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Re: [alsa-devel] VOIP xrun problem
by Jamey Drennan 24 Oct '12

24 Oct '12

Maybe I don't quite understand your question but I use RTP (oRTP library) to manage the sending/receiving of the audio streams. On Mon, Oct 22, 2012 at 4:45 PM, Florian Faber <faber(a)faberman.de> wrote: > On 10/22/12 22:40, Jamey Drennan wrote: > > Now in the second phase, events are timed due to specification > > requirements. The same Alsa setup code is used yet xruns (mostly > > underruns) are all over the place. The underruns are periodic and can be > > reduced by increasing the period size and number of periods (this also > > increases the latency obviously and is undersirable) but never removed > > completely. > > > > My question is how to reduce/remove the xruns. > > How do you synchronize the streams? You will either have to synchronize > the media clocks or do some sort of SRC. > > > Flo > -- > Machines can do the work, so people have time to think. > public key B3B9226C x-hkp://wwwkeys.eu.pgp.net >

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Re: [alsa-devel] [PATCH 2/2] ASoC: ux500_msp_i2s: Fix devm_* and return code merge error
by Mark Brown 24 Oct '12

24 Oct '12

On Wed, Oct 24, 2012 at 11:34:00AM +0200, Ulf Hansson wrote: > Acked-by: Ulf Hansson <ulf.hansson(a)linaro.org> > > I assume this shall go into 3.7, right? > > Kind regards > Ulf Hansson You may recall that only yesterday I reminded you not to top post...

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[alsa-devel] [PATCH] ASoC: wm8994: Lower AIFnCLK divisor when dropping to 32kHz
by Mark Brown 24 Oct '12

24 Oct '12

When lowering SYSCLK to 50kHz for accessory detection also lower the AIFnCLK divisor to normalise the clocking configuration within the device. This will not disrupt audio as we cannot support active audio with such a low SYSCLK. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0932d7a..c16a9a3 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2443,6 +2443,18 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, configure_clock(codec); + /* + * If SYSCLK will be less than 50kHz adjust AIFnCLK dividers + * for detection. + */ + if (max(wm8994->aifclk[0], wm8994->aifclk[1]) < 50000) { + dev_dbg(codec->dev, "Configuring AIFs for 128fs\n"); + snd_soc_update_bits(codec, WM8994_AIF1_RATE, + WM8994_AIF1CLK_RATE_MASK, 0x1); + snd_soc_update_bits(codec, WM8994_AIF2_RATE, + WM8994_AIF2CLK_RATE_MASK, 0x1); + } + return 0; } -- 1.7.10.4

1 0

[alsa-devel] [GIT PULL] dummy: allow disabling mixer controls
by Clemens Ladisch 24 Oct '12

24 Oct '12

The following changes since commit a0d271cbfed1dd50278c6b06bead3d00ba0a88f9: Linux 3.6 (2012-09-30 16:47:46 -0700) are available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git dummy-ctl-inactive ---------------------------------------------------------------- Quite a few mixer applications do not handle deactivated controls correctly. This patch adds such controls to snd-dummy to make crash^H^H^H^H^Htesting these apps easier. ---------------------------------------------------------------- Clemens Ladisch (1): ALSA: dummy: allow disabling mixer controls sound/drivers/dummy.c | 73 +++++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 71 insertions(+), 2 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 54bb664..4f522cf 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -134,6 +134,9 @@ struct snd_dummy { spinlock_t mixer_lock; int mixer_volume[MIXER_ADDR_LAST+1][2]; int capture_source[MIXER_ADDR_LAST+1][2]; + int iobox; + struct snd_kcontrol *cd_volume_ctl; + struct snd_kcontrol *cd_switch_ctl; const struct dummy_timer_ops *timer_ops; }; @@ -817,6 +820,57 @@ static int snd_dummy_capsrc_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el return change; } +static int snd_dummy_iobox_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + const char *const names[] = { "None", "CD Player" }; + + return snd_ctl_enum_info(info, 1, 2, names); +} + +static int snd_dummy_iobox_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + struct snd_dummy *dummy = snd_kcontrol_chip(kcontrol); + + value->value.enumerated.item[0] = dummy->iobox; + return 0; +} + +static int snd_dummy_iobox_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + struct snd_dummy *dummy = snd_kcontrol_chip(kcontrol); + int changed; + + if (value->value.enumerated.item[0] > 1) + return -EINVAL; + + changed = value->value.enumerated.item[0] != dummy->iobox; + if (changed) { + dummy->iobox = value->value.enumerated.item[0]; + + if (dummy->iobox) { + dummy->cd_volume_ctl->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + dummy->cd_switch_ctl->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } else { + dummy->cd_volume_ctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_INACTIVE; + dummy->cd_switch_ctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } + + snd_ctl_notify(dummy->card, SNDRV_CTL_EVENT_MASK_INFO, + &dummy->cd_volume_ctl->id); + snd_ctl_notify(dummy->card, SNDRV_CTL_EVENT_MASK_INFO, + &dummy->cd_switch_ctl->id); + } + + return changed; +} + static struct snd_kcontrol_new snd_dummy_controls[] = { DUMMY_VOLUME("Master Volume", 0, MIXER_ADDR_MASTER), DUMMY_CAPSRC("Master Capture Switch", 0, MIXER_ADDR_MASTER), @@ -827,22 +881,37 @@ DUMMY_CAPSRC("Line Capture Switch", 0, MIXER_ADDR_LINE), DUMMY_VOLUME("Mic Volume", 0, MIXER_ADDR_MIC), DUMMY_CAPSRC("Mic Capture Switch", 0, MIXER_ADDR_MIC), DUMMY_VOLUME("CD Volume", 0, MIXER_ADDR_CD), -DUMMY_CAPSRC("CD Capture Switch", 0, MIXER_ADDR_CD) +DUMMY_CAPSRC("CD Capture Switch", 0, MIXER_ADDR_CD), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "External I/O Box", + .info = snd_dummy_iobox_info, + .get = snd_dummy_iobox_get, + .put = snd_dummy_iobox_put, +}, }; static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy) { struct snd_card *card = dummy->card; + struct snd_kcontrol *kcontrol; unsigned int idx; int err; spin_lock_init(&dummy->mixer_lock); strcpy(card->mixername, "Dummy Mixer"); + dummy->iobox = 1; for (idx = 0; idx < ARRAY_SIZE(snd_dummy_controls); idx++) { - err = snd_ctl_add(card, snd_ctl_new1(&snd_dummy_controls[idx], dummy)); + kcontrol = snd_ctl_new1(&snd_dummy_controls[idx], dummy); + err = snd_ctl_add(card, kcontrol); if (err < 0) return err; + if (!strcmp(kcontrol->id.name, "CD Volume")) + dummy->cd_volume_ctl = kcontrol; + else if (!strcmp(kcontrol->id.name, "CD Capture Switch")) + dummy->cd_switch_ctl = kcontrol; + } return 0; }

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[alsa-devel] Passing audio info from HDMI EDID beside speaker allocation
by Patrick Lai 24 Oct '12

24 Oct '12

Hi Takashi, You proposed snd_pcm_query_chmaps() to query underlying hardware speaker allocation. Please correct me if I have wrong understanding of this API. On the other hand, HDMI EDID contains audio data other than speaker allocation such as audio formats(LPCM, AC3, and etc). How are these information other than speaker allocation passed back to the user-space? Are there a standard kernel and alsa query interfaces now? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] [PATCH -next] sound: ice1712 build errors
by Randy Dunlap 23 Oct '12

23 Oct '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix build errors by using correct kconfig symbol name: sound/pci/ice1712/psc724.c:417:5: error: 'struct snd_ice1712' has no member named 'pm_resume' sound/pci/ice1712/psc724.c:418:5: error: 'struct snd_ice1712' has no member named 'pm_suspend_enabled' Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> Cc: Ondrej Zary <linux(a)rainbow-software.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/pci/ice1712/psc724.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20121023.orig/sound/pci/ice1712/psc724.c +++ linux-next-20121023/sound/pci/ice1712/psc724.c @@ -413,7 +413,7 @@ static int __devinit psc724_init(struct snd_wm8776_init(&spec->wm8776); spec->wm8766.ops.write = psc724_wm8766_write; spec->wm8766.card = ice->card; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP ice->pm_resume = psc724_resume; ice->pm_suspend_enabled = 1; #endif

3 2

[alsa-devel] [PATCH 1/3] ALSA: core: keep track of boundary wrap-around
by Pierre-Louis Bossart 23 Oct '12

23 Oct '12

Keep track of boundary crossing when hw_ptr exceeds boundary limit and wraps-around. This will help keep track of total number of frames played/received at the kernel level Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 25 ++++++++++++++++++++----- 2 files changed, 21 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 6268a41..28fd9f9 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -281,6 +281,7 @@ struct snd_pcm_runtime { unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ unsigned long hw_ptr_buffer_jiffies; /* buffer time in jiffies */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ + u64 hw_ptr_wrap; /* offset for hw_ptr due to boundary wrap-around */ /* -- HW params -- */ snd_pcm_access_t access; /* access mode */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index f42c10a..3dc029e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -316,6 +316,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, unsigned long jdelta; unsigned long curr_jiffies; struct timespec curr_tstamp; + int crossed_boundary = 0; old_hw_ptr = runtime->status->hw_ptr; @@ -360,8 +361,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, hdelta = curr_jiffies - runtime->hw_ptr_jiffies; if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) + if (hw_base >= runtime->boundary) { hw_base = 0; + crossed_boundary++; + } new_hw_ptr = hw_base + pos; goto __delta; } @@ -371,8 +374,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* pointer crosses the end of the ring buffer */ if (new_hw_ptr < old_hw_ptr) { hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) + if (hw_base >= runtime->boundary) { hw_base = 0; + crossed_boundary++; + } new_hw_ptr = hw_base + pos; } __delta: @@ -410,8 +415,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, while (hdelta > xrun_threshold) { delta += runtime->buffer_size; hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) + if (hw_base >= runtime->boundary) { hw_base = 0; + crossed_boundary++; + } new_hw_ptr = hw_base + pos; hdelta -= runtime->hw_ptr_buffer_jiffies; } @@ -456,8 +463,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* the delta value is small or zero in most cases */ while (delta > 0) { new_hw_ptr += runtime->period_size; - if (new_hw_ptr >= runtime->boundary) + if (new_hw_ptr >= runtime->boundary) { new_hw_ptr -= runtime->boundary; + crossed_boundary--; + } delta--; } /* align hw_base to buffer_size */ @@ -507,6 +516,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = curr_jiffies; + if (crossed_boundary) { + snd_BUG_ON(crossed_boundary != 1); + runtime->hw_ptr_wrap += runtime->boundary; + } if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) runtime->status->tstamp = curr_tstamp; @@ -1661,8 +1674,10 @@ static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream, if (snd_pcm_running(substream) && snd_pcm_update_hw_ptr(substream) >= 0) runtime->status->hw_ptr %= runtime->buffer_size; - else + else { runtime->status->hw_ptr = 0; + runtime->hw_ptr_wrap = 0; + } snd_pcm_stream_unlock_irqrestore(substream, flags); return 0; } -- 1.7.9.5

2 3

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for Reloop Play
by Didier Villevalois 23 Oct '12

23 Oct '12

The Reloop Audio needs a fixed endpoint quirk with S24_3LE format and UAC_EP_CS_ATTR_SAMPLE_RATE attribute. Signed-off-by: Didier Villevalois <ptitjes(a)free.fr> --- sound/usb/quirks-table.h | 41 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 41 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d73ac9b..48ede8b 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2834,6 +2834,47 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Reloop Play */ +{ + USB_DEVICE(0x200c, 0x100b), + .bInterfaceClass = USB_CLASS_PER_INTERFACE, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_SYNC_ADAPTIVE, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = -1 + } + } + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- 1.7.10.4

2 1

[alsa-devel] sound/pci/au88x0/au88x0_core.c:2464 vortex_interrupt() warn: if();
by Fengguang Wu 23 Oct '12

23 Oct '12

Hi, smatch warns about a weird "if();" in sound/pci/au88x0/au88x0_core.c Any ideas to fix it? 2461 #ifndef CHIP_AU8810 2462 for (i = 0; i < NR_WT; i++) { 2463 if (vortex->dma_wt[i].fifo_status == FIFO_START) { ==> 2464 if (vortex_wtdma_bufshift(vortex, i)) ; 2465 spin_unlock(&vortex->lock); 2466 snd_pcm_period_elapsed(vortex->dma_wt[i]. 2467 substream); 2468 spin_lock(&vortex->lock); 2469 } 2470 } 2471 #endif Thanks, Fengguang

2 1

[alsa-devel] [PATCH] [v2] ASoC: add support for the Freescale / iVeia P1022 RDK reference board
by Timur Tabi 23 Oct '12

