omap2/board-4430sdp.c | 12 ++-- arch/arm/mach-omap2/board-generic.c | 2 +- arch/arm/mach-omap2/board-omap4panda.c | 13 ++-- arch/arm/mach-omap2/twl-common.c | 37 +++++++++-- arch/arm/mach-omap2/twl-common.h | 10 +-- drivers/input/misc/Kconfig | 3 +- drivers/input/misc/twl6040-vibra.c | 4 +- drivers/mfd/Kconfig | 11 +++- drivers/mfd/twl6040-core.c | 114 +++++++++++++++++++------------- include/linux/i2c/twl.h | 12 ---- include/linux/mfd/twl6040.h | 27 ++++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/twl6040.c | 3 +- sound/soc/omap/Kconfig | 2 +- 14 files changed, 159 insertions(+), 94 deletions(-) diff --git a/arch/arm/mach-omap2/board-4430sdp.c b/arch/arm/mach-omap2/board-4430sdp.c index a39fc4b..130ab00 100644 --- a/arch/arm/mach-omap2/board-4430sdp.c +++ b/arch/arm/mach-omap2/board-4430sdp.c @@ -20,6 +20,7 @@ #include <linux/usb/otg.h> #include <linux/spi/spi.h> #include <linux/i2c/twl.h> +#include <linux/mfd/twl6040.h> #include <linux/gpio_keys.h> #include <linux/regulator/machine.h> #include <linux/regulator/fixed.h> @@ -560,7 +561,7 @@ static struct regulator_init_data sdp4430_vusim = { }, }; -static struct twl4030_codec_data twl6040_codec = { +static struct twl6040_codec_data twl6040_codec = { /* single-step ramp for headset and handsfree */ .hs_left_step = 0x0f, .hs_right_step = 0x0f, @@ -568,7 +569,7 @@ static struct twl4030_codec_data twl6040_codec = { .hf_right_step = 0x1d, }; -static struct twl4030_vibra_data twl6040_vibra = { +static struct twl6040_vibra_data twl6040_vibra = { .vibldrv_res = 8, .vibrdrv_res = 3, .viblmotor_res = 10, @@ -577,16 +578,14 @@ static struct twl4030_vibra_data twl6040_vibra = { .vddvibr_uV = 0, /* fixed volt supply - VBAT */ }; -static struct twl4030_audio_data twl6040_audio = { +static struct twl6040_platform_data twl6040_data = { .codec = &twl6040_codec, .vibra = &twl6040_vibra, .audpwron_gpio = 127, - .naudint_irq = OMAP44XX_IRQ_SYS_2N, .irq_base = TWL6040_CODEC_IRQ_BASE, }; static struct twl4030_platform_data sdp4430_twldata = { - .audio = &twl6040_audio, /* Regulators */ .vusim = &sdp4430_vusim, .vaux1 = &sdp4430_vaux1, @@ -617,7 +616,8 @@ static int __init omap4_i2c_init(void) TWL_COMMON_REGULATOR_VCXIO | TWL_COMMON_REGULATOR_VUSB | TWL_COMMON_REGULATOR_CLK32KG); - omap4_pmic_init("twl6030", &sdp4430_twldata); + omap4_pmic_init("twl6030", &sdp4430_twldata, + &twl6040_data, OMAP44XX_IRQ_SYS_2N); omap_register_i2c_bus(2, 400, NULL, 0); omap_register_i2c_bus(3, 400, sdp4430_i2c_3_boardinfo, ARRAY_SIZE(sdp4430_i2c_3_boardinfo)); diff --git a/arch/arm/mach-omap2/board-generic.c b/arch/arm/mach-omap2/board-generic.c index 74e1687..098d183 100644 --- a/arch/arm/mach-omap2/board-generic.c +++ b/arch/arm/mach-omap2/board-generic.c @@ -137,7 +137,7 @@ static struct twl4030_platform_data sdp4430_twldata = { static void __init omap4_i2c_init(void) { - omap4_pmic_init("twl6030", &sdp4430_twldata); + omap4_pmic_init("twl6030", &sdp4430_twldata, NULL, 0); } static void __init omap4_init(void) diff --git a/arch/arm/mach-omap2/board-omap4panda.c b/arch/arm/mach-omap2/board-omap4panda.c index d8c0e89..1b782ba 100644 --- a/arch/arm/mach-omap2/board-omap4panda.c +++ b/arch/arm/mach-omap2/board-omap4panda.c @@ -25,6 +25,7 @@ #include <linux/gpio.h> #include <linux/usb/otg.h> #include <linux/i2c/twl.h> +#include <linux/mfd/twl6040.h> #include <linux/regulator/machine.h> #include <linux/regulator/fixed.h> #include <linux/wl12xx.h> @@ -284,7 +285,7 @@ static int __init omap4_twl6030_hsmmc_init(struct omap2_hsmmc_info *controllers) return 0; } -static struct twl4030_codec_data twl6040_codec = { +static struct twl6040_codec_data twl6040_codec = { /* single-step ramp for headset and handsfree */ .hs_left_step = 0x0f, .hs_right_step = 0x0f, @@ -292,17 +293,14 @@ static struct twl4030_codec_data twl6040_codec = { .hf_right_step = 0x1d, }; -static struct twl4030_audio_data twl6040_audio = { +static struct twl6040_platform_data twl6040_data = { .codec = &twl6040_codec, .audpwron_gpio = 127, - .naudint_irq = OMAP44XX_IRQ_SYS_2N, .irq_base = TWL6040_CODEC_IRQ_BASE, }; /* Panda board uses the common PMIC configuration */ -static struct twl4030_platform_data omap4_panda_twldata = { - .audio = &twl6040_audio, -}; +static struct twl4030_platform_data omap4_panda_twldata; /* * Display monitor features are burnt in their EEPROM as EDID data. The EEPROM @@ -326,7 +324,8 @@ static int __init omap4_panda_i2c_init(void) TWL_COMMON_REGULATOR_VCXIO | TWL_COMMON_REGULATOR_VUSB | TWL_COMMON_REGULATOR_CLK32KG); - omap4_pmic_init("twl6030", &omap4_panda_twldata); + omap4_pmic_init("twl6030", &omap4_panda_twldata, + &twl6040_data, OMAP44XX_IRQ_SYS_2N); omap_register_i2c_bus(2, 400, NULL, 0); /* * Bus 3 is attached to the DVI port where devices like the pico DLP diff --git a/arch/arm/mach-omap2/twl-common.c b/arch/arm/mach-omap2/twl-common.c index 4b57757..7a7b893 100644 --- a/arch/arm/mach-omap2/twl-common.c +++ b/arch/arm/mach-omap2/twl-common.c @@ -37,6 +37,16 @@ static struct i2c_board_info __initdata pmic_i2c_board_info = { .flags = I2C_CLIENT_WAKE, }; +static struct i2c_board_info __initdata omap4_i2c1_board_info[] = { + { + .addr = 0x48, + .flags = I2C_CLIENT_WAKE, + }, + { + I2C_BOARD_INFO("twl6040", 0x4b), + }, +}; + void __init omap_pmic_init(int bus, u32 clkrate, const char *pmic_type, int pmic_irq, struct twl4030_platform_data *pmic_data) @@ -49,14 +59,31 @@ void __init omap_pmic_init(int bus, u32 clkrate, omap_register_i2c_bus(bus, clkrate, &pmic_i2c_board_info, 1); } +void __init omap4_pmic_init(const char *pmic_type, + struct twl4030_platform_data *pmic_data, + struct twl6040_platform_data *twl6040_data, int twl6040_irq) +{ + /* PMIC part*/ + strncpy(omap4_i2c1_board_info[0].type, pmic_type, + sizeof(omap4_i2c1_board_info[0].type)); + omap4_i2c1_board_info[0].irq = OMAP44XX_IRQ_SYS_1N; + omap4_i2c1_board_info[0].platform_data = pmic_data; + + /* TWL6040 audio IC part */ + omap4_i2c1_board_info[1].irq = twl6040_irq; + omap4_i2c1_board_info[1].platform_data = twl6040_data; + + omap_register_i2c_bus(1, 400, omap4_i2c1_board_info, 2); + +} + void __init omap_pmic_late_init(void) { /* Init the OMAP TWL parameters (if PMIC has been registerd) */ - if (!pmic_i2c_board_info.irq) - return; - - omap3_twl_init(); - omap4_twl_init(); + if (pmic_i2c_board_info.irq) + omap3_twl_init(); + if (omap4_i2c1_board_info[0].irq) + omap4_twl_init(); } #if defined(CONFIG_ARCH_OMAP3) diff --git a/arch/arm/mach-omap2/twl-common.h b/arch/arm/mach-omap2/twl-common.h index 275dde8..0962748 100644 --- a/arch/arm/mach-omap2/twl-common.h +++ b/arch/arm/mach-omap2/twl-common.h @@ -29,6 +29,7 @@ struct twl4030_platform_data; +struct twl6040_platform_data; void omap_pmic_init(int bus, u32 clkrate, const char *pmic_type, int pmic_irq, struct twl4030_platform_data *pmic_data); @@ -46,12 +47,9 @@ static inline void omap3_pmic_init(const char *pmic_type, omap_pmic_init(1, 2600, pmic_type, INT_34XX_SYS_NIRQ, pmic_data); } -static inline void omap4_pmic_init(const char *pmic_type, - struct twl4030_platform_data *pmic_data) -{ - /* Phoenix Audio IC needs I2C1 to start with 400 KHz or less */ - omap_pmic_init(1, 400, pmic_type, OMAP44XX_IRQ_SYS_1N, pmic_data); -} +void omap4_pmic_init(const char *pmic_type, + struct twl4030_platform_data *pmic_data, + struct twl6040_platform_data *audio_data, int twl6040_irq); void omap3_pmic_get_config(struct twl4030_platform_data *pmic_data, u32 pdata_flags, u32 regulators_flags); diff --git a/drivers/input/misc/Kconfig b/drivers/input/misc/Kconfig index 2d78779..7faf4a7 100644 --- a/drivers/input/misc/Kconfig +++ b/drivers/input/misc/Kconfig @@ -380,8 +380,7 @@ config INPUT_TWL4030_VIBRA config INPUT_TWL6040_VIBRA tristate "Support for TWL6040 Vibrator" - depends on TWL4030_CORE - select TWL6040_CORE + depends on TWL6040_CORE select INPUT_FF_MEMLESS help This option enables support for TWL6040 Vibrator Driver. diff --git a/drivers/input/misc/twl6040-vibra.c b/drivers/input/misc/twl6040-vibra.c index 45874fe..14e94f5 100644 --- a/drivers/input/misc/twl6040-vibra.c +++ b/drivers/input/misc/twl6040-vibra.c @@ -28,7 +28,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/workqueue.h> -#include <linux/i2c/twl.h> +#include <linux/input.h> #include <linux/mfd/twl6040.h> #include <linux/slab.h> #include <linux/delay.h> @@ -257,7 +257,7 @@ static SIMPLE_DEV_PM_OPS(twl6040_vibra_pm_ops, twl6040_vibra_suspend, NULL); static int __devinit twl6040_vibra_probe(struct platform_device *pdev) { - struct twl4030_vibra_data *pdata = pdev->dev.platform_data; + struct twl6040_vibra_data *pdata = pdev->dev.platform_data; struct vibra_info *info; int ret; diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 29f463c..11e4438 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -268,10 +268,17 @@ config TWL6030_PWM This is used to control charging LED brightness. config TWL6040_CORE - bool - depends on TWL4030_CORE && GENERIC_HARDIRQS + bool "Support for TWL6040 audio codec" + depends on I2C=y && GENERIC_HARDIRQS select MFD_CORE + select REGMAP_I2C default n + help + Say yes here if you want support for Texas Instruments TWL6040 audio + codec. + This driver provides common support for accessing the device, + additional drivers must be enabled in order to use the + functionality of the device (audio, vibra). config MFD_STMPE bool "Support STMicroelectronics STMPE" diff --git a/drivers/mfd/twl6040-core.c b/drivers/mfd/twl6040-core.c index b2d8e51..2d6beda 100644 --- a/drivers/mfd/twl6040-core.c +++ b/drivers/mfd/twl6040-core.c @@ -30,7 +30,9 @@ #include <linux/platform_device.h> #include <linux/gpio.h> #include <linux/delay.h> -#include <linux/i2c/twl.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/err.h> #include <linux/mfd/core.h> #include <linux/mfd/twl6040.h> @@ -39,7 +41,7 @@ int twl6040_reg_read(struct twl6040 *twl6040, unsigned int reg) { int ret; - u8 val = 0; + unsigned int val; mutex_lock(&twl6040->io_mutex); /* Vibra control registers from cache */ @@ -47,7 +49,7 @@ int twl6040_reg_read(struct twl6040 *twl6040, unsigned int reg) reg == TWL6040_REG_VIBCTLR)) { val = twl6040->vibra_ctrl_cache[VIBRACTRL_MEMBER(reg)]; } else { - ret = twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &val, reg); + ret = regmap_read(twl6040->regmap, reg, &val); if (ret < 0) { mutex_unlock(&twl6040->io_mutex); return ret; @@ -64,7 +66,7 @@ int twl6040_reg_write(struct twl6040 *twl6040, unsigned int reg, u8 val) int ret; mutex_lock(&twl6040->io_mutex); - ret = twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, val, reg); + ret = regmap_write(twl6040->regmap, reg, val); /* Cache the vibra control registers */ if (reg == TWL6040_REG_VIBCTLL || reg == TWL6040_REG_VIBCTLR) twl6040->vibra_ctrl_cache[VIBRACTRL_MEMBER(reg)] = val; @@ -77,16 +79,9 @@ EXPORT_SYMBOL(twl6040_reg_write); int twl6040_set_bits(struct twl6040 *twl6040, unsigned int reg, u8 mask) { int ret; - u8 val; mutex_lock(&twl6040->io_mutex); - ret = twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &val, reg); - if (ret) - goto out; - - val |= mask; - ret = twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, val, reg); -out: + ret = regmap_update_bits(twl6040->regmap, reg, mask, mask); mutex_unlock(&twl6040->io_mutex); return ret; } @@ -95,16 +90,9 @@ EXPORT_SYMBOL(twl6040_set_bits); int twl6040_clear_bits(struct twl6040 *twl6040, unsigned int reg, u8 mask) { int ret; - u8 val; mutex_lock(&twl6040->io_mutex); - ret = twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &val, reg); - if (ret) - goto out; - - val &= ~mask; - ret = twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, val, reg); -out: + ret = regmap_update_bits(twl6040->regmap, reg, mask, 0); mutex_unlock(&twl6040->io_mutex); return ret; } @@ -494,32 +482,58 @@ static struct resource twl6040_codec_rsrc[] = { }, }; -static int __devinit twl6040_probe(struct platform_device *pdev) +static bool twl6040_readable_reg(struct device *dev, unsigned int reg) { - struct twl4030_audio_data *pdata = pdev->dev.platform_data; + /* Register 0 is not readable */ + if (!reg) + return false; + return true; +} + +static struct regmap_config twl6040_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = TWL6040_REG_STATUS, /* 0x2e */ + + .readable_reg = twl6040_readable_reg, +}; + +static int __devinit twl6040_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct twl6040_platform_data *pdata = client->dev.platform_data; struct twl6040 *twl6040; struct mfd_cell *cell = NULL; int ret, children = 0; if (!pdata) { - dev_err(&pdev->dev, "Platform data is missing\n"); + dev_err(&client->dev, "Platform data is missing\n"); return -EINVAL; } /* In order to operate correctly we need valid