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

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

15 May '13

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

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

15 May '13

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

2 1

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

15 May '13

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

2 13

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

15 May '13

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

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

15 May '13

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

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

15 May '13

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

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

15 May '13

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

4 3

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

15 May '13

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

3 21

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

15 May '13

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

3 3

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

14 May '13

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

2 1

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

14 May '13

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

2 1

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

14 May '13

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

3 2

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

14 May '13

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

2 1

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

14 May '13

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

3 2

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

14 May '13

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

3 2

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

14 May '13

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

3 4

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

14 May '13

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

2 1

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

13 May '13

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

3 19

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

13 May '13

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

2 4

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

13 May '13

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

3 3

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

13 May '13

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

2 1

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

13 May '13

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

2 1

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

13 May '13

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

5 21

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

13 May '13

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

2 1

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

13 May '13

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

2 1

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

13 May '13

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

1 0

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

13 May '13

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

2 1

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

12 May '13

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

2 1

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

11 May '13

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

6 11

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

10 May '13

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

4 16

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

10 May '13

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

1 0

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

10 May '13

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

1 0

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

10 May '13

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

1 0

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

10 May '13

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

1 0

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

10 May '13

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

4 4

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

10 May '13

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

1 0

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

10 May '13

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

4 6

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

10 May '13

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

4 5

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

10 May '13

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

2 1

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

09 May '13

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

1 0

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

09 May '13

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

2 1

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

09 May '13

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

3 2

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

08 May '13

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

3 5

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

08 May '13

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

2 1

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

08 May '13

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

2 1

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

08 May '13

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

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

08 May '13

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

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

08 May '13

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

2 1

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

07 May '13

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

3 5

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

07 May '13

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

1 0