23 Oct '12

This small reference boards has a Freescale P1022 dual-core PowerPC SOC and a Wolfson Microelectronics WM8960 codec. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/Kconfig | 14 ++ sound/soc/fsl/Makefile | 4 + sound/soc/fsl/p1022_rdk.c | 392 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 410 insertions(+), 0 deletions(-) create mode 100644 sound/soc/fsl/p1022_rdk.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 4563b28..8e48432 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -46,6 +46,20 @@ config SND_SOC_P1022_DS This will also include the Wolfson Microelectronics WM8776 codec driver. +config SND_SOC_P1022_RDK + tristate "ALSA SoC support for the Freescale / iVeia P1022 RDK board" + # I2C is necessary for the WM8960 driver + depends on P1022_RDK && I2C + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + select SND_SOC_POWERPC_DMA + select SND_SOC_WM8960 + default y if P1022_RDK + help + Say Y if you want to enable audio on the Freescale / iVeia + P1022 RDK board. This will also include the Wolfson + Microelectronics WM8960 codec driver. + config SND_SOC_MPC5200_I2S tristate "Freescale MPC5200 PSC in I2S mode driver" depends on PPC_MPC52xx && PPC_BESTCOMM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 5f3cf3f..515ba66 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -6,6 +6,10 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o snd-soc-p1022-ds-objs := p1022_ds.o obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o +# P1022 RDK Machine Support +snd-soc-p1022-rdk-objs := p1022_rdk.o +obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o + # Freescale PowerPC SSI/DMA Platform Support snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-utils-objs := fsl_utils.o diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c new file mode 100644 index 0000000..897e32f --- /dev/null +++ b/sound/soc/fsl/p1022_rdk.c @@ -0,0 +1,392 @@ +/** + * Freescale P1022RDK ALSA SoC Machine driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2012 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + * + * Note: in order for audio to work correctly, the output controls need + * to be enabled, because they control the clock. So for playback, for + * example: + * + * amixer sset 'Left Output Mixer PCM' on + * amixer sset 'Right Output Mixer PCM' on + */ + +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/of_device.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <asm/fsl_guts.h> + +#include "fsl_dma.h" +#include "fsl_ssi.h" +#include "fsl_utils.h" + +/* P1022-specific PMUXCR and DMUXCR bit definitions */ + +#define CCSR_GUTS_PMUXCR_UART0_I2C1_MASK 0x0001c000 +#define CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI 0x00010000 +#define CCSR_GUTS_PMUXCR_UART0_I2C1_SSI 0x00018000 + +#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK 0x00000c00 +#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI 0x00000000 + +#define CCSR_GUTS_DMUXCR_PAD 1 /* DMA controller/channel set to pad */ +#define CCSR_GUTS_DMUXCR_SSI 2 /* DMA controller/channel set to SSI */ + +/* + * Set the DMACR register in the GUTS + * + * The DMACR register determines the source of initiated transfers for each + * channel on each DMA controller. Rather than have a bunch of repetitive + * macros for the bit patterns, we just have a function that calculates + * them. + * + * guts: Pointer to GUTS structure + * co: The DMA controller (0 or 1) + * ch: The channel on the DMA controller (0, 1, 2, or 3) + * device: The device to set as the target (CCSR_GUTS_DMUXCR_xxx) + */ +static inline void guts_set_dmuxcr(struct ccsr_guts __iomem *guts, + unsigned int co, unsigned int ch, unsigned int device) +{ + unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch)); + + clrsetbits_be32(&guts->dmuxcr, 3 << shift, device << shift); +} + +/* There's only one global utilities register */ +static phys_addr_t guts_phys; + +/** + * machine_data: machine-specific ASoC device data + * + * This structure contains data for a single sound platform device on an + * P1022 RDK. Some of the data is taken from the device tree. + */ +struct machine_data { + struct snd_soc_dai_link dai[2]; + struct snd_soc_card card; + unsigned int dai_format; + unsigned int codec_clk_direction; + unsigned int cpu_clk_direction; + unsigned int clk_frequency; + unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ + unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ + char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ +}; + +/** + * p1022_rdk_machine_probe: initialize the board + * + * This function is used to initialize the board-specific hardware. + * + * Here we program the DMACR and PMUXCR registers. + */ +static int p1022_rdk_machine_probe(struct snd_soc_card *card) +{ + struct machine_data *mdata = + container_of(card, struct machine_data, card); + struct ccsr_guts __iomem *guts; + + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + if (!guts) { + dev_err(card->dev, "could not map global utilities\n"); + return -ENOMEM; + } + + /* Enable SSI Tx signal */ + clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK, + CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI); + + /* Enable SSI Rx signal */ + clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK, + CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI); + + /* Enable DMA Channel for SSI */ + guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0], + CCSR_GUTS_DMUXCR_SSI); + + guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1], + CCSR_GUTS_DMUXCR_SSI); + + iounmap(guts); + + return 0; +} + +/** + * p1022_rdk_startup: program the board with various hardware parameters + * + * This function takes board-specific information, like clock frequencies + * and serial data formats, and passes that information to the codec and + * transport drivers. + */ +static int p1022_rdk_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct machine_data *mdata = + container_of(rtd->card, struct machine_data, card); + struct device *dev = rtd->card->dev; + int ret = 0; + + /* Tell the codec driver what the serial protocol is. */ + ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format); + if (ret < 0) { + dev_err(dev, "could not set codec driver audio format (ret=%i)\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_pll(rtd->codec_dai, 0, 0, mdata->clk_frequency, + mdata->clk_frequency); + if (ret < 0) { + dev_err(dev, "could not set codec PLL frequency (ret=%i)\n", + ret); + return ret; + } + + return 0; +} + +/** + * p1022_rdk_machine_remove: Remove the sound device + * + * This function is called to remove the sound device for one SSI. We + * de-program the DMACR and PMUXCR register. + */ +static int p1022_rdk_machine_remove(struct snd_soc_card *card) +{ + struct machine_data *mdata = + container_of(card, struct machine_data, card); + struct ccsr_guts __iomem *guts; + + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + if (!guts) { + dev_err(card->dev, "could not map global utilities\n"); + return -ENOMEM; + } + + /* Restore the signal routing */ + clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK); + clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK); + guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0], 0); + guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1], 0); + + iounmap(guts); + + return 0; +} + +/** + * p1022_rdk_ops: ASoC machine driver operations + */ +static struct snd_soc_ops p1022_rdk_ops = { + .startup = p1022_rdk_startup, +}; + +/** + * p1022_rdk_probe: platform probe function for the machine driver + * + * Although this is a machine driver, the SSI node is the "master" node with + * respect to audio hardware connections. Therefore, we create a new ASoC + * device for each new SSI node that has a codec attached. + */ +static int p1022_rdk_probe(struct platform_device *pdev) +{ + struct device *dev = pdev->dev.parent; + /* ssi_pdev is the platform device for the SSI node that probed us */ + struct platform_device *ssi_pdev = + container_of(dev, struct platform_device, dev); + struct device_node *np = ssi_pdev->dev.of_node; + struct device_node *codec_np = NULL; + struct machine_data *mdata; + const u32 *iprop; + int ret; + + /* Find the codec node for this SSI. */ + codec_np = of_parse_phandle(np, "codec-handle", 0); + if (!codec_np) { + dev_err(dev, "could not find codec node\n"); + return -EINVAL; + } + + mdata = kzalloc(sizeof(struct machine_data), GFP_KERNEL); + if (!mdata) { + ret = -ENOMEM; + goto error_put; + } + + mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); + mdata->dai[0].ops = &p1022_rdk_ops; + + /* ASoC core can match codec with device node */ + mdata->dai[0].codec_of_node = codec_np; + + /* + * We register two DAIs per SSI, one for playback and the other for + * capture. We support codecs that have separate DAIs for both playback + * and capture. + */ + memcpy(&mdata->dai[1], &mdata->dai[0], sizeof(struct snd_soc_dai_link)); + + /* The DAI names from the codec (snd_soc_dai_driver.name) */ + mdata->dai[0].codec_dai_name = "wm8960-hifi"; + mdata->dai[1].codec_dai_name = mdata->dai[0].codec_dai_name; + + /* + * Configure the SSI for I2S slave mode. Older device trees have + * an fsl,mode property, but we ignore that since there's really + * only one way to configure the SSI. + */ + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; + mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; + mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; + + /* + * In i2s-slave mode, the codec has its own clock source, so we + * need to get the frequency from the device tree and pass it to + * the codec driver. + */ + iprop = of_get_property(codec_np, "clock-frequency", NULL); + if (!iprop || !*iprop) { + dev_err(&pdev->dev, "codec bus-frequency property is missing or invalid\n"); + ret = -EINVAL; + goto error; + } + mdata->clk_frequency = be32_to_cpup(iprop); + + if (!mdata->clk_frequency) { + dev_err(&pdev->dev, "unknown clock frequency\n"); + ret = -EINVAL; + goto error; + } + + /* Find the playback DMA channel to use. */ + mdata->dai[0].platform_name = mdata->platform_name[0]; + ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0], + &mdata->dma_channel_id[0], + &mdata->dma_id[0]); + if (ret) { + dev_err(&pdev->dev, "missing/invalid playback DMA phandle (ret=%i)\n", + ret); + goto error; + } + + /* Find the capture DMA channel to use. */ + mdata->dai[1].platform_name = mdata->platform_name[1]; + ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1], + &mdata->dma_channel_id[1], + &mdata->dma_id[1]); + if (ret) { + dev_err(&pdev->dev, "missing/invalid capture DMA phandle (ret=%i)\n", + ret); + goto error; + } + + /* Initialize our DAI data structure. */ + mdata->dai[0].stream_name = "playback"; + mdata->dai[1].stream_name = "capture"; + mdata->dai[0].name = mdata->dai[0].stream_name; + mdata->dai[1].name = mdata->dai[1].stream_name; + + mdata->card.probe = p1022_rdk_machine_probe; + mdata->card.remove = p1022_rdk_machine_remove; + mdata->card.name = pdev->name; /* The platform driver name */ + mdata->card.owner = THIS_MODULE; + mdata->card.dev = &pdev->dev; + mdata->card.num_links = 2; + mdata->card.dai_link = mdata->dai; + + /* Register with ASoC */ + ret = snd_soc_register_card(&mdata->card); + if (ret) { + dev_err(&pdev->dev, "could not register card (ret=%i)\n", ret); + goto error; + } + + return 0; + +error: + kfree(mdata); +error_put: + of_node_put(codec_np); + return ret; +} + +/** + * p1022_rdk_remove: remove the platform device + * + * This function is called when the platform device is removed. + */ +static int __devexit p1022_rdk_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct machine_data *mdata = + container_of(card, struct machine_data, card); + + snd_soc_unregister_card(card); + kfree(mdata); + + return 0; +} + +static struct platform_driver p1022_rdk_driver = { + .probe = p1022_rdk_probe, + .remove = __devexit_p(p1022_rdk_remove), + .driver = { + /* + * The name must match 'compatible' property in the device tree, + * in lowercase letters. + */ + .name = "snd-soc-p1022rdk", + .owner = THIS_MODULE, + }, +}; + +/** + * p1022_rdk_init: machine driver initialization. + * + * This function is called when this module is loaded. + */ +static int __init p1022_rdk_init(void) +{ + struct device_node *guts_np; + struct resource res; + + /* Get the physical address of the global utilities registers */ + guts_np = of_find_compatible_node(NULL, NULL, "fsl,p1022-guts"); + if (of_address_to_resource(guts_np, 0, &res)) { + pr_err("snd-soc-p1022rdk: missing/invalid global utils node\n"); + of_node_put(guts_np); + return -EINVAL; + } + guts_phys = res.start; + of_node_put(guts_np); + + return platform_driver_register(&p1022_rdk_driver); +} + +/** + * p1022_rdk_exit: machine driver exit + * + * This function is called when this driver is unloaded. + */ +static void __exit p1022_rdk_exit(void) +{ + platform_driver_unregister(&p1022_rdk_driver); +} + +late_initcall(p1022_rdk_init); +module_exit(p1022_rdk_exit); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Freescale / iVeia P1022 RDK ALSA SoC machine driver"); +MODULE_LICENSE("GPL v2"); -- 1.7.3.4

3 2

[alsa-devel] VOIP xrun problem
by Jamey Drennan 22 Oct '12

22 Oct '12

A quick background: I'm developing a VOIP program for an embedded system. In the first phase of the program everything was event driven. Basically ask ALSA for samples, encode them, and send them via UDP to the connected client. On the other end, receive the packet, decode, play the samples. After setting the correct value for the start_threshold function everything worked great no xruns at all. Now in the second phase, events are timed due to specification requirements. The same Alsa setup code is used yet xruns (mostly underruns) are all over the place. The underruns are periodic and can be reduced by increasing the period size and number of periods (this also increases the latency obviously and is undersirable) but never removed completely. My question is how to reduce/remove the xruns. I am not currently at a point where installing a real-time OS or patching to get these features is possible. Also, when aplay/arecord is run on the system, no xruns occur. Finally, the CPU is not being overloaded, actually in the second phase the CPU usage is much less. Any help/suggestions are greatly appreciated. Thanks in advance!