interrupt config */ - if (!pdata->naudint_irq || !pdata->irq_base) { - dev_err(&pdev->dev, "Invalid IRQ configuration\n"); + if (!client->irq || !pdata->irq_base) { + dev_err(&client->dev, "Invalid IRQ configuration\n"); return -EINVAL; } - twl6040 = kzalloc(sizeof(struct twl6040), GFP_KERNEL); - if (!twl6040) - return -ENOMEM; + twl6040 = devm_kzalloc(&client->dev, sizeof(struct twl6040), + GFP_KERNEL); + if (!twl6040) { + ret = -ENOMEM; + goto err; + } + + twl6040->regmap = regmap_init_i2c(client, &twl6040_regmap_config); + if (IS_ERR(twl6040->regmap)) { + ret = PTR_ERR(twl6040->regmap); + goto err; + } - platform_set_drvdata(pdev, twl6040); + i2c_set_clientdata(client, twl6040); - twl6040->dev = &pdev->dev; - twl6040->irq = pdata->naudint_irq; + twl6040->dev = &client->dev; + twl6040->irq = client->irq; twl6040->irq_base = pdata->irq_base; mutex_init(&twl6040->mutex); @@ -588,12 +602,12 @@ static int __devinit twl6040_probe(struct platform_device *pdev) } if (children) { - ret = mfd_add_devices(&pdev->dev, pdev->id, twl6040->cells, + ret = mfd_add_devices(&client->dev, -1, twl6040->cells, children, NULL, 0); if (ret) goto mfd_err; } else { - dev_err(&pdev->dev, "No platform data found for children\n"); + dev_err(&client->dev, "No platform data found for children\n"); ret = -ENODEV; goto mfd_err; } @@ -608,14 +622,15 @@ gpio2_err: if (gpio_is_valid(twl6040->audpwron)) gpio_free(twl6040->audpwron); gpio1_err: - platform_set_drvdata(pdev, NULL); - kfree(twl6040); + i2c_set_clientdata(client, NULL); + regmap_exit(twl6040->regmap); +err: return ret; } -static int __devexit twl6040_remove(struct platform_device *pdev) +static int __devexit twl6040_remove(struct i2c_client *client) { - struct twl6040 *twl6040 = platform_get_drvdata(pdev); + struct twl6040 *twl6040 = i2c_get_clientdata(client); if (twl6040->power_count) twl6040_power(twl6040, 0); @@ -626,23 +641,30 @@ static int __devexit twl6040_remove(struct platform_device *pdev) free_irq(twl6040->irq_base + TWL6040_IRQ_READY, twl6040); twl6040_irq_exit(twl6040); - mfd_remove_devices(&pdev->dev); - platform_set_drvdata(pdev, NULL); - kfree(twl6040); + mfd_remove_devices(&client->dev); + i2c_set_clientdata(client, NULL); + regmap_exit(twl6040->regmap); return 0; } -static struct platform_driver twl6040_driver = { +static const struct i2c_device_id twl6040_i2c_id[] = { + { "twl6040", 0, }, + { }, +}; +MODULE_DEVICE_TABLE(i2c, twl6040_i2c_id); + +static struct i2c_driver twl6040_driver = { + .driver = { + .name = "twl6040", + .owner = THIS_MODULE, + }, .probe = twl6040_probe, .remove = __devexit_p(twl6040_remove), - .driver = { - .owner = THIS_MODULE, - .name = "twl6040", - }, + .id_table = twl6040_i2c_id, }; -module_platform_driver(twl6040_driver); +module_i2c_driver(twl6040_driver); MODULE_DESCRIPTION("TWL6040 MFD"); MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez(a)ti.com>"); diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index 2463b61..1f90de0 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -666,23 +666,11 @@ struct twl4030_codec_data { unsigned int check_defaults:1; unsigned int reset_registers:1; unsigned int hs_extmute:1; - u16 hs_left_step; - u16 hs_right_step; - u16 hf_left_step; - u16 hf_right_step; void (*set_hs_extmute)(int mute); }; struct twl4030_vibra_data { unsigned int coexist; - - /* twl6040 */ - unsigned int vibldrv_res; /* left driver resistance */ - unsigned int vibrdrv_res; /* right driver resistance */ - unsigned int viblmotor_res; /* left motor resistance */ - unsigned int vibrmotor_res; /* right motor resistance */ - int vddvibl_uV; /* VDDVIBL volt, set 0 for fixed reg */ - int vddvibr_uV; /* VDDVIBR volt, set 0 for fixed reg */ }; struct twl4030_audio_data { diff --git a/include/linux/mfd/twl6040.h b/include/linux/mfd/twl6040.h index 9bc9ac6..b15b5f0 100644 --- a/include/linux/mfd/twl6040.h +++ b/include/linux/mfd/twl6040.h @@ -174,8 +174,35 @@ #define TWL6040_SYSCLK_SEL_LPPLL 0 #define TWL6040_SYSCLK_SEL_HPPLL 1 +struct twl6040_codec_data { + u16 hs_left_step; + u16 hs_right_step; + u16 hf_left_step; + u16 hf_right_step; +}; + +struct twl6040_vibra_data { + unsigned int vibldrv_res; /* left driver resistance */ + unsigned int vibrdrv_res; /* right driver resistance */ + unsigned int viblmotor_res; /* left motor resistance */ + unsigned int vibrmotor_res; /* right motor resistance */ + int vddvibl_uV; /* VDDVIBL volt, set 0 for fixed reg */ + int vddvibr_uV; /* VDDVIBR volt, set 0 for fixed reg */ +}; + +struct twl6040_platform_data { + int audpwron_gpio; /* audio power-on gpio */ + unsigned int irq_base; + + struct twl6040_codec_data *codec; + struct twl6040_vibra_data *vibra; +}; + +struct regmap; + struct twl6040 { struct device *dev; + struct regmap *regmap; struct mutex mutex; struct mutex io_mutex; struct mutex irq_mutex; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6508e8b..59d8efa 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,7 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE - select SND_SOC_TWL6040 if TWL4030_CORE + select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WL1273 if MFD_WL1273_CORE @@ -276,7 +276,6 @@ config SND_SOC_TWL4030 tristate config SND_SOC_TWL6040 - select TWL6040_CORE tristate config SND_SOC_UDA134X diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2d8c6b8..dc7509b 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -26,7 +26,6 @@ #include <linux/pm.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <linux/i2c/twl.h> #include <linux/mfd/twl6040.h> #include <sound/core.h> @@ -1528,7 +1527,7 @@ static int twl6040_resume(struct snd_soc_codec *codec) static int twl6040_probe(struct snd_soc_codec *codec) { struct twl6040_data *priv; - struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev); + struct twl6040_codec_data *pdata = dev_get_platdata(codec->dev); struct platform_device *pdev = container_of(codec->dev, struct platform_device, dev); int ret = 0; diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index e00dd0b..deafbfa 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -97,7 +97,7 @@ config SND_OMAP_SOC_SDP3430 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL4030_CORE && SND_OMAP_SOC && ARCH_OMAP4 + depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4 select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 -- 1.7.8.5

1 1

[alsa-devel] HDA userspace
by Claudio 11 Apr '12

11 Apr '12

(Re-sending this since the first attempt got caught in the moderator queue.) Hi Takashi and all, It has been some time since my previous opportunity to play with ALSA, and I'm happy to be able to contribute again. I'll have some time allocated specifically for upstream work, so if you have any specific tasks that could use some help, please feel free to contact me. I'm still interested in HDA userpace and I see there were some significant developments since the last time I checked. Could someone point me to the development tree so I can run the current version and see how it works in my test machines? Best regards, Claudio

2 2

[alsa-devel] paravirtualized alsa kernel driver for XEN
by Stefano Panella 11 Apr '12

11 Apr '12

Hi, I am Stefano Panella, I am new to the list and I would like to take the opportunity to ask some questions since I am trying to write a paravirtualized alsa driver for XEN. If all goes well I would also like to upstream it on linux. I have been reading the documentation on "Writing an ALSA Driver" and I am still not completely clear on the meaning of the "pointer" callback in the pcm operations. The description say: "This callback is called when the PCM middle layer inquires the current hardware position on the buffer." My question are: 1) In case of a playback stream, is the pointer referring to wich sample is currently playing on the DAC or to which is it the last frame read by the HW from the alsa memory buffer? 2) What does the pointer mean in case of a capture stream? Is it the position of the current frame on the ADC or is the latest frame written into the alsa buffer? 3) in case it is the frame on the DAC/ADC, what happens if the callback does not return the real DAC/ADC frame position but an approximate value, let say rounded to 64 frames only? 4) is there any test I could run to check I have implemented correctly the "pointer" callback? Or any application which would need very high "pointer" precision like frame precision? Thanks very much in advance, Stefano

3 9

Re: [alsa-devel] [PATCH] Fix for Asus A6Vm sound
by Takashi Iwai 11 Apr '12

11 Apr '12

At Wed, 11 Apr 2012 16:34:37 +0200, Albert Pool wrote: > > I have been told that model=z71v is necessary for Asus A6Vm (1043:1173) sound to work. > > Signed-off-by: Albert Pool<albertpool(a)solcon.nl> It's very old quick code and was already dropped in the recent kernel. Did you verify that it really works for your device? If yes, please give alsa-info.sh output (run with --no-upload option). thanks, Takashi > > --- > > sound/pci/hda/alc880_quirks.c | 2 +- > > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c > > index 501501e..1b0358a 100644 > > --- a/sound/pci/hda/alc880_quirks.c > > +++ b/sound/pci/hda/alc880_quirks.c > > @@ -1375,7 +1375,7 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { > > SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG), > > SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG), > > SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG), > > - SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG), > > + SND_PCI_QUIRK(0x1043, 0x1173, "ASUS A6V", ALC880_Z71V), > > SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_Z71V), > > /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */ > > SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG), > > -- > > 1.7.5.4 >

1 0

[alsa-devel] [PATCH 1/2] ASoC: ac97: Fix build due to removal of 'runtime' definition
by Fabio Estevam 11 Apr '12

11 Apr '12

Fix the following build error: sound/soc/codecs/ac97.c: In function 'ac97_prepare': sound/soc/codecs/ac97.c:33: error: 'runtime' undeclared (first use in this function) This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers), which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/ac97.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index a99a1b3..2023c74 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -30,7 +30,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; - return snd_ac97_set_rate(codec->ac97, reg, runtime->rate); + return snd_ac97_set_rate(codec->ac97, reg, substream->runtime->rate); } #define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ -- 1.7.1

2 2

[alsa-devel] FriendlyArm mini2440 S3C2440 UDA1341 duplex support
by jarek 11 Apr '12

11 Apr '12

Hello! Can someone confirm, that this board has full-duplex audio support ? best regards Jarek

1 0

[alsa-devel] [PATCH v2] ASoC: wm9712: Fix build due to removal of 'runtime' definition
by Fabio Estevam 11 Apr '12

11 Apr '12

Fix the following build error: sound/soc/codecs/wm9712.c:482:32: error: 'runtime' undeclared (first use in this function) sound/soc/codecs/wm9712.c:499:33: error: 'runtime' undeclared (first use in this function) This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers), which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> sound/soc/codecs/wm9712.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2603863..eee8170 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -479,7 +479,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, else reg = AC97_PCM_LR_ADC_RATE; - return ac97_write(codec, reg, runtime->rate); + return ac97_write(codec, reg, substream->runtime->rate); } static int ac97_aux_prepare(struct snd_pcm_substream *substream, @@ -496,7 +496,8 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; - return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); + return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, + substream->runtime->rate); } #define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ -- 1.7.1