1 0

Re: [alsa-devel] A reliable kernel panic (3.6.2) and system crash when visiting a particular website
by Daniel Mack 22 Oct '12

22 Oct '12

On 22.10.2012 17:17, Alan Stern wrote: > On Sun, 21 Oct 2012, Artem S. Tashkinov wrote: > >> dmesg messages up to a crash can be seen here: https://bugzilla.kernel.org/attachment.cgi?id=84221 > > The first problem in the log is endpoint list corruption. Here's a > debugging patch which should provide a little more information. Maybe add a BUG() after each of these dev_err() so we stop at the first occurance and also see where we're coming from? > drivers/usb/core/hcd.c | 36 ++++++++++++++++++++++++++++++++++++ > 1 file changed, 36 insertions(+) > > Index: usb-3.6/drivers/usb/core/hcd.c > =================================================================== > --- usb-3.6.orig/drivers/usb/core/hcd.c > +++ usb-3.6/drivers/usb/core/hcd.c > @@ -1083,6 +1083,8 @@ EXPORT_SYMBOL_GPL(usb_calc_bus_time); > > /*-------------------------------------------------------------------------*/ > > +static bool list_error; > + > /** > * usb_hcd_link_urb_to_ep - add an URB to its endpoint queue > * @hcd: host controller to which @urb was submitted > @@ -1126,6 +1128,20 @@ int usb_hcd_link_urb_to_ep(struct usb_hc > */ > if (HCD_RH_RUNNING(hcd)) { > urb->unlinked = 0; > + > + { > + struct list_head *cur = &urb->ep->urb_list; > + struct list_head *prev = cur->prev; > + > + if (prev->next != cur && !list_error) { > + list_error = true; > + dev_err(&urb->dev->dev, > + "ep %x list add corruption: %p %p %p\n", > + urb->ep->desc.bEndpointAddress, > + cur, prev, prev->next); > + } > + } > + > list_add_tail(&urb->urb_list, &urb->ep->urb_list); > } else { > rc = -ESHUTDOWN; > @@ -1193,6 +1209,26 @@ void usb_hcd_unlink_urb_from_ep(struct u > { > /* clear all state linking urb to this dev (and hcd) */ > spin_lock(&hcd_urb_list_lock); > + { > + struct list_head *cur = &urb->urb_list; > + struct list_head *prev = cur->prev; > + struct list_head *next = cur->next; > + > + if (prev->next != cur && !list_error) { > + list_error = true; > + dev_err(&urb->dev->dev, > + "ep %x list del corruption prev: %p %p %p\n", > + urb->ep->desc.bEndpointAddress, > + cur, prev, prev->next); > + } > + if (next->prev != cur && !list_error) { > + list_error = true; > + dev_err(&urb->dev->dev, > + "ep %x list del corruption next: %p %p %p\n", > + urb->ep->desc.bEndpointAddress, > + cur, next, next->prev); > + } > + } > list_del_init(&urb->urb_list); > spin_unlock(&hcd_urb_list_lock); > } >

1 0

[alsa-devel] [PATCH repost] ASoC: fsl/pcm030 add unique dai_link names
by Eric Millbrandt 22 Oct '12

22 Oct '12

Assign each dai_link a unique name to avoid this run-time error. [ 18.978043] pcm030-audio-fabric sound.2: wm9712-hifi <-> mpc5200-psc-ac97.0 mapping ok [ 19.003179] sysfs: cannot create duplicate filename '/devices/sound.2/AC97' Signed-off-by: Eric Millbrandt <emillbrandt(a)dekaresearch.com> --- sound/soc/fsl/pcm030-audio-fabric.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 4b63ec8..4d26192 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -29,14 +29,14 @@ struct pcm030_audio_data { static struct snd_soc_dai_link pcm030_fabric_dai[] = { { - .name = "AC97", + .name = "AC97.0", .stream_name = "AC97 Analog", .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "mpc5200-psc-ac97.0", .codec_name = "wm9712-codec", }, { - .name = "AC97", + .name = "AC97.1", .stream_name = "AC97 IEC958", .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", -- 1.7.8

2 1

Re: [alsa-devel] [PATCH] check for the kzalloc return value
by Takashi Iwai 22 Oct '12

22 Oct '12

At Mon, 22 Oct 2012 17:05:53 +0400, Denis Kirjanov wrote: > > > Signed-off-by: Denis Kirjanov <kirjanov(a)gmail.com> Thanks, applied. Takashi > --- > sound/pci/als300.c | 4 ++++ > 1 file changed, 4 insertions(+) > > diff --git a/sound/pci/als300.c b/sound/pci/als300.c > index 68c4469..e1f74d4 100644 > --- a/sound/pci/als300.c > +++ b/sound/pci/als300.c > @@ -394,6 +394,8 @@ static int snd_als300_playback_open(struct snd_pcm_substream *substream) > struct snd_als300_substream_data *data = kzalloc(sizeof(*data), > GFP_KERNEL); > > + if (!data) > + return -ENOMEM; > snd_als300_dbgcallenter(); > chip->playback_substream = substream; > runtime->hw = snd_als300_playback_hw; > @@ -425,6 +427,8 @@ static int snd_als300_capture_open(struct snd_pcm_substream *substream) > struct snd_als300_substream_data *data = kzalloc(sizeof(*data), > GFP_KERNEL); > > + if (!data) > + return -ENOMEM; > snd_als300_dbgcallenter(); > chip->capture_substream = substream; > runtime->hw = snd_als300_capture_hw; > -- > 1.7.9.5 >

1 0

[alsa-devel] [PATCH resend 1/2] ASoC: davinci: replace private sram api with genalloc
by Daniel Mack 22 Oct '12

22 Oct '12

From: Matt Porter <mporter(a)ti.com> Removes the DaVinci private SRAM API and replaces it with the genalloc API. The SRAM gen_pool is passed in pdata since DaVinci is in the early stages of DT conversion. [zonque(a)gmail.com: stub out gen_pool functions for !CONFIG_GENERIC_ALLOCATOR] Signed-off-by: Matt Porter <mporter(a)ti.com> --- include/linux/platform_data/davinci_asp.h | 3 +++ sound/soc/davinci/davinci-mcasp.c | 2 ++ sound/soc/davinci/davinci-pcm.c | 37 ++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.h | 2 ++ 4 files changed, 38 insertions(+), 6 deletions(-) diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index d0c5825..f3d6e4f 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -16,6 +16,8 @@ #ifndef __DAVINCI_ASP_H #define __DAVINCI_ASP_H +#include <linux/genalloc.h> + struct snd_platform_data { u32 tx_dma_offset; u32 rx_dma_offset; @@ -30,6 +32,7 @@ struct snd_platform_data { unsigned enable_channel_combine:1; unsigned sram_size_playback; unsigned sram_size_capture; + struct gen_pool *sram_pool; /* * If McBSP peripheral gets the clock from an external pin, diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 714e51e..52194bb 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1098,6 +1098,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_playback; dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + mem->start); @@ -1115,6 +1116,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_capture; dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + mem->start); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 93ea3bf..0ae7b6d 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -16,6 +16,7 @@ #include <linux/slab.h> #include <linux/dma-mapping.h> #include <linux/kernel.h> +#include <linux/genalloc.h> #include <sound/core.h> #include <sound/pcm.h> @@ -23,7 +24,6 @@ #include <sound/soc.h> #include <asm/dma.h> -#include <mach/sram.h> #include "davinci-pcm.h" @@ -259,7 +259,9 @@ static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) } } -static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, +#ifdef CONFIG_GENERIC_ALLOCATOR +static int allocate_sram(struct snd_pcm_substream *substream, + struct gen_pool *sram_pool, unsigned size, struct snd_pcm_hardware *ppcm) { struct snd_dma_buffer *buf = &substream->dma_buffer; @@ -271,9 +273,10 @@ static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, return 0; ppcm->period_bytes_max = size; - iram_virt = sram_alloc(size, &iram_phys); + iram_virt = (void *)gen_pool_alloc(sram_pool, size); if (!iram_virt) goto exit1; + iram_phys = gen_pool_virt_to_phys(sram_pool, (unsigned)iram_virt); iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); if (!iram_dma) goto exit2; @@ -285,11 +288,33 @@ static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, return 0; exit2: if (iram_virt) - sram_free(iram_virt, size); + gen_pool_free(sram_pool, (unsigned)iram_virt, size); exit1: return -ENOMEM; } +static void davinci_free_sram(struct snd_pcm_substream *substream, + struct snd_dma_buffer *iram_dma) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct gen_pool *sram_pool = prtd->params->sram_pool; + + gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes); +} +#else +static int allocate_sram(struct snd_pcm_substream *substream, + struct gen_pool *sram_pool, unsigned size, + struct snd_pcm_hardware *ppcm) +{ + return 0; +} + +static void davinci_free_sram(struct snd_pcm_substream *substream, + struct snd_dma_buffer *iram_dma) +{ +} +#endif + /* * Only used with ping/pong. * This is called after runtime->dma_addr, period_bytes and data_type are valid @@ -676,7 +701,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? &pcm_hardware_playback : &pcm_hardware_capture; - allocate_sram(substream, params->sram_size, ppcm); + allocate_sram(substream, params->sram_pool, params->sram_size, ppcm); snd_soc_set_runtime_hwparams(substream, ppcm); /* ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, @@ -819,7 +844,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) buf->area = NULL; iram_dma = buf->private_data; if (iram_dma) { - sram_free(iram_dma->area, iram_dma->bytes); + davinci_free_sram(substream, iram_dma); kfree(iram_dma); } } diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fc4d01c..b6ef703 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,6 +12,7 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <linux/genalloc.h> #include <linux/platform_data/davinci_asp.h> #include <mach/edma.h> @@ -20,6 +21,7 @@ struct davinci_pcm_dma_params { unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ unsigned sram_size; + struct gen_pool *sram_pool; /* SRAM gen_pool for ping pong */ enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ unsigned char data_type; /* xfer data type */ -- 1.7.11.7

2 2

[alsa-devel] [PATCH] ASoC: jack: remove unused variable in snd_soc_jack_report()
by Wei Yongjun 22 Oct '12

22 Oct '12

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> The variable oldstatus is initialized but never used otherwise, so remove the unused variable. dpatch engine is used to auto generate this patch. (https://github.com/weiyj/dpatch) Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/soc-jack.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fa0fd8d..933eb25 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -66,7 +66,6 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; int enable; - int oldstatus; trace_snd_soc_jack_report(jack, mask, status); @@ -78,8 +77,6 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) mutex_lock(&jack->mutex); - oldstatus = jack->status; - jack->status &= ~mask; jack->status |= status & mask;

2 1

[alsa-devel] [PATCH -next] ASoC: fix wm2200.c printk format warnings
by Randy Dunlap 22 Oct '12

22 Oct '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix wm2200.c printk format warnings (seen on x86_64): sound/soc/codecs/wm2200.c:1027:4: warning: format '%d' expects type 'int', but argument 4 has type 'size_t' sound/soc/codecs/wm2200.c:1139:5: warning: format '%d' expects type 'int', but argument 5 has type 'long unsigned int' sound/soc/codecs/wm2200.c:1181:2: warning: format '%d' expects type 'int', but argument 7 has type 'size_t' sound/soc/codecs/wm2200.c:1201:5: warning: format '%x' expects type 'unsigned int', but argument 3 has type 'long unsigned int' sound/soc/codecs/wm2200.c:1264:4: warning: format '%d' expects type 'int', but argument 4 has type 'size_t' sound/soc/codecs/wm2200.c:1328:5: warning: format '%d' expects type 'int', but argument 5 has type 'long unsigned int' Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org (moderated for non-subscribers) --- sound/soc/codecs/wm2200.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) --- linux-next-20121019.orig/sound/soc/codecs/wm2200.c +++ linux-next-20121019/sound/soc/codecs/wm2200.c @@ -1023,7 +1023,7 @@ static int wm2200_dsp_load(struct snd_so pos = sizeof(*header) + sizeof(*adsp1_sizes) + sizeof(*footer); if (pos >= firmware->size) { - dev_err(codec->dev, "%s: file too short, %d bytes\n", + dev_err(codec->dev, "%s: file too short, %zu bytes\n", file, firmware->size); return -EINVAL; } @@ -1135,7 +1135,7 @@ static int wm2200_dsp_load(struct snd_so } if (pos > firmware->size) - dev_warn(codec->dev, "%s.%d: %d bytes at end of file\n", + dev_warn(codec->dev, "%s.%d: %zu bytes at end of file\n", file, regions, pos - firmware->size); out: @@ -1178,7 +1178,7 @@ static int wm2200_setup_algs(struct snd_ } algs = be32_to_cpu(id.algs); - dev_info(codec->dev, "Firmware: %x v%d.%d.%d, %d algorithms\n", + dev_info(codec->dev, "Firmware: %x v%d.%d.%d, %zu algorithms\n", be32_to_cpu(id.fw.id), (be32_to_cpu(id.fw.ver) & 0xff000) >> 16, (be32_to_cpu(id.fw.ver) & 0xff00) >> 8, @@ -1196,7 +1196,7 @@ static int wm2200_setup_algs(struct snd_ } if (be32_to_cpu(val) != 0xbedead) - dev_warn(codec->dev, "Algorithm list end %x 0x%x != 0xbeadead\n", + dev_warn(codec->dev, "Algorithm list end %zx 0x%x != 0xbeadead\n", (sizeof(id) + (algs * sizeof(*alg))) / 2, be32_to_cpu(val)); @@ -1260,7 +1260,7 @@ static int wm2200_load_coeff(struct snd_ } if (sizeof(*hdr) >= firmware->size) { - dev_err(codec->dev, "%s: file too short, %d bytes\n", + dev_err(codec->dev, "%s: file too short, %zu bytes\n", file, firmware->size); return -EINVAL; } @@ -1324,7 +1324,7 @@ static int wm2200_load_coeff(struct snd_ } if (pos > firmware->size) - dev_warn(codec->dev, "%s.%d: %d bytes at end of file\n", + dev_warn(codec->dev, "%s.%d: %zu bytes at end of file\n", file, blocks, pos - firmware->size); return 0;

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Re: [alsa-devel] [PATCH 2/2] ASoC: Ux500: Control apb clock
by Mark Brown 22 Oct '12

22 Oct '12

On Mon, Oct 22, 2012 at 02:32:05PM +0200, Ulf Hansson wrote: > From: Ulf Hansson <ulf.hansson(a)linaro.org> > > When switching to common clock driver for ux500 this clock needs to > be handled as well. Before this clock was internally managed by the > clock driver itself. Applied but why isn't this a deficiency in the common clock implementation for the platform?