1 0

[alsa-devel] [PATCH] ASoC: tegra: convert to regmap
by Stephen Warren 11 Apr '12

11 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> For the Tegra I2S, SPDIF, and DAS drivers Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, this depends on the regmap->ASoC git pull request that I just sent. sound/soc/tegra/Kconfig | 1 + sound/soc/tegra/tegra20_das.c | 99 ++++++++-------------- sound/soc/tegra/tegra20_das.h | 3 +- sound/soc/tegra/tegra20_i2s.c | 149 ++++++++++++++++----------------- sound/soc/tegra/tegra20_i2s.h | 3 +- sound/soc/tegra/tegra20_spdif.c | 177 ++++++++++++++++++++------------------- sound/soc/tegra/tegra20_spdif.h | 3 +- 7 files changed, 203 insertions(+), 232 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 556cac2..d70fe94 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,6 +1,7 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA + select REGMAP_MMIO help Say Y or M here if you want support for SoC audio on Tegra. diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index 812696d..dacd5cf 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -20,12 +20,11 @@ * */ -#include <linux/debugfs.h> #include <linux/device.h> #include <linux/io.h> #include <linux/module.h> #include <linux/platform_device.h> -#include <linux/seq_file.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/soc.h> #include "tegra20_das.h" @@ -36,12 +35,14 @@ static struct tegra20_das *das; static inline void tegra20_das_write(u32 reg, u32 val) { - __raw_writel(val, das->regs + reg); + regmap_write(das->regmap, reg, val); } static inline u32 tegra20_das_read(u32 reg) { - return __raw_readl(das->regs + reg); + u32 val; + regmap_read(das->regmap, reg, &val); + return val; } int tegra20_das_connect_dap_to_dac(int dap, int dac) @@ -104,68 +105,38 @@ int tegra20_das_connect_dac_to_dap(int dac, int dap) } EXPORT_SYMBOL_GPL(tegra20_das_connect_dac_to_dap); -#ifdef CONFIG_DEBUG_FS -static int tegra20_das_show(struct seq_file *s, void *unused) -{ - int i; - u32 addr; - u32 reg; - - for (i = 0; i < TEGRA20_DAS_DAP_CTRL_SEL_COUNT; i++) { - addr = TEGRA20_DAS_DAP_CTRL_SEL + - (i * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE); - reg = tegra20_das_read(addr); - seq_printf(s, "TEGRA20_DAS_DAP_CTRL_SEL[%d] = %08x\n", i, reg); - } - - for (i = 0; i < TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT; i++) { - addr = TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL + - (i * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); - reg = tegra20_das_read(addr); - seq_printf(s, "TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL[%d] = %08x\n", - i, reg); - } - - return 0; -} - -static int tegra20_das_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra20_das_show, inode->i_private); -} +#define LAST_REG(name) \ + (TEGRA20_DAS_##name + \ + (TEGRA20_DAS_##name##_STRIDE * TEGRA20_DAS_##name##_COUNT) - 4) -static const struct file_operations tegra20_das_debug_fops = { - .open = tegra20_das_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; +#define REG_IN_ARRAY(reg, name) \ + ((reg >= TEGRA20_DAS_##name) && \ + (reg <= LAST_REG(name) && \ + (!((reg - TEGRA20_DAS_##name) % TEGRA20_DAS_##name##_STRIDE)))) -static void tegra20_das_debug_add(struct tegra20_das *das) +static bool tegra20_das_wr_rd_reg(struct device *dev, unsigned int reg) { - das->debug = debugfs_create_file(DRV_NAME, S_IRUGO, - snd_soc_debugfs_root, das, - &tegra20_das_debug_fops); -} + if (REG_IN_ARRAY(reg, DAP_CTRL_SEL) || + REG_IN_ARRAY(reg, DAC_INPUT_DATA_CLK_SEL)) + return true; -static void tegra20_das_debug_remove(struct tegra20_das *das) -{ - if (das->debug) - debugfs_remove(das->debug); -} -#else -static inline void tegra20_das_debug_add(struct tegra20_das *das) -{ + return false; } -static inline void tegra20_das_debug_remove(struct tegra20_das *das) -{ -} -#endif +static const struct regmap_config tegra20_das_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = LAST_REG(DAC_INPUT_DATA_CLK_SEL), + .writeable_reg = tegra20_das_wr_rd_reg, + .readable_reg = tegra20_das_wr_rd_reg, + .cache_type = REGCACHE_RBTREE, +}; static int __devinit tegra20_das_probe(struct platform_device *pdev) { struct resource *res, *region; + void __iomem *regs; int ret = 0; if (das) @@ -194,13 +165,21 @@ static int __devinit tegra20_das_probe(struct platform_device *pdev) goto err; } - das->regs = devm_ioremap(&pdev->dev, res->start, resource_size(res)); - if (!das->regs) { + regs = devm_ioremap(&pdev->dev, res->start, resource_size(res)); + if (!regs) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENOMEM; goto err; } + das->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra20_das_regmap_config); + if (IS_ERR(das->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(das->regmap); + goto err; + } + ret = tegra20_das_connect_dap_to_dac(TEGRA20_DAS_DAP_ID_1, TEGRA20_DAS_DAP_SEL_DAC1); if (ret) { @@ -214,8 +193,6 @@ static int __devinit tegra20_das_probe(struct platform_device *pdev) goto err; } - tegra20_das_debug_add(das); - platform_set_drvdata(pdev, das); return 0; @@ -230,8 +207,6 @@ static int __devexit tegra20_das_remove(struct platform_device *pdev) if (!das) return -ENODEV; - tegra20_das_debug_remove(das); - das = NULL; return 0; diff --git a/sound/soc/tegra/tegra20_das.h b/sound/soc/tegra/tegra20_das.h index ade4fe0..be217f3 100644 --- a/sound/soc/tegra/tegra20_das.h +++ b/sound/soc/tegra/tegra20_das.h @@ -85,8 +85,7 @@ struct tegra20_das { struct device *dev; - void __iomem *regs; - struct dentry *debug; + struct regmap *regmap; }; /* diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 6957658..0c7af63 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -29,14 +29,13 @@ */ #include <linux/clk.h> -#include <linux/debugfs.h> #include <linux/device.h> #include <linux/io.h> #include <linux/module.h> #include <linux/of.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> -#include <linux/seq_file.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -49,12 +48,14 @@ static inline void tegra20_i2s_write(struct tegra20_i2s *i2s, u32 reg, u32 val) { - __raw_writel(val, i2s->regs + reg); + regmap_write(i2s->regmap, reg, val); } static inline u32 tegra20_i2s_read(struct tegra20_i2s *i2s, u32 reg) { - return __raw_readl(i2s->regs + reg); + u32 val; + regmap_read(i2s->regmap, reg, &val); + return val; } static int tegra20_i2s_runtime_suspend(struct device *dev) @@ -80,74 +81,6 @@ static int tegra20_i2s_runtime_resume(struct device *dev) return 0; } -#ifdef CONFIG_DEBUG_FS -static int tegra20_i2s_show(struct seq_file *s, void *unused) -{ -#define REG(r) { r, #r } - static const struct { - int offset; - const char *name; - } regs[] = { - REG(TEGRA20_I2S_CTRL), - REG(TEGRA20_I2S_STATUS), - REG(TEGRA20_I2S_TIMING), - REG(TEGRA20_I2S_FIFO_SCR), - REG(TEGRA20_I2S_PCM_CTRL), - REG(TEGRA20_I2S_NW_CTRL), - REG(TEGRA20_I2S_TDM_CTRL), - REG(TEGRA20_I2S_TDM_TX_RX_CTRL), - }; -#undef REG - - struct tegra20_i2s *i2s = s->private; - int i; - - clk_enable(i2s->clk_i2s); - - for (i = 0; i < ARRAY_SIZE(regs); i++) { - u32 val = tegra20_i2s_read(i2s, regs[i].offset); - seq_printf(s, "%s = %08x\n", regs[i].name, val); - } - - clk_disable(i2s->clk_i2s); - - return 0; -} - -static int tegra20_i2s_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra20_i2s_show, inode->i_private); -} - -static const struct file_operations tegra20_i2s_debug_fops = { - .open = tegra20_i2s_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static void tegra20_i2s_debug_add(struct tegra20_i2s *i2s) -{ - i2s->debug = debugfs_create_file(i2s->dai.name, S_IRUGO, - snd_soc_debugfs_root, i2s, - &tegra20_i2s_debug_fops); -} - -static void tegra20_i2s_debug_remove(struct tegra20_i2s *i2s) -{ - if (i2s->debug) - debugfs_remove(i2s->debug); -} -#else -static inline void tegra20_i2s_debug_add(struct tegra_i2s *i2s) -{ -} - -static inline void tegra20_i2s_debug_remove(struct tegra20_i2s *i2s) -{ -} -#endif - static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { @@ -343,12 +276,68 @@ static const struct snd_soc_dai_driver tegra20_i2s_dai_template = { .symmetric_rates = 1, }; +static bool tegra20_i2s_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_I2S_CTRL: + case TEGRA20_I2S_STATUS: + case TEGRA20_I2S_TIMING: + case TEGRA20_I2S_FIFO_SCR: + case TEGRA20_I2S_PCM_CTRL: + case TEGRA20_I2S_NW_CTRL: + case TEGRA20_I2S_TDM_CTRL: + case TEGRA20_I2S_TDM_TX_RX_CTRL: + case TEGRA20_I2S_FIFO1: + case TEGRA20_I2S_FIFO2: + return true; + default: + return false; + }; +} + +static bool tegra20_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_I2S_STATUS: + case TEGRA20_I2S_FIFO_SCR: + case TEGRA20_I2S_FIFO1: + case TEGRA20_I2S_FIFO2: + return true; + default: + return false; + }; +} + +static bool tegra20_i2s_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_I2S_FIFO1: + case TEGRA20_I2S_FIFO2: + return true; + default: + return false; + }; +} + +static const struct regmap_config tegra20_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA20_I2S_FIFO2, + .writeable_reg = tegra20_i2s_wr_rd_reg, + .readable_reg = tegra20_i2s_wr_rd_reg, + .volatile_reg = tegra20_i2s_volatile_reg, + .precious_reg = tegra20_i2s_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; + static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev) { struct tegra20_i2s *i2s; struct resource *mem, *memregion, *dmareq; u32 of_dma[2]; u32 dma_ch; + void __iomem *regs; int ret; i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_i2s), GFP_KERNEL); @@ -398,13 +387,21 @@ static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev) goto err_clk_put; } - i2s->regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!i2s->regs) { + regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!regs) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENOMEM; goto err_clk_put; } + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra20_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(i2s->regmap); + goto err_clk_put; + } + i2s->capture_dma_data.addr = mem->start + TEGRA20_I2S_FIFO2; i2s->capture_dma_data.wrap = 4; i2s->capture_dma_data.width = 32; @@ -437,8 +434,6 @@ static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev) goto err_unregister_dai; } - tegra20_i2s_debug_add(i2s); - return 0; err_unregister_dai: @@ -465,8 +460,6 @@ static int __devexit tegra20_i2s_platform_remove(struct platform_device *pdev) tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); - tegra20_i2s_debug_remove(i2s); - clk_put(i2s->clk_i2s); return 0; diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h index 86ab327..a57efc6 100644 --- a/sound/soc/tegra/tegra20_i2s.h +++ b/sound/soc/tegra/tegra20_i2s.h @@ -157,8 +157,7 @@ struct tegra20_i2s { struct clk *clk_i2s; struct tegra_pcm_dma_params capture_dma_data; struct tegra_pcm_dma_params playback_dma_data; - void __iomem *regs; - struct dentry *debug; + struct regmap *regmap; u32 reg_ctrl; }; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 94c5764..f9b5741 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -21,13 +21,12 @@ */ #include <linux/clk.h> -#include <linux/debugfs.h> #include <linux/device.h> #include <linux/io.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> -#include <linux/seq_file.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -41,12 +40,14 @@ static inline void tegra20_spdif_write(struct tegra20_spdif *spdif, u32 reg, u32 val) { - __raw_writel(val, spdif->regs + reg); + regmap_write(spdif->regmap, reg, val); } static inline u32 tegra20_spdif_read(struct tegra20_spdif *spdif, u32 reg) { - return __raw_readl(spdif->regs + reg); + u32 val; + regmap_read(spdif->regmap, reg, &val); + return val; } static int tegra20_spdif_runtime_suspend(struct device *dev) @@ -72,82 +73,6 @@ static int tegra20_spdif_runtime_resume(struct device *dev) return 0; } -#ifdef CONFIG_DEBUG_FS -static int tegra20_spdif_show(struct seq_file *s, void *unused) -{ -#define REG(r) { r, #r } - static const struct { - int offset; - const char *name; - } regs[] = { - REG(TEGRA20_SPDIF_CTRL), - REG(TEGRA20_SPDIF_STATUS), - REG(TEGRA20_SPDIF_STROBE_CTRL), - REG(TEGRA20_SPDIF_DATA_FIFO_CSR), - REG(TEGRA20_SPDIF_CH_STA_RX_A), - REG(TEGRA20_SPDIF_CH_STA_RX_B), - REG(TEGRA20_SPDIF_CH_STA_RX_C), - REG(TEGRA20_SPDIF_CH_STA_RX_D), - REG(TEGRA20_SPDIF_CH_STA_RX_E), - REG(TEGRA20_SPDIF_CH_STA_RX_F), - REG(TEGRA20_SPDIF_CH_STA_TX_A), - REG(TEGRA20_SPDIF_CH_STA_TX_B), - REG(TEGRA20_SPDIF_CH_STA_TX_C), - REG(TEGRA20_SPDIF_CH_STA_TX_D), - REG(TEGRA20_SPDIF_CH_STA_TX_E), - REG(TEGRA20_SPDIF_CH_STA_TX_F), - }; -#undef REG - - struct tegra20_spdif *spdif = s->private; - int i; - - clk_enable(spdif->clk_spdif_out); - - for (i = 0; i < ARRAY_SIZE(regs); i++) { - u32 val = tegra20_spdif_read(spdif, regs[i].offset); - seq_printf(s, "%s = %08x\n", regs[i].name, val); - } - - clk_disable(spdif->clk_spdif_out); - - return 0; -} - -static int tegra20_spdif_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra20_spdif_show, inode->i_private); -} - -static const struct file_operations tegra20_spdif_debug_fops = { - .open = tegra20_spdif_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static void tegra20_spdif_debug_add(struct tegra20_spdif *spdif) -{ - spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO, - snd_soc_debugfs_root, spdif, - &tegra20_spdif_debug_fops); -} - -static void tegra20_spdif_debug_remove(struct tegra20_spdif *spdif) -{ - if (spdif->debug) - debugfs_remove(spdif->debug); -} -#else -static inline void tegra20_spdif_debug_add(struct tegra20_spdif *spdif) -{ -} - -static inline void tegra20_spdif_debug_remove(struct tegra20_spdif *spdif) -{ -} -#endif - static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -265,10 +190,86 @@ static struct snd_soc_dai_driver tegra20_spdif_dai = { .ops = &tegra20_spdif_dai_ops, }; +static bool tegra20_spdif_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_SPDIF_CTRL: + case TEGRA20_SPDIF_STATUS: + case TEGRA20_SPDIF_STROBE_CTRL: + case TEGRA20_SPDIF_DATA_FIFO_CSR: + case TEGRA20_SPDIF_DATA_OUT: + case TEGRA20_SPDIF_DATA_IN: + case TEGRA20_SPDIF_CH_STA_RX_A: + case TEGRA20_SPDIF_CH_STA_RX_B: + case TEGRA20_SPDIF_CH_STA_RX_C: + case TEGRA20_SPDIF_CH_STA_RX_D: + case TEGRA20_SPDIF_CH_STA_RX_E: + case TEGRA20_SPDIF_CH_STA_RX_F: + case TEGRA20_SPDIF_CH_STA_TX_A: + case TEGRA20_SPDIF_CH_STA_TX_B: + case TEGRA20_SPDIF_CH_STA_TX_C: + case TEGRA20_SPDIF_CH_STA_TX_D: + case TEGRA20_SPDIF_CH_STA_TX_E: + case TEGRA20_SPDIF_CH_STA_TX_F: + case TEGRA20_SPDIF_USR_STA_RX_A: + case TEGRA20_SPDIF_USR_DAT_TX_A: + return true; + default: + return false; + }; +} + +static bool tegra20_spdif_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_SPDIF_STATUS: + case TEGRA20_SPDIF_DATA_FIFO_CSR: + case TEGRA20_SPDIF_DATA_OUT: + case TEGRA20_SPDIF_DATA_IN: + case TEGRA20_SPDIF_CH_STA_RX_A: + case TEGRA20_SPDIF_CH_STA_RX_B: + case TEGRA20_SPDIF_CH_STA_RX_C: + case TEGRA20_SPDIF_CH_STA_RX_D: + case TEGRA20_SPDIF_CH_STA_RX_E: + case TEGRA20_SPDIF_CH_STA_RX_F: + case TEGRA20_SPDIF_USR_STA_RX_A: + case TEGRA20_SPDIF_USR_DAT_TX_A: + return true; + default: + return false; + }; +} + +static bool tegra20_spdif_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_SPDIF_DATA_OUT: + case TEGRA20_SPDIF_DATA_IN: + case TEGRA20_SPDIF_USR_STA_RX_A: + case TEGRA20_SPDIF_USR_DAT_TX_A: + return true; + default: + return false; + }; +} + +static const struct regmap_config tegra20_spdif_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA20_SPDIF_USR_DAT_TX_A, + .writeable_reg = tegra20_spdif_wr_rd_reg, + .readable_reg = tegra20_spdif_wr_rd_reg, + .volatile_reg = tegra20_spdif_volatile_reg, + .precious_reg = tegra20_spdif_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; + static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) { struct tegra20_spdif *spdif; struct resource *mem, *memregion, *dmareq; + void __iomem *regs; int ret; spdif = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_spdif), @@ -309,13 +310,21 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) goto err_clk_put; } - spdif->regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!spdif->regs) { + regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!regs) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENOMEM; goto err_clk_put; } + spdif->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra20_spdif_regmap_config); + if (IS_ERR(spdif->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(spdif->regmap); + goto err_clk_put; + } + spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; spdif->playback_dma_data.wrap = 4; spdif->playback_dma_data.width = 32; @@ -341,8 +350,6 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) goto err_unregister_dai; } - tegra20_spdif_debug_add(spdif); - return 0; err_unregister_dai: @@ -369,8 +376,6 @@ static int __devexit tegra20_spdif_platform_remove(struct platform_device *pdev) tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); - tegra20_spdif_debug_remove(spdif); - clk_put(spdif->clk_spdif_out); return 0; diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h index 823af4c..ed75652 100644 --- a/sound/soc/tegra/tegra20_spdif.h +++ b/sound/soc/tegra/tegra20_spdif.h @@ -464,8 +464,7 @@ struct tegra20_spdif { struct clk *clk_spdif_out; struct tegra_pcm_dma_params capture_dma_data; struct tegra_pcm_dma_params playback_dma_data; - void __iomem *regs; - struct dentry *debug; + struct regmap *regmap; u32 reg_ctrl; }; -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: wm9712: Fix build due to missing definition of "runtime"
by Fabio Estevam 10 Apr '12

10 Apr '12

Fix the following build error: sound/soc/codecs/wm9712.c:482:32: error: 'runtime' undeclared (first use in this function) sound/soc/codecs/wm9712.c:499:33: error: 'runtime' undeclared (first use in this function) Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm9712.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2603863..a154141 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -470,6 +470,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); @@ -487,6 +488,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; u16 vra, xsle; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); -- 1.7.1

2 3

[alsa-devel] [PATCH] ASoC: tegra: utils: Don't use of_have_populated_dt()
by Stephen Warren 10 Apr '12

10 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> Recent list discussions concluded that drivers should not be calling of_have_populated_dt(), and hence of_have_populated_dt() should not be exported. Use a different mechanism to detect DT vs. non-DT boot. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_asoc_utils.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 266189d..9515ce5 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -119,13 +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")) + 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 if (!dev->of_node) + /* non-DT is always Tegra20 */ + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else + /* DT boot, but unknown SoC */ return -EINVAL; data->clk_pll_a = clk_get_sys(NULL, "pll_a"); -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: wm9712: Fix build due to missing definition of "runtime"
by Fabio Estevam 10 Apr '12

10 Apr '12

Fix the following build error: sound/soc/codecs/wm9712.c:482:32: error: 'runtime' undeclared (first use in this function) sound/soc/codecs/wm9712.c:499:33: error: 'runtime' undeclared (first use in this function) Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm9712.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2603863..a154141 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -470,6 +470,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); @@ -487,6 +488,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; u16 vra, xsle; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); -- 1.7.1

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[alsa-devel] [GIT PULL]
by Stephen Warren 10 Apr '12

10 Apr '12

Mark, Could you please pull the following branches into ASoC's for-next: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap.git topic/mmio git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap.git topic/stride Those two are a dependency for the Tegra20 ASoC regmap conversion that I can post now. git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap.git topic/name That is an additional dependency for the new Tegra30 ASoC drivers. Thanks.