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Re: [alsa-devel] [PATCH 1/2] ASoC: Ux500: Fixup use of clocks
by Mark Brown 22 Oct '12

22 Oct '12

On Mon, Oct 22, 2012 at 02:32:04PM +0200, Ulf Hansson wrote: > From: Ulf Hansson <ulf.hansson(a)linaro.org> > > Make sure clocks are being prepared and unprepared as well > as enabled and disabled. Applied, thanks.

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Re: [alsa-devel] [PATCH 2/2] ASoC: Ux500: Control apb clock
by Linus Walleij 22 Oct '12

22 Oct '12

On Mon, Oct 22, 2012 at 2:32 PM, Ulf Hansson <ulf.hansson(a)stericsson.com> wrote: > From: Ulf Hansson <ulf.hansson(a)linaro.org> > > When switching to common clock driver for ux500 this clock needs to > be handled as well. Before this clock was internally managed by the > clock driver itself. > > Signed-off-by: Ulf Hansson <ulf.hansson(a)linaro.org> Looks correct, Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Yours, Linus Walleij

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Re: [alsa-devel] [PATCH 1/2] ASoC: Ux500: Fixup use of clocks
by Linus Walleij 22 Oct '12

22 Oct '12

On Mon, Oct 22, 2012 at 2:32 PM, Ulf Hansson <ulf.hansson(a)stericsson.com> wrote: > From: Ulf Hansson <ulf.hansson(a)linaro.org> > > Make sure clocks are being prepared and unprepared as well > as enabled and disabled. > > Signed-off-by: Ulf Hansson <ulf.hansson(a)linaro.org> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Thanks, Linus Walleij

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[alsa-devel] dynamic PCM and dynamic DSP graphs
by Sebastien LEDUC 22 Oct '12

22 Oct '12

Hi Mark / Liam, We have started to use Dynamic PCM to manage our on-SoC DSP. In our case, the API exposed by the DSP is graph oriented, and allows to dynamically create and control the running graph on the DSP. What we are doing is that we use DAPM widgets and routes to manage the DSP graph dynamically: We use some dedicated widgets that have power callbacks. When a widget is powered on/off, the callback will use the DSP API to create/destroy the corresponding parts of the DSP graph. We wanted to get your feedback on this approach, so as to be sure it won't be an issue when we want to mainline our driver. We can of course share more details if needed, or even some preliminary version of the driver. Thanks in advance for your feedback Regards Sebastien

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[alsa-devel] [sound-unstable:topic/hda-spdif-fix 99/124] ERROR: "snd_dmaengine_pcm_open" [sound/soc/omap/snd-soc-omap.ko] undefined!
by Fengguang Wu 22 Oct '12

22 Oct '12

Hi Peter, FYI, kernel build failed on tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git topic/hda-spdif-fix head: afc51e492b36bd61b46f4d70dd646678d96f5f97 commit: e7736cdea223f3a5b867c359fb35cf08250dd771 [99/124] dmaengine: Add flags parameter to dmaengine_prep_dma_cyclic() config: arm-omap2plus_defconfig # make ARCH=arm omap2plus_defconfig All error/warnings: ERROR: "snd_hwparams_to_dma_slave_config" [sound/soc/omap/snd-soc-omap.ko] undefined! ERROR: "snd_dmaengine_pcm_pointer" [sound/soc/omap/snd-soc-omap.ko] undefined! ERROR: "snd_dmaengine_pcm_open" [sound/soc/omap/snd-soc-omap.ko] undefined! ERROR: "snd_dmaengine_pcm_close" [sound/soc/omap/snd-soc-omap.ko] undefined! ERROR: "snd_dmaengine_pcm_get_chan" [sound/soc/omap/snd-soc-omap.ko] undefined! ERROR: "snd_dmaengine_pcm_trigger" [sound/soc/omap/snd-soc-omap.ko] undefined! --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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Re: [alsa-devel] [PATCH] ALSA: hda - remove unused variable in azx_position_ok()
by Takashi Iwai 22 Oct '12

22 Oct '12

At Mon, 22 Oct 2012 16:30:10 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > The variable stream is initialized but never used > otherwise, so remove the unused variable. > > dpatch engine is used to auto generate this patch. > (https://github.com/weiyj/dpatch) > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Thanks, applied. Takashi > --- > sound/pci/hda/hda_intel.c | 2 -- > 1 file changed, 2 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index 72b085a..0ceed85 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -2193,13 +2193,11 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) > { > u32 wallclk; > unsigned int pos; > - int stream; > > wallclk = azx_readl(chip, WALLCLK) - azx_dev->start_wallclk; > if (wallclk < (azx_dev->period_wallclk * 2) / 3) > return -1; /* bogus (too early) interrupt */ > > - stream = azx_dev->substream->stream; > pos = azx_get_position(chip, azx_dev, true); > > if (WARN_ONCE(!azx_dev->period_bytes, >

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Re: [alsa-devel] [PATCH 2/2] ASoC: Ux500: Fixup compile error
by Lee Jones 22 Oct '12

22 Oct '12

On Fri, 19 Oct 2012, Ulf Hansson wrote: > From: Ulf Hansson <ulf.hansson(a)linaro.org> > > The below commit introduced a compile error for a missing include file. > ASoC: ux500_msp_i2s: better use devm functions and fix error return code Already fixed: https://patchwork.kernel.org/patch/1593881/ -- Lee Jones Linaro ST-Ericsson Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

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Re: [alsa-devel] [PATCH 1/2] ASoC: Ux500: Fixup complile errors due to merge
by Lee Jones 22 Oct '12

22 Oct '12

On Fri, 19 Oct 2012, Ulf Hansson wrote: > From: Ulf Hansson <ulf.hansson(a)linaro.org> > > Likely during merge of the below commits ended up breaking compilation: > ASoC: Ux500: Enable ux500 MSP driver for Device Tree > ASoC: ux500_msp_i2s: better use devm functions and fix error return code This is already fixed: https://patchwork.kernel.org/patch/1593881/ -- Lee Jones Linaro ST-Ericsson Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

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Re: [alsa-devel] A reliable kernel panic (3.6.2) and system crash when visiting a particular website
by Daniel Mack 21 Oct '12

21 Oct '12

On 21.10.2012 22:43, Artem S. Tashkinov wrote: >> Nice. Could you do that again with the patch applied I sent yo some >> hours ago? > > That patch was of no help - the system has crashed and I couldn't spot relevant > messages. > > I've no idea what it means. The sequence of driver callbacks issued on a stream start is .open() .hw_params() .prepare() .trigger() If the ALSA part really causes this issue, the bad things happen either in any of the driver callback functions or in the core underneath. The patch I sent returns an error from the hw_params callback, and as you still see the problem, that means that the crash happens before any of the USB audio streaming really starts. Could you try and return -EINVAL from snd_usb_capture_open() please? If anyone has a better idea on how to debug this, please chime in. Daniel

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Re: [alsa-devel] A reliable kernel panic (3.6.2) and system crash when visiting a particular website
by Daniel Mack 21 Oct '12

21 Oct '12

On 21.10.2012 21:49, Artem S. Tashkinov wrote: >> >> On Oct 21, 2012, Borislav Petkov <bp(a)alien8.de> wrote: >> >> On Sun, Oct 21, 2012 at 11:59:36AM +0000, Artem S. Tashkinov wrote: >>> http://imageshack.us/a/img685/9452/panicz.jpg >>> >>> list_del corruption. prev->next should be ... but was ... >> >> Btw, this is one of the debug options I told you to enable. >> >>> I cannot show you more as I have no serial console to use :( and the kernel >>> doesn't have enough time to push error messages to rsyslog and fsync >>> /var/log/messages >> >> I already told you how to catch that oops: boot with "pause_on_oops=600" >> on the kernel command line and photograph the screen when the first oops >> happens. This'll show us where the problem begins. > > This option didn't have any effect, or maybe it's because it's such a serious crash > the kernel has no time to actually print an ooops/panic message. > > dmesg messages up to a crash can be seen here: https://bugzilla.kernel.org/attachment.cgi?id=84221 Nice. Could you do that again with the patch applied I sent yo some hours ago? Thanks, Daniel

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Re: [alsa-devel] was: Re: A reliable kernel panic (3.6.2) and system crash when visiting a particular website
by Daniel Mack 21 Oct '12

21 Oct '12

On Oct 21, 2012 5:28 PM, "Alan Stern" <stern(a)rowland.harvard.edu> wrote: > > On Sun, 21 Oct 2012, Daniel Mack wrote: > > > As the usb list is still in Cc: - Artem's lcpci dump shows that his > > machine features XHCI controllers. Can anyone think of a relation to > > this problem? > > > > And Artem, is there any way you boot your system on an older machine > > that only has EHCI ports? Thinking about it, I wonder whether the freeze > > in VBox and the crashes on native hardware have the same root cause. In > > that case, would it be possible to share that VBox image? > > Don't grasp at straws. All of the kernel logs Artem has posted show > ehci-hcd; none of them show xhci-hcd. Therefore the xHCI controller is > highly unlikely to be involved. You might be right - I'm just looking for differences between his setup and mine that would explain why nobody else sees a severe bug that is 100% reproducible for him.

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Re: [alsa-devel] was: Re: A reliable kernel panic (3.6.2) and system crash when visiting a particular website
by Daniel Mack 21 Oct '12

21 Oct '12

On 21.10.2012 16:57, Artem S. Tashkinov wrote: >> On Oct 21, 2012, Daniel Mack wrote: >> >> [Cc: alsa-devel] >> >> On 21.10.2012 14:30, Artem S. Tashkinov wrote: >>> On Oct 21, 2012, Daniel Mack wrote: >>> >>>> A hint at least. How did you enable the audio record exactly? Can you >>>> reproduce this with arecord? >>>> >>>> What chipset are you on? Please provide both "lspci -v" and "lsusb -v" >>>> dumps. As I said, I fail to reproduce that issue on any of my machines. >>> >>> All other applications can read from the USB audio without problems, it's >>> just something in the way Adobe Flash polls my audio input which causes >>> a crash. >>> >>> Just video capture (without audio) works just fine in Adobe Flash. >> >> Ok, so that pretty much rules out the host controller. I just wonder why >> I still don't see it here, and I haven't heard of any such problem from >> anyone else. >> >> Some more questions: >> >> - Which version of Flash are you running? > > Google Chrome has its own version of Adobe Flash: > > Name: Shockwave Flash > Description: Shockwave Flash 11.4 r31 > Version: 11.4.31.110 So that's the same that I'm using. >> - Does this also happen with Firefox? > > No, Adobe Flash in Firefox is an older version (Shockwave Flash 11.1 r102), it shows > just two input devices instead of three which the newer Flash players sees. > > * HDA Intel PCH > * USB Device 0x46d:0x81d And that works, I assume? Does the second choice in the newer Flash version work maybe? >> - Does flash access the device directly or via PulseAudio? > > PA is not installed on my computer, so Flash accesses it directly via ALSA calls. Ok, Same here. >> - Could you please apply the attached patch and see what it spits out to >> dmesg once Flash opens the device? It returns -EINVAL in the hw_params >> callback to prevent the actual streaming. On my machine with Flash >> 11.4.31.110, I get values of 2/44800/1/32768/2048/0, which seems sane. >> Or does your machine still crash before anything is written to the logs? > > I will try it a bit later. Yes, we need to trace the call chain and see at which point the trouble starts. What could help is tracing the google-chrome binary with strace maybe. At least we would see the ioctl command sequence, if the log file survives the crash. As the usb list is still in Cc: - Artem's lcpci dump shows that his machine features XHCI controllers. Can anyone think of a relation to this problem? And Artem, is there any way you boot your system on an older machine that only has EHCI ports? Thinking about it, I wonder whether the freeze in VBox and the crashes on native hardware have the same root cause. In that case, would it be possible to share that VBox image? Daniel

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Re: [alsa-devel] was: Re: A reliable kernel panic (3.6.2) and system crash when visiting a particular website
by Daniel Mack 21 Oct '12

21 Oct '12

[Cc: alsa-devel] On 21.10.2012 14:30, Artem S. Tashkinov wrote: > On Oct 21, 2012, Daniel Mack wrote: > >> A hint at least. How did you enable the audio record exactly? Can you >> reproduce this with arecord? >> >> What chipset are you on? Please provide both "lspci -v" and "lsusb -v" >> dumps. As I said, I fail to reproduce that issue on any of my machines. > > All other applications can read from the USB audio without problems, it's > just something in the way Adobe Flash polls my audio input which causes > a crash. > > Just video capture (without audio) works just fine in Adobe Flash. Ok, so that pretty much rules out the host controller. I just wonder why I still don't see it here, and I haven't heard of any such problem from anyone else. Some more questions: - Which version of Flash are you running? - Does this also happen with Firefox? - Does flash access the device directly or via PulseAudio? - Could you please apply the attached patch and see what it spits out to dmesg once Flash opens the device? It returns -EINVAL in the hw_params callback to prevent the actual streaming. On my machine with Flash 11.4.31.110, I get values of 2/44800/1/32768/2048/0, which seems sane. Or does your machine still crash before anything is written to the logs? > Only and only when I choose to use > > USB Device 0x46d:0x81d my system crashes in Adobe Flash. > > See the screenshot: > > https://bugzilla.kernel.org/attachment.cgi?id=84151 When exactly does the crash happen? Right after you selected that entry from the list? There's a little recording level meter in that dialog. Does that show any input from the microphone? > My hardware information can be fetched from here: > > https://bugzilla.kernel.org/show_bug.cgi?id=49181 > > On a second thought that can be even an ALSA crash or pretty much > anything else. We'll see. Thanks for your help to sort this out! Daniel

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[alsa-devel] usb-audio: Reloop Play support (TI TUSB3200AC)
by Didier 'Ptitjes' Villevalois 21 Oct '12

21 Oct '12

Hello, I've bought a Reloop Play usb soundcard and I'd like to make it work with ALSA. It is a 4-channels USB soundcard. It is currently correctly recognized by ALSA as a 4-channels soundcard and I can correctly open the channels and feed them with audio. However, no audio is output by the box. It seems to contain a Texas Instrument TUSB3200AC chipset (same chipset as the Maya44 USB and Hercules DJ console). I guess the problem is a firmware problem but I don't know what to do about it. I've got a MacOS X driver that works correctly, if that of any help. (I suppose the Windows driver works too but I don't have any Windows at hand.) I'm willing to do whatever you need me to make it run with ALSA. Could you help me ? I attached my alsa-info.sh output. Didier Villevalois.

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[alsa-devel] softvol master has preference over virtual master in amixer
by Raymond Yau 21 Oct '12

21 Oct '12

does softvol plugin "master" has any preference over vmaster in amixer ? http://www.alsa-project.org/db/?f=44b26f5cd37f33bfef2871ff1fd250301bdb5364 control.32 { iface MIXER name 'Master Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.42 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } Card hw:0 'Intel'/'HDA Intel at 0xfeaf8000 irq 43' Mixer name: 'Realtek ALC887-VD' Components: 'HDA:10ec0887,10438445,00100302' Controls : 42 Simple ctrls : 22 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [191.00dB] [on] Front Right: Playback 255 [100%] [191.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] https://launchpadlibrarian.net/119782219/Symptom_PulseAudioLog.txt Oct 15 10:49:11 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:11 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:11 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:11 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument Oct 15 10:49:12 xaser pulseaudio[1601]: [pulseaudio] alsa-util.c: Unable to load mixer: Invalid argument

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[alsa-devel] [PATCH 1/5] sound/isa/opti9xx/miro.c: eliminate possible double free
by Julia Lawall 21 Oct '12

21 Oct '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> snd_miro_probe is a static function that is only called twice in the file that defines it. At each call site, its argument is freed using snd_card_free. Thus, there is no need for snd_miro_probe to call snd_card_free on its argument on any of its error exit paths. Because snd_card_free both reads the fields of its argument and kfrees its argments, the results of the second snd_card_free should be unpredictable. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r@ identifier f,free,a; parameter list[n] ps; type T; expression e; @@ f(ps,T a,...) { ... when any when != a = e if(...) { ... free(a); ... return ...; } ... when any } @@ identifier r.f,r.free; expression x,a; expression list[r.n] xs; @@ * x = f(xs,a,...); if (...) { ... free(a); ... return ...; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- Not tested. sound/isa/opti9xx/miro.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 3d1afb6..4a7ff4e 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1286,7 +1286,6 @@ static int __devinit snd_miro_probe(struct snd_card *card) error = snd_card_miro_aci_detect(card, miro); if (error < 0) { - snd_card_free(card); snd_printk(KERN_ERR "unable to detect aci chip\n"); return -ENODEV; }

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[alsa-devel] [PATCH 1/1] Add USB audio quirk supporting Roland VG-99 advanced mode
by pete.leigh＠gmail.com 21 Oct '12

21 Oct '12

From: Pete Leigh <pete.leigh(a)gmail.com> Signed-off-by: Pete Leigh <pete.leigh(a)gmail.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 88d8ceb..ffeabf8 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1457,6 +1457,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Advanced mode of the Roland VG-99, with MIDI and 24-bit PCM at 44.1 + * kHz. In standard mode, the device has ID 0582:00b3, and offers + * 16-bit PCM at 44.1 kHz with no MIDI. + */ + USB_DEVICE(0x0582, 0x00b2), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "VG-99", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ /* Roland SonicCell */ USB_DEVICE(0x0582, 0x00c2), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- 1.7.11.7

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[alsa-devel] [PATCH 1/1] USB audio quirk for Roland VG-99 advanced mode
by pete.leigh＠gmail.com 21 Oct '12

21 Oct '12

From: Pete Leigh <pete.leigh(a)gmail.com> Without this quirk the VG-99 will work in standard mode (set under USB on System menu page 2) giving 16 bits at 44.1 Khz audio in/out but no midi, and is not recognised when set to advanced mode. After applying this, I can also use the VG-99 in advanced mode: 24 24 bits audio in/out at 44.1 Khz, and midi in/out. Sysex is so far untested. In standard mode, the device appears with ID 0x00b3, so the behaviour isn't affected by this quirk. Thanks to Clemens Ladisch for simplifying and correcting my initial attempt! Signed-off-by: Pete Leigh <pete.leigh(a)gmail.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 88d8ceb..ffeabf8 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1457,6 +1457,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Advanced mode of the Roland VG-99, with MIDI and 24-bit PCM at 44.1 + * kHz. In standard mode, the device has ID 0582:00b3, and offers + * 16-bit PCM at 44.1 kHz with no MIDI. + */ + USB_DEVICE(0x0582, 0x00b2), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "VG-99", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ /* Roland SonicCell */ USB_DEVICE(0x0582, 0x00c2), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- 1.7.11.7

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[alsa-devel] [PATCH - my module 1/1] USB audio quirk for Roland VG-99 advanced mode
by pete.leigh＠gmail.com 21 Oct '12

21 Oct '12

From: Pete Leigh <pete.leigh(a)gmail.com> Without this quirk the VG-99 will work in standard mode (set under USB on System menu page 2) giving 16 bits at 44.1 Khz audio in/out but no midi, and is not recognised when set to advanced mode. After applying this, I can also use the VG-99 in advanced mode: 24 24 bits audio in/out at 44.1 Khz, and midi in/out. Sysex is so far untested. In standard mode, the device appears with ID 0x00b3, so the behaviour isn't affected by this quirk. Thanks to Clemens Ladisch for simplifying and correcting my initial attempt! Signed-off-by: Pete Leigh <pete.leigh(a)gmail.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 88d8ceb..ffeabf8 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1457,6 +1457,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Advanced mode of the Roland VG-99, with MIDI and 24-bit PCM at 44.1 + * kHz. In standard mode, the device has ID 0582:00b3, and offers + * 16-bit PCM at 44.1 kHz with no MIDI. + */ + USB_DEVICE(0x0582, 0x00b2), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "VG-99", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ /* Roland SonicCell */ USB_DEVICE(0x0582, 0x00c2), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- 1.7.11.7

2 1

[alsa-devel] [PATCH] ALSA: aloop - do not reschedule timer if deleted
by Omair Mohammed Abdullah 21 Oct '12

21 Oct '12

If the timer is deleted while the timer handler is running, it may get rescheduled and an unnecessary period elapsed will be sent. Add a flag to reschedule the timer only if it has not been stopped. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/drivers/aloop.c | 21 ++++++++++++++++++--- 1 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 0fe6d64..fec4e17 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -121,6 +121,7 @@ struct loopback_pcm { unsigned long last_jiffies; struct timer_list timer; spinlock_t timer_lock; + bool timer_stopped; }; static struct platform_device *devices[SNDRV_CARDS]; @@ -166,12 +167,11 @@ static inline unsigned int get_rate_shift(struct loopback_pcm *dpcm) return get_setup(dpcm)->rate_shift; } -static void loopback_timer_start(struct loopback_pcm *dpcm) +static void do_timer_start(struct loopback_pcm *dpcm) { unsigned long tick; unsigned int rate_shift = get_rate_shift(dpcm); - spin_lock(&dpcm->timer_lock); if (rate_shift != dpcm->pcm_rate_shift) { dpcm->pcm_rate_shift = rate_shift; dpcm->period_size_frac = frac_pos(dpcm, dpcm->pcm_period_size); @@ -184,6 +184,13 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) tick = (tick + dpcm->pcm_bps - 1) / dpcm->pcm_bps; dpcm->timer.expires = jiffies + tick; add_timer(&dpcm->timer); +} + +static inline void loopback_timer_start(struct loopback_pcm *dpcm) +{ + spin_lock(&dpcm->timer_lock); + dpcm->timer_stopped = false; + do_timer_start(dpcm); spin_unlock(&dpcm->timer_lock); } @@ -192,6 +199,7 @@ static inline void loopback_timer_stop(struct loopback_pcm *dpcm) spin_lock(&dpcm->timer_lock); del_timer(&dpcm->timer); dpcm->timer.expires = 0; + dpcm->timer_stopped = true; spin_unlock(&dpcm->timer_lock); } @@ -540,7 +548,13 @@ static void loopback_timer_function(unsigned long data) running = loopback_pos_update(dpcm->cable); if (running & (1 << dpcm->substream->stream)) { - loopback_timer_start(dpcm); + spin_lock(&dpcm->timer_lock); + if (dpcm->timer_stopped) { + spin_unlock(&dpcm->timer_lock); + return; + } + do_timer_start(dpcm); + spin_unlock(&dpcm->timer_lock); if (dpcm->period_update_pending) { dpcm->period_update_pending = 0; snd_pcm_period_elapsed(dpcm->substream); @@ -670,6 +684,7 @@ static int loopback_open(struct snd_pcm_substream *substream) } dpcm->loopback = loopback; dpcm->substream = substream; + dpcm->timer_stopped = true; setup_timer(&dpcm->timer, loopback_timer_function, (unsigned long)dpcm); spin_lock_init(&dpcm->timer_lock); -- 1.7.4.1

2 3

[alsa-devel] [PATCH] ALSA: snd-usb: remove unused variable in init_pitch_v2()
by Wei Yongjun 21 Oct '12

21 Oct '12

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> The variable ep is initialized but never used otherwise, so remove the unused variable. dpatch engine is used to auto generate this patch. (https://github.com/weiyj/dpatch) Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/usb/pcm.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 55e19e1..f77b87a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -171,11 +171,8 @@ static int init_pitch_v2(struct snd_usb_audio *chip, int iface, { struct usb_device *dev = chip->dev; unsigned char data[1]; - unsigned int ep; int err; - ep = get_endpoint(alts, 0)->bEndpointAddress; - data[0] = 1; if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,

2 1

[alsa-devel] calling snd_ctl_notify from mixer ctl_put
by Ondrej Zary 21 Oct '12

21 Oct '12

Hello, I want to enable/disable some mixer controls based on state of another control (switching between limiter and AGC mode). So I (re)set the SNDRV_CTL_ELEM_ACCESS_INACTIVE flag in kctl->vd[0].access and then call snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &kctl->id); Everyting from ctl_put() mixer callback. When changing the control value in alsamixer, it crashes with: alsamixer: simple.c:948: snd_mixer_selem_get_enum_item: Assertion `(elem)->type == SND_MIXER_ELEM_SIMPLE' failed. When the snd_ctl_notify() call is commented-out, it does not crash. There is snd_ctl_activate_id() function in sound/core/control.c (not used by any driver, btw.) which does the same, protected by card->controls_rwsem. However, it locks up on that semaphore when called from the mixer callback. Oxygen/Xonar WM8776 driver is doing the same thing - I wonder if I'm doing something wrong or if that driver crashes alsamixer too? Is there any way to do this properly? -- Ondrej Zary

3 3

Re: [alsa-devel] usb-audio: Reloop Play support (TI TUSB3200AC)
by Daniel Mack 20 Oct '12

20 Oct '12

On 20.10.2012 16:24, Didier Villevalois wrote: > OK this quirk works perfectly! (hooray!!!) > > /* Reloop Play */ > { > USB_DEVICE(0x200c, 0x100b), > .bInterfaceClass = USB_CLASS_PER_INTERFACE, > .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { > .vendor_name = "Reloop", > .product_name = "Play", > .ifnum = QUIRK_ANY_INTERFACE, > .type = QUIRK_COMPOSITE, > .data = &(const struct snd_usb_audio_quirk[]) { > { > .ifnum = 0, > .type = QUIRK_AUDIO_STANDARD_MIXER, > }, > { > .ifnum = 1, > .type = QUIRK_AUDIO_FIXED_ENDPOINT, > .data = &(const struct audioformat) { > .formats = SNDRV_PCM_FMTBIT_S24_3LE, > .channels = 4, > .iface = 1, > .altsetting = 1, > .altset_idx = 1, > .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, > .endpoint = 0x01, > .rates = SNDRV_PCM_RATE_44100 | > SNDRV_PCM_RATE_48000, > .rate_min = 44100, > .rate_max = 48000, > .nr_rates = 2, > .rate_table = (unsigned int[]) { > 44100, 48000 > } > } > }, > { > .ifnum = -1 > } > } > } > }, > > Does anyone have additional comments to do before I make a patch ? Let's ask Clemens - the quirks master - for comments :) > Also are there some special ALSA rules or straight signed-off > git-formated patches over the kernel tree are OK ? That's ok. The patch should apply on top of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git though. > Thanks again for your help! I'm really happy to have a working > device! :D Good work! I just really wonder what that updated firmware is all about ... Daniel

1 0

[alsa-devel] [PATCH 0/9] Fixes for RME MADI cards
by Adrian Knoth 20 Oct '12

20 Oct '12

Hi! I've spent another afternoon at IOSONO to fix a series of bugs in the hdspm driver. Andre Schramm and me have tested almost all patches on a box with two PCIe MADI cards and one PCI MADI card. The DDS/Varispeed patch has only been tested on my RayDAT, the sync_in detection for AES32 has not been tested at all, though the documentation is pretty clear that the previous bit position was wrong. If there are AES(32) owners with two or more cards out there, please test and report back. The typo and the coding style patches have not been tested, since they're trivial. Cheers Adrian Knoth (9): ALSA: hdspm - Allow DDS/Varispeed to be set from userspace ALSA: hdspm - Report external rate in slave mode on PCI MADI ALSA: hdspm - Fix sync check reporting on all RME HDSPM cards ALSA: hdspm - Fix reported autosync_sample_rate ALSA: hdspm - Also report autosync_sample_rate on MADI and MADIface ALSA: hdspm - Fix sync_in reporting on RME MADI cards ALSA: hdspm - Fix sync_in detection on AES/AES32 ALSA: hdspm - Fix typo in kcontrol element on RME MADI cards ALSA: hdspm - Fix coding style in CTL_ELEM macros sound/pci/rme9652/hdspm.c | 232 ++++++++++++++++++++++++++------------------- 1 file changed, 137 insertions(+), 95 deletions(-) -- 1.7.10.4

2 13

[alsa-devel] Fwd: Support for Roland VG-99 (working code, but could use help)
by Pete Leigh 20 Oct '12

20 Oct '12

Hi, I tried posting this a couple of days ago, but no response... I'm wondering if the (text) attachment stopped people from seeing it, or if there's something else I missed? It's just a request to look over the lsusb output and new quirks-table entry included, with a view to getting this added to a release. Since the initial post, I've done more testing (I'm using a kernel based on 3.4.11-1.rt19.1.fc17.ccrma.x86_64.rt) and everything seems to work fine (including midi in, which I wasn't sure of at the time of posting) apart from one apparently cosmetic issue - the device name no longer shows up in lsusb output, though it did before applying the quirk. (?) I'll include the quirk entry as text rather than an attachment this time. If I actually need to go through the process of generating a patch and sending mail via git, as per the developer page on the alsa-project site, could someone let me know, and I'll give it a go (it's something I was hoping to avoid, I admit!) Thanks, - Pete. ---------- Forwarded message ---------- The VG-99 is a guitar processor with a usb interface which offers 2x16bit 44100 pcm in and out streams in its basic usb mode, and 2x24bit 44100 in/out + midi in the advanced mode connection mode. Basic mode works fine with usb snd driver, but the device isn't recognised when set to advanced mode (this is done under USB on page 2 of the "system" config menu) I've used the lsusb -v output (below) to guess at an addition to sound/usb/quirks-table.h to try and get this working. On initial (brief) testing, audio in/out at 24bits (as shown by jackd output) and midi (out, at least) are all working. I'll attach the quirk, and would be pleased if someone could look at it, and the lsusb output, and fix up any deficiencies (it's purely voodoo here :-D) and maybe get it or something like it into the official source... please let me know if changes need to be tested. Cheers, - Pete. Here's the lsusb output: Bus 001 Device 007: ID 0582:00b2 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 8 idVendor 0x0582 Roland Corp. idProduct 0x00b2 bcdDevice 1.00 iManufacturer 1 Roland iProduct 2 VG-99 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 2mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 03 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0120 1x 288 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 03 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0120 1x 288 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 02 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Device Status: 0x0001 Self Powered ------------------------------------ And the quirk entry: { USB_DEVICE(0x0582, 0x00b2), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Roland", .product_name = "VG-99", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 0, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x01, .ep_attr = 0x09, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 44100, .rate_max = 44100, } }, { .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX, .endpoint = 0x82, .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 44100, .rate_max = 44100, } }, { .ifnum = 2, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_cables = 0x0007, .in_cables = 0x0007 } }, { .ifnum = -1 } } } },

3 18

[alsa-devel] [PATCH v3 1/3] ALSA: core: keep track of boundary wrap-around
by Pierre-Louis Bossart 19 Oct '12

19 Oct '12

Keep track of boundary crossing when hw_ptr exceeds boundary limit and wraps-around. This will help keep track of total number of frames played/received at the kernel level Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 25 ++++++++++++++++++++----- 2 files changed, 21 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index cdca2ab..ba13e0b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -281,6 +281,7 @@ struct snd_pcm_runtime { unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ unsigned long hw_ptr_buffer_jiffies; /* buffer time in jiffies */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ + u64 hw_ptr_wrap; /* offset for hw_ptr due to boundary wrap-around */ /* -- HW params -- */ snd_pcm_access_t access; /* access mode */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7ae6719..bbc6ba4 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -315,6 +315,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, unsigned long jdelta; unsigned long curr_jiffies; struct timespec curr_tstamp; + int crossed_boundary = 0; old_hw_ptr = runtime->status->hw_ptr; @@ -359,8 +360,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, hdelta = curr_jiffies - runtime->hw_ptr_jiffies; if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) + if (hw_base >= runtime->boundary) { hw_base = 0; + crossed_boundary++; + } new_hw_ptr = hw_base + pos; goto __delta; } @@ -370,8 +373,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* pointer crosses the end of the ring buffer */ if (new_hw_ptr < old_hw_ptr) { hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) + if (hw_base >= runtime->boundary) { hw_base = 0; + crossed_boundary++; + } new_hw_ptr = hw_base + pos; } __delta: @@ -409,8 +414,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, while (hdelta > xrun_threshold) { delta += runtime->buffer_size; hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) + if (hw_base >= runtime->boundary) { hw_base = 0; + crossed_boundary++; + } new_hw_ptr = hw_base + pos; hdelta -= runtime->hw_ptr_buffer_jiffies; } @@ -455,8 +462,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* the delta value is small or zero in most cases */ while (delta > 0) { new_hw_ptr += runtime->period_size; - if (new_hw_ptr >= runtime->boundary) + if (new_hw_ptr >= runtime->boundary) { new_hw_ptr -= runtime->boundary; + crossed_boundary--; + } delta--; } /* align hw_base to buffer_size */ @@ -506,6 +515,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = curr_jiffies; + if (crossed_boundary) { + snd_BUG_ON(crossed_boundary != 1); + runtime->hw_ptr_wrap += runtime->boundary; + } if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) runtime->status->tstamp = curr_tstamp; @@ -1660,8 +1673,10 @@ static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream, if (snd_pcm_running(substream) && snd_pcm_update_hw_ptr(substream) >= 0) runtime->status->hw_ptr %= runtime->buffer_size; - else + else { runtime->status->hw_ptr = 0; + runtime->hw_ptr_wrap = 0; + } snd_pcm_stream_unlock_irqrestore(substream, flags); return 0; } -- 1.7.9.5

2 5

[alsa-devel] [PATCH 0/2] USB-audio disconnection race fixes (v2)
by Takashi Iwai 19 Oct '12

19 Oct '12

Hi, this is a series of fixes for the disconnection races in USB-audio. A couple of new patches to improve and cover more points since the previous patch set. Takashi

1 7

Re: [alsa-devel] [PATCH v2 1/1] ALSA: hda - add quirk for Thinkpad T430
by Jonathan Nieder 19 Oct '12

19 Oct '12

joseph.salisbury(a)canonical.com wrote: > This patch adds a quirk to enable headphone jack audio on T430 Thinkpads. > > Signed-off-by: Joseph Salisbury <joseph.salisbury(a)canonical.com> > Cc: <stable(a)vger.kernel.org> Thanks --- this is a little better. One more nit: it looks like this was reported, tested, and based on a patch by Stefan Freyr. Takashi, could you amend the message to reflect that, or should the patch be resent? Something like -- >8 -- From: Stefán Freyr <stefan.freyr(a)gmail.com> Subject: ALSA: hda - add dock support for Thinkpad T430 I have a Lenovo ThinkPad T430 and an UltraBase Series 3 docking station. Without this patch, if I plug my headphones into the jack on the computer, everything works fine. The computer speakers mute and the audio is played in the headphones. However, if I plug into the docking station headphone jack the computer speakers are muted but there is no audio in the headphones. Addresses https://bugs.launchpad.net/bugs/1060372 Signed-off-by: Joseph Salisbury <joseph.salisbury(a)canonical.com> Cc: stable(a)vger.kernel.org

2 1

[alsa-devel] HDSPM - Problems with SyncIn
by Andre Schramm 19 Oct '12

19 Oct '12

Hello, I did some tests with the SyncIn-Feature of the HDSPe cards. [root@localhost ~]# lspci |grep Mult 01:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP MADI (rev d2) 02:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP MADI (rev d2) The SyncOut of card 2:00 was plugged to the SyncIn of 1:00 and 1:00 was put to AutoSync-mode automatically upon boot. Problem is that the amixer output seems to be inconsistent and differs from the proc output as well, so I'm not sure which one to trust. [root@localhost ~]# amixer -c0 contents numid=5,iface=MIXER,name='AutoSync Reference' ; type=ENUMERATED,access=r-------,values=1,items=5 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'TCO' ; Item #3 'Sync In' ; Item #4 'None' : values=3 numid=2,iface=MIXER,name='Internal Clock' ; type=ENUMERATED,access=rw------,values=1,items=9 ; Item #0 '32 kHz' ; Item #1 '44.1 kHz' ; Item #2 '48 kHz' ; Item #3 '64 kHz' ; Item #4 '88.2 kHz' ; Item #5 '96 kHz' ; Item #6 '128 kHz' ; Item #7 '176.4 kHz' ; Item #8 '192 kHz' : values=2 numid=4,iface=MIXER,name='Preferred Sync Reference' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'Sync In' : values=2 numid=10,iface=MIXER,name='SYNC IN SyncCheck' ; type=ENUMERATED,access=r-------,values=1,items=4 ; Item #0 'No Lock' ; Item #1 'Lock' ; Item #2 'Sync' ; Item #3 'N/A' : values=0 numid=3,iface=MIXER,name='System Clock Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Master' ; Item #1 'AutoSync' : values=1 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=47999 'Preferred Sync Reference' is SyncIn, 'SYNC IN SyncCheck' says 'No Lock', meaning no signal at all, but the 'System Sample Rate' coming from the master card is detected with 1Hz offset. When putting the master to 32000Hz, 31999Hz are shown and so on. [root@localhost ~]# cat /proc/asound/card0/hdspm RME MADI (Card #1) Rev.d2 Status2first3bits: 1 HW Serial: 0x0000019196d4 IRQ: 16 Registers bus: 0xfe9f0000 VM: 0xf8920000 --- System --- IRQ Pending: Audio=0, MIDI0=0, MIDI1=0, IRQcount=0 HW pointer: id = 0, rawptr = 0 (0->16320) estimated= 0 (bytes) MIDI FIFO: Out1=0x0, Out2=0x0, In1=0x0, In2=0x0 MIDIoverMADI FIFO: In=0x0, Out=0x0 Register: ctrl1=0x10344ca, ctrl2=0x0, status1=0xc30004, status2=0xc0020361 No TCO module detected. --- Settings --- Size (Latency): 2048 samples (2 periods of 8192 bytes) Line out: on ClearTrackMarker = off, Transmit in 64 Channel Mode, Auto Input off AutoSync Reference: AutoSync Preferred Sync Reference: TCO System Clock Frequency: 48000 --- Status: Inputs MADI=NoLock, WordClock=NoLock AutoSync: Reference= Sync In, Freq=0 (MADI = 3, Word = 3) Input: Coax, Mode=56 channels Now the proc-output reports 'No TCO module detected', which is correct, as no expansion modules where plugged to the cards. But it also says 'Preferred Sync Reference: TCO' and 'Freq=0', where amixer said SyncIn and 47999? Output of the master card is below. Regards, Andre [root@localhost ~]# amixer -c1 contents numid=5,iface=MIXER,name='AutoSync Reference' ; type=ENUMERATED,access=r-------,values=1,items=5 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'TCO' ; Item #3 'Sync In' ; Item #4 'None' : values=4 numid=15,iface=MIXER,name='Input Select' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'optical' ; Item #1 'coaxial' : values=0 numid=2,iface=MIXER,name='Internal Clock' ; type=ENUMERATED,access=rw------,values=1,items=9 ; Item #0 '32 kHz' ; Item #1 '44.1 kHz' ; Item #2 '48 kHz' ; Item #3 '64 kHz' ; Item #4 '88.2 kHz' ; Item #5 '96 kHz' ; Item #6 '128 kHz' ; Item #7 '176.4 kHz' ; Item #8 '192 kHz' : values=2 numid=4,iface=MIXER,name='Preferred Sync Reference' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'Sync In' : values=0 numid=10,iface=MIXER,name='SYNC IN SyncCheck' ; type=ENUMERATED,access=r-------,values=1,items=4 ; Item #0 'No Lock' ; Item #1 'Lock' ; Item #2 'Sync' ; Item #3 'N/A' : values=0 numid=3,iface=MIXER,name='System Clock Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Master' ; Item #1 'AutoSync' : values=0 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=48000 [root@localhost ~]# cat /proc/asound/card1/hdspm RME MADI (Card #2) Rev.d2 Status2first3bits: 1 HW Serial: 0x000001b1c606 IRQ: 17 Registers bus: 0xfeaf0000 VM: 0xf9280000 --- System --- IRQ Pending: Audio=0, MIDI0=0, MIDI1=0, IRQcount=0 HW pointer: id = 0, rawptr = 0 (0->16320) estimated= 0 (bytes) MIDI FIFO: Out1=0x0, Out2=0x0, In1=0x0, In2=0x0 MIDIoverMADI FIFO: In=0x0, Out=0x0 Register: ctrl1=0x10004da, ctrl2=0x0, status1=0x2800000, status2=0xc0020701 No TCO module detected. --- Settings --- Size (Latency): 2048 samples (2 periods of 8192 bytes) Line out: on ClearTrackMarker = off, Transmit in 64 Channel Mode, Auto Input off AutoSync Reference: Master Preferred Sync Reference: Word Clock System Clock Frequency: 48000 --- Status: Inputs MADI=NoLock, WordClock=NoLock AutoSync: Reference= Input not valid, Freq=0 (MADI = 10, Word = 0) Input: Optical, Mode=56 channels

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Re: [alsa-devel] [PATCH 0/1] ALSA: hda - add quirk for Thinkpad T430
by Jonathan Nieder 19 Oct '12

19 Oct '12

Hi Joseph, joseph.salisbury(a)canonical.com wrote: > [Subject: [PATCH 0/1] ALSA: hda - add quirk for Thinkpad T430]] [...] > This patch is to enable audio on T430 Thinkpads. There is no audo > from the headphone jack without this patch. This patch adds a quirk > for the T430 model. Please don't do this cover-letter thing. A cover letter can be a good way to *summarize* a long patch series so readers are less daunted, but any information that was needed to know in order to evaluate a patch, briefly stated, should be in the commit message that actually gets committed. Thanks and hope that helps, Jonathan

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[alsa-devel] Non-interleaved with plughw
by Prasant J 19 Oct '12

19 Oct '12

Hi, My hardware supports only interleaved access and I'm recording and playing using hw:0,0. Currently I provide interleaved buffers to the device. What I read is that plughw device can convert non-interleaved buffers to interleave buffers before sending to hardware. I want to reduce complexity of my application and want to use plughw. But I'm not able to get good documentation or examples of the same. Can anyone point out to me? Regards, -Prasant

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Re: [alsa-devel] ASoC: mmap or read/write?
by Mike Looijmans 19 Oct '12

19 Oct '12

Clemens ------------------------------ Message: 6 Date: Thu, 18 Oct 2012 15:07:45 +0530 From: Prasant J <pj0585(a)gmail.com> To: Clemens Ladisch <clemens(a)ladisch.de> Cc: alsa-devel(a)alsa-project.org Subject: Re: [alsa-devel] ASoC: mmap or read/write? Message-ID: <CAA2DH4v4DUxf5Qh79nehTPD+22v6XddzDrEMmnBu2eM8m4JXzg(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Thu, Oct 18, 2012 at 2:39 PM, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> Prasant J wrote: >>> I'm writing an alsa application for recording & playing 16 channel audio. >>> >>> The application is intended to run on an embedded platform & the >>> resources are highly constrained. >> Then I wonder how you manage to handle 16 channels. >> What hardware is this? > I'm working with Beaglebone (ARM Cortex-A8 720 MHz). (Currently > failures are for 8 channels 32 bit / 16 bit, but our final target is > 16 channels) > >>> My application is not performing well with read/write methods. >> How much CPU usage? > The CPU usage is around 35% (as seen in top) > > I have managed to get Audio capture working fine (after certain linux > optimizations) but playback is giving under-runs (only when PLAYBACK & > CAPTURE run together) > I was wondering that changing access method to mmap (currently I'm > using read/write) may give me some more CPU time. You must be doing something wrong. I'm doing 32-bit, 16-channel input at 51.2kHz on a OMAP-L138 (an ARM9 CPU at 450 MHz) under Linux and it's using no more than 25% CPU for that. Just using"arecord" for that. Mike. (sorry for the company sig) Met vriendelijke groet / kind regards, Mike Looijmans TOPIC Embedded Systems Eindhovenseweg 32-C, NL-5683 KH Best Postbus 440, NL-5680 AK Best Telefoon: (+31) – (0)499 - 33.69.79 Telefax: (+31) - (0)499 - 33.69.70 E-mail: mike.looijmans(a)topic.nl Website: www.topic.nl Dit e-mail bericht en de eventueel daarbij behorende bijlagen zijn uitsluitend bestemd voor de geadresseerde, zoals die blijkt uit het e-mail bericht en/of de bijlagen. Er kunnen gegevens met betrekking tot een derde instaan. Indien u als niet-geadresseerde dit bericht en de bijlagen ontvangt, terwijl u niet bevoegd of gemachtigd bent om dit bericht namens de geadresseerde te ontvangen, wordt u verzocht de afzender hierover direct te informeren en het e-mail bericht met de bijlagen te vernietigen. Ieder gebruik van de inhoud van het e-mail bericht, waaronder de daarbij behorende bijlagen, door een ander dan de geadresseerde is onrechtmatig jegens ons dan wel de eventueel in het e-mail bericht of de bijlagen voorkomende andere personen. TOPIC Embedded Systems is niet aansprakelijk voor enigerlei schade voortvloeiend uit het gebruik en/of acceptatie van dit e-mail bericht of de daarbij behorende bijlagen. The contents of this message, as well as any enclosures, are addressed personally to, and thus solely intended for the addressee. They may contain information regarding a third party. A recipient who is neither the addressee, nor empowered to receive this message on behalf of the addressee, is kindly requested to immediately inform the sender of receipt, and to destroy the message and the enclosures. Any use of the contents of this message and/or the enclosures by any other person than the addressee or person who is empowered to receive this message, is illegal towards the sender and/or the aforementioned third party. TOPIC Embedded Systems is not liable for any damage as a result of the use and/or acceptance of this message and as well as any enclosures.

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[alsa-devel] Kernel panic with SB Live (snd_emu10k1) when sharing interrupts
by John Paul Adrian Glaubitz 19 Oct '12

19 Oct '12

Hello, I have a Creative SoundBlaster Live! PCI (PCI-IDs: 1102:0002 and 1102:7002) which I recently ran into trouble with. I used the SB Live! to replace my SB128 PCI which ran fine in the configuration (ASUS P8B-C/4L, see attached output from dmesg and lspci). Once I swapped the sound cards, I booted up the machine and immediately ran into a kernel panic [1] which seems to indicate that there is a problem with interrupt sharing. I there removed the Firewire card which was placed adjacent to the SB Live and the machine ran stable again. I assume that the snd_emu10k1 driver is sometimes not handling shared interrupts properly, isn't it? I wanted to report this as a bug in the ALSA bug tracker [2], but alas the tracker is currently down. I'm attaching the output of dmesg [3] as well as lspci [4]. Will provide more information if requested. Cheers, Adrian > [1] http://users.physik.fu-berlin.de/~glaubitz/emu10k1_panic.jpg > [2] https://bugtrack.alsa-project.org/alsa-bug/ > [3] http://users.physik.fu-berlin.de/~glaubitz/dmesg_emu10k1.txt > [4] http://users.physik.fu-berlin.de/~glaubitz/lspci_emu10k1.txt

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[alsa-devel] [PATCH v2 0/2] ASoC: omap-mcpdm updates for 3.7
by Peter Ujfalusi 19 Oct '12

19 Oct '12

Hello Mark, Tony, Change since v1: - Fixed the second patch to keep the omap_mcpdm_open_stream() - after a coffee The mcpdm driver no longer needs to include any plat/*.h file, clearing the way for the single kernel image. Regards, Peter --- Peter Ujfalusi (1): ASoC: omap-mcpdm: Remove OMAP revision check Tony Lindgren (1): ASoC: Fix wrong include for McPDM sound/soc/omap/omap-mcpdm.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) -- 1.7.12

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[alsa-devel] [PATCH 0/5] snd-ice1712: Add Philips PSC724 Ultimate Edge
by Ondrej Zary 19 Oct '12

19 Oct '12

Hello, this patch series adds full support for Philips PSC724 Ultimate Edge sound card to snd-ice1712 driver. Unlike other snd-ice1712 subdrivers, the codec code is splitted into separate files, reusable by other ice1712 subdrivers. Working: all analog outputs (front, rear, center+lfe, headphone) and inputs (front mic, rear mic, line, cd, aux), mixer, headphone jack detection, suspend/resume Untested: SPDIF -- Ondrej Zary

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[alsa-devel] ESI drivers
by Neil Mcbennett 18 Oct '12

18 Oct '12

Hi, I am not a developer and probably shouldn't even be here, but I do have a question that is probably best answered from this forum. I've been looking at the ESI website and I notice that there is some ALSA support for a few of their cards... http://www.alsa-project.org/main/index.php/Matrix:Vendor-ESI The interface I'm most interested in isn't on this list (MaXiO 032e) I decided to call the technical support number in the UK and ask them what their position is regarding Linux. The guy I spoke to said that ESI are not going to provide an ALSA driver for these devices but they would be more than happy if someone from the ALSA community were to approach them. They would then assist in providing the necessary information. So, just wondering if anyone has already started on this or if there might be some interest in doing so? I was attracted to ESI by the fact that they have a number of PCIe interfaces which are still quite something of a rarity. Regards Neil

1 0

[alsa-devel] LAC 2013: the Linux Audio Conference - Call for Participation
by Robin Gareus 18 Oct '12

18 Oct '12

sorry for >< please >> << We are happy to announce the next issue of the Linux Audio Conference (LAC), May 9-12, 2013 @ IEM, the Institute of Electronic Music and Acoustics, in Graz, Austria. The Linux Audio Conference is an international conference that brings together musicians, sound artists, software developers and researchers, working with Linux as an open, stable, professional platform for audio and media research and music production. LAC includes paper sessions, workshops, and a diverse program of electronic music. *Call for Papers, Workshops, Music and Installations* We invite submissions of papers addressing all areas of audio processing and media creation based on Linux. Papers can focus on technical, artistic and scientific issues and should target developers or users. In our call for music, we are looking for works that have been produced or composed entirely/mostly using Linux. The online submission of papers, workshops, music and installations is now open at http://lac.iem.at/ The Deadline for all submissions is February 4th, 2013 (23:59 HAST) You are invited to register for participation on our conference website. There you will find up-to-date instructions, as well as important information about dates, travel, lodging, and so on. This year's conference is hosted by IEM, Graz, in cooperation with local artists and FLOSS enthusiasts. The Institute of Electronic Music and Acoustics (IEM) at the University of Music and Performing Arts Graz is considered Austria's leading institution in computer music, acoustics and audio engineering and has gained international reputation for its research on spatial audio and its artistic production and research. IEM has been embracing Linux audio as a production and research environment since the mid-1990s, and has contributed to FLOSS/Linux projects, amongst others by providing drivers for multichannel audio interfaces and hosting the Pure Data community portal and mailing lists. http://iem.at/ We look forward to seeing you in Graz in May! Sincerely, The LAC 2013 Organizing Team

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[alsa-devel] [PATCH v2 0/2] Convert davinci ASoC to genalloc SRAM
by Matt Porter 18 Oct '12

18 Oct '12

Changes since v1: - Rebased against uio_pruss/genalloc series v4 patches, requiring another header to be included. This series converts davinci ASoC to use genalloc and enables that support on DA850. It applies to mainline on top of the uio_pruss/genalloc series [1] which allows DaVinci to provide a gen_pool via pdata for driver use. I've tested this on the AM180x EVM. Note that prior to this, the SRAM paths in the driver were completely unused. I've only enabled ping-pong buffering on the platform I can test as it's best to allow those with DM644x and similar platforms to set the playback/capture sram size to something that's known to work for them. [1] https://lkml.org/lkml/2012/10/5/425 Matt Porter (2): ASoC: davinci: replace private sram api with genalloc ARM: davinci: enable SRAM ping ping buffering on DA850 arch/arm/mach-davinci/board-da850-evm.c | 25 +++++++++++++++---------- include/linux/platform_data/davinci_asp.h | 3 +++ sound/soc/davinci/davinci-mcasp.c | 2 ++ sound/soc/davinci/davinci-pcm.c | 18 ++++++++++++------ sound/soc/davinci/davinci-pcm.h | 2 ++ 5 files changed, 34 insertions(+), 16 deletions(-) -- 1.7.9.5

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Re: [alsa-devel] [PATCH] dma: add new DMA control commands
by Huang Shijie 18 Oct '12

18 Oct '12

On Thu, Oct 18, 2012 at 5:29 AM, Jassi Brar <jaswinder.singh(a)linaro.org> wrote: > On 18 October 2012 12:15, Huang Shijie <b32955(a)freescale.com> wrote: >> 于 2012年10月18日 14:18, Vinod Koul 写道: >> >>> Why cant you do start (prepare clock etc) when you submit the descriptor >>> to dmaengine. Can be done in tx_submit callback. >>> Similarly remove the clock when dma transaction gets completed. >> >> I ever thought this method too. >> >> But it will become low efficient in the following case: >> >> Assuming the gpmi-nand driver has to read out 1024 pages in one _SINGLE_ >> read operation. >> The gpmi-nand will submit the descriptor to dmaengine per page. So with your >> method, >> the system will repeat the enable/disable dma clock 1024 time. At every >> enable/disable dma clock, >> the system has to enable the clock chain and it's parents ... >> >> But with this patch, we only need to enable/disable dma clock one time, just >> at we select the nand chip. >> > If the clock is the dmac's property (not channels'), the toggling > seems too aggressive. > You could try using runtime_suspend/resume for clock > disabling/enabling. How about employing autosuspend with a few ms > delay? Yes, employing the autosuspend is workable method too.o But it's a little more complicated then this patch. You have to create a thread or workqueue to do the job. What's more, I think other DMA engine may also meets the same issue as the mxs-dma, such as the imx-sdma. It's somehow a common issue to disable the clocks when no one use the DMA engine. Do you also suggest employing the autosuspend? Why not use a simple way to handle this issue? When the dma device want to use a DMA engine, it just needs to issue a DMA_START command. thanks Huang Shijie Huang Shijie

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Re: [alsa-devel] [PATCH v2 2/2] ARM: davinci: enable SRAM ping ping buffering on DA850
by Matt Porter 18 Oct '12

18 Oct '12

On Fri, Oct 12, 2012 at 07:11:06PM +0200, Sergei Shtylyov wrote: > Hello. > > On 11-10-2012 23:33, Matt Porter wrote: > > >Passes the DA850 shared SRAM gen_pool to the McASP driver > >and enables the ping-pong buffer DMA support. > > Here "ping-pong is correct but the subject have a typo. Thanks. Daniel: can you correct this when you squash these into your series? -Matt

1 0

[alsa-devel] [PATCH] dma: add new DMA control commands
by Huang Shijie 18 Oct '12

18 Oct '12

From: Huang Shijie <shijie8(a)gmail.com> [1] Why add these new DMA control commands? In mx6q, the gpmi-nand driver is the only user of the APBH-DMA. The dma clock is enabled when we have successfully requested a DMA channel. So even when the gpmi-nand driver does not work, the dma clock(apbh-dma) still runs in high speed (198MHz). To save some power, it is better to disable the dma clock when the dma device, such as gpmi-nand, is not working anymore. When the dma device becomes work again, enable the dma clock again. [2] add new DMA control commands: DMA_START/DMA_END DMA_START: do some preprations to start the DMA engine, such as enable the necessary clocks. DMA_END: do some works to end the DMA engine, such as disable the necessary clocks. [3] This patch does not change any logic in i2c-mxs driver and mxs-pcm driver. But for gpmi-nand driver, we will enable the the clock only when we select the nand chip, and want to do some real jobs with the nand chip. For mxs-mmc driver, disable the dma clock in the suspend; and enable the dma clock in the resume. Signed-off-by: Huang Shijie <b32955(a)freescale.com> --- drivers/dma/mxs-dma.c | 17 +++++++++++++++++ drivers/i2c/busses/i2c-mxs.c | 10 +++++++++- drivers/mmc/host/mxs-mmc.c | 19 ++++++++++++++++--- drivers/mtd/nand/gpmi-nand/gpmi-lib.c | 10 ++++++++++ drivers/mtd/nand/gpmi-nand/gpmi-nand.c | 10 ++++++---- include/linux/dmaengine.h | 6 ++++++ sound/soc/mxs/mxs-pcm.c | 12 ++++++++++++ 7 files changed, 76 insertions(+), 8 deletions(-) diff --git a/drivers/dma/mxs-dma.c b/drivers/dma/mxs-dma.c index 9f02e79..89286f4 100644 --- a/drivers/dma/mxs-dma.c +++ b/drivers/dma/mxs-dma.c @@ -384,6 +384,8 @@ static int mxs_dma_alloc_chan_resources(struct dma_chan *chan) /* the descriptor is ready */ async_tx_ack(&mxs_chan->desc); + clk_disable_unprepare(mxs_dma->clk); + return 0; err_clk: @@ -399,6 +401,14 @@ static void mxs_dma_free_chan_resources(struct dma_chan *chan) { struct mxs_dma_chan *mxs_chan = to_mxs_dma_chan(chan); struct mxs_dma_engine *mxs_dma = mxs_chan->mxs_dma; + int ret; + + ret = clk_prepare_enable(mxs_dma->clk); + if (ret) { + dev_err(mxs_dma->dma_device.dev, + "failed in enabling the dma clock\n"); + return; + } mxs_dma_disable_chan(mxs_chan); @@ -597,9 +607,13 @@ static int mxs_dma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, unsigned long arg) { struct mxs_dma_chan *mxs_chan = to_mxs_dma_chan(chan); + struct mxs_dma_engine *mxs_dma = mxs_chan->mxs_dma; int ret = 0; switch (cmd) { + case DMA_START: + ret = clk_prepare_enable(mxs_dma->clk); + break; case DMA_TERMINATE_ALL: mxs_dma_reset_chan(mxs_chan); mxs_dma_disable_chan(mxs_chan); @@ -610,6 +624,9 @@ static int mxs_dma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, case DMA_RESUME: mxs_dma_resume_chan(mxs_chan); break; + case DMA_END: + clk_disable_unprepare(mxs_dma->clk); + break; default: ret = -ENOSYS; } diff --git a/drivers/i2c/busses/i2c-mxs.c b/drivers/i2c/busses/i2c-mxs.c index 1f58197..da1e881 100644 --- a/drivers/i2c/busses/i2c-mxs.c +++ b/drivers/i2c/busses/i2c-mxs.c @@ -643,6 +643,12 @@ static int __devinit mxs_i2c_probe(struct platform_device *pdev) dev_err(dev, "Failed to request dma\n"); return -ENODEV; } + err = dmaengine_device_control(i2c->dmach, DMA_START, 0); + if (err) { + dma_release_channel(i2c->dmach); + dev_err(dev, "Failed to start dma\n"); + return err; + } } platform_set_drvdata(pdev, i2c); @@ -680,8 +686,10 @@ static int __devexit mxs_i2c_remove(struct platform_device *pdev) if (ret) return -EBUSY; - if (i2c->dmach) + if (i2c->dmach) { + dmaengine_device_control(i2c->dmach, DMA_END, 0); dma_release_channel(i2c->dmach); + } writel(MXS_I2C_CTRL0_SFTRST, i2c->regs + MXS_I2C_CTRL0_SET); diff --git a/drivers/mmc/host/mxs-mmc.c b/drivers/mmc/host/mxs-mmc.c index 80d1e6d..aa91830 100644 --- a/drivers/mmc/host/mxs-mmc.c +++ b/drivers/mmc/host/mxs-mmc.c @@ -676,6 +676,9 @@ static int mxs_mmc_probe(struct platform_device *pdev) "%s: failed to request dma\n", __func__); goto out_clk_put; } + ret = dmaengine_device_control(ssp->dmach, DMA_START, 0); + if (ret) + goto out_free_dma; /* set mmc core parameters */ mmc->ops = &mxs_mmc_ops; @@ -717,18 +720,20 @@ static int mxs_mmc_probe(struct platform_device *pdev) ret = devm_request_irq(&pdev->dev, irq_err, mxs_mmc_irq_handler, 0, DRIVER_NAME, host); if (ret) - goto out_free_dma; + goto out_end_free_dma; spin_lock_init(&host->lock); ret = mmc_add_host(mmc); if (ret) - goto out_free_dma; + goto out_end_free_dma; dev_info(mmc_dev(host->mmc), "initialized\n"); return 0; +out_end_free_dma: + dmaengine_device_control(ssp->dmach, DMA_END, 0); out_free_dma: if (ssp->dmach) dma_release_channel(ssp->dmach); @@ -750,8 +755,10 @@ static int mxs_mmc_remove(struct platform_device *pdev) platform_set_drvdata(pdev, NULL); - if (ssp->dmach) + if (ssp->dmach) { + dmaengine_device_control(ssp->dmach, DMA_END, 0); dma_release_channel(ssp->dmach); + } clk_disable_unprepare(ssp->clk); clk_put(ssp->clk); @@ -772,6 +779,7 @@ static int mxs_mmc_suspend(struct device *dev) ret = mmc_suspend_host(mmc); clk_disable_unprepare(ssp->clk); + dmaengine_device_control(ssp->dmach, DMA_END, 0); return ret; } @@ -784,8 +792,13 @@ static int mxs_mmc_resume(struct device *dev) int ret = 0; clk_prepare_enable(ssp->clk); + ret = dmaengine_device_control(ssp->dmach, DMA_START, 0); + if (ret) + return ret; ret = mmc_resume_host(mmc); + if (ret) + dmaengine_device_control(ssp->dmach, DMA_END, 0); return ret; } diff --git a/drivers/mtd/nand/gpmi-nand/gpmi-lib.c b/drivers/mtd/nand/gpmi-nand/gpmi-lib.c index 3502acc..20ed3f3 100644 --- a/drivers/mtd/nand/gpmi-nand/gpmi-lib.c +++ b/drivers/mtd/nand/gpmi-nand/gpmi-lib.c @@ -958,6 +958,7 @@ void gpmi_begin(struct gpmi_nand_data *this) uint32_t reg; unsigned int dll_wait_time_in_us; struct gpmi_nfc_hardware_timing hw; + struct dma_chan *channel = get_dma_chan(this); int ret; /* Enable the clock. */ @@ -967,6 +968,12 @@ void gpmi_begin(struct gpmi_nand_data *this) goto err_out; } + ret = dmaengine_device_control(channel, DMA_START, 0); + if (ret) { + gpmi_disable_clk(this); + goto err_out; + } + /* Only initialize the timing once */ if (this->flags & GPMI_TIMING_INIT_OK) return; @@ -1035,7 +1042,10 @@ err_out: void gpmi_end(struct gpmi_nand_data *this) { + struct dma_chan *channel = get_dma_chan(this); + gpmi_disable_clk(this); + dmaengine_device_control(channel, DMA_END, 0); } /* Clears a BCH interrupt. */ diff --git a/drivers/mtd/nand/gpmi-nand/gpmi-nand.c b/drivers/mtd/nand/gpmi-nand/gpmi-nand.c index e2c56fc..5694d03 100644 --- a/drivers/mtd/nand/gpmi-nand/gpmi-nand.c +++ b/drivers/mtd/nand/gpmi-nand/gpmi-nand.c @@ -815,12 +815,14 @@ static void gpmi_select_chip(struct mtd_info *mtd, int chipnr) struct nand_chip *chip = mtd->priv; struct gpmi_nand_data *this = chip->priv; - if ((this->current_chip < 0) && (chipnr >= 0)) + if ((this->current_chip < 0) && (chipnr >= 0)) { + /* set the current_chip before we call gpmi_begin(