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Re: [alsa-devel] [PATCH RESEND] ASoC: Fix setting aif_val in lm49453_set_dai_fmt
by MR Swami Reddy 10 Apr '12

10 Apr '12

This issues was fixed last week itself. URL for the patch - http://mailman.alsa-project.org/pipermail/alsa-devel/2012-April/050967.html Thanks Swami On Monday 09 April 2012 05:18 PM, Axel Lin wrote: > CBS/CBM/CFS/CFM settings are controlled by BIT[3:4] of 0x60h > > for example, for SND_SOC_DAIFMT_CBS_CFS: what we want is to clear both BIT[3:4]. > > aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; > actually set both bits. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > Seems the my mail failed to reach alsa mailing list. > I resend to LKML instead. > > Axel > sound/soc/codecs/lm49453.c | 8 ++------ > 1 files changed, 2 insertions(+), 6 deletions(-) > > diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c > index 744063d..a25e80a 100644 > --- a/sound/soc/codecs/lm49453.c > +++ b/sound/soc/codecs/lm49453.c > @@ -1185,16 +1185,12 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) > > switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { > case SND_SOC_DAIFMT_CBS_CFS: > - aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | > - ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; > break; > case SND_SOC_DAIFMT_CBS_CFM: > - aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | > - LM49453_AUDIO_PORT1_BASIC_SYNC_MS; > + aif_val = LM49453_AUDIO_PORT1_BASIC_SYNC_MS; > break; > case SND_SOC_DAIFMT_CBM_CFS: > - aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | > - ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; > + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS; > break; > case SND_SOC_DAIFMT_CBM_CFM: > aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | > -- > 1.7.5.4 > > >

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[alsa-devel] [PATCH] ASoC: dapm: Allow DAI widgets to be routed through
by Mark Brown 10 Apr '12

10 Apr '12

In order to allow CODEC<->CODEC links to function we will need to allow DAPM paths to be created that pass through DAIs rather than only ones that are source or sunk at the DAI. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d97e652..2c3363e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -923,7 +923,10 @@ static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) { DAPM_UPDATE_STAT(w, power_checks); - return w->active; + if (w->active) + return w->active; + + return dapm_generic_check_power(w); } /* Check to see if an ADC has power */ -- 1.7.9.1

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Re: [alsa-devel] [PATCH 6/6] ALSA: hda - clean up CX20549 test mixer setup
by David Henningsson 10 Apr '12

10 Apr '12

On 04/06/2012 03:34 PM, Michael Karcher wrote: > name pins consistently (MIC1/LINE1/HP-OUT/CD) on all controls > affecting those pins. If this is a test mixer only, so maybe it does not matter that much, but anyway: Loopback controls are usually labeled "Line Playback Volume" "Mic Playback Volume" And outputs: "Headphone Playback Volume" For line outputs, I'm not sure if the right thing is "Front Playback Volume", "Line-Out Playback Volume", or "Line Out Playback Volume" > remove duplicate SET_AMP_GAIN_MUTE to 0x17/index 0 and 0x17/index 1 > > really select MIC1, not Mixer out for recording > > "Mixer out" for recording is not a "pin", adjust comment > > Signed-off-by: Michael Karcher<kernel(a)mkarcher.dialup.fu-berlin.de> > --- > sound/pci/hda/patch_conexant.c | 38 +++++++++++++++++--------------------- > 1 files changed, 17 insertions(+), 21 deletions(-) > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 8fe9f9d..a36488d 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -930,10 +930,10 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { > /* Output controls */ > HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT), > HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT), > - HDA_CODEC_VOLUME("Node 11 Playback Volume", 0x11, 0x0, HDA_OUTPUT), > - HDA_CODEC_MUTE("Node 11 Playback Switch", 0x11, 0x0, HDA_OUTPUT), > - HDA_CODEC_VOLUME("Node 12 Playback Volume", 0x12, 0x0, HDA_OUTPUT), > - HDA_CODEC_MUTE("Node 12 Playback Switch", 0x12, 0x0, HDA_OUTPUT), > + HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT), > + HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT), > + HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT), > + HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT), > > /* Modes for retasking pin widgets */ > CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT), > @@ -944,16 +944,16 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { > > /* Loopback mixer controls */ > > - HDA_CODEC_VOLUME("Mixer-1 Volume", 0x17, 0x0, HDA_INPUT), > - HDA_CODEC_MUTE("Mixer-1 Switch", 0x17, 0x0, HDA_INPUT), > - HDA_CODEC_VOLUME("Mixer-2 Volume", 0x17, 0x1, HDA_INPUT), > - HDA_CODEC_MUTE("Mixer-2 Switch", 0x17, 0x1, HDA_INPUT), > - HDA_CODEC_VOLUME("Mixer-3 Volume", 0x17, 0x2, HDA_INPUT), > - HDA_CODEC_MUTE("Mixer-3 Switch", 0x17, 0x2, HDA_INPUT), > - HDA_CODEC_VOLUME("Mixer-4 Volume", 0x17, 0x3, HDA_INPUT), > - HDA_CODEC_MUTE("Mixer-4 Switch", 0x17, 0x3, HDA_INPUT), > - HDA_CODEC_VOLUME("Mixer-5 Volume", 0x17, 0x4, HDA_INPUT), > - HDA_CODEC_MUTE("Mixer-5 Switch", 0x17, 0x4, HDA_INPUT), > + HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT), > + HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT), > + HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT), > + HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT), > + HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT), > + HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT), > + HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT), > + HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT), Eh, is there a loopback control from the headphone? > + HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT), > + HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT), > { > .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > .name = "Input Source", > @@ -985,10 +985,6 @@ static const struct hda_verb cxt5045_test_init_verbs[] = { > {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, > {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0}, > > - /* Start with output sum widgets muted and their output gains at min */ > - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, > - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, > - > /* Unmute retasking pin widget output buffers since the default > * state appears to be output. As the pin mode is changed by the > * user the pin mode control will take care of enabling the pin's > @@ -1003,11 +999,11 @@ static const struct hda_verb cxt5045_test_init_verbs[] = { > /* Set ADC connection select to match default mixer setting (mic1 > * pin) > */ > - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, > - {0x17, AC_VERB_SET_CONNECT_SEL, 0x00}, > + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, > + {0x17, AC_VERB_SET_CONNECT_SEL, 0x01}, > > /* Mute all inputs to mixer widget (even unconnected ones) */ > - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer pin */ > + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */ > {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */ > {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */ > {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */ -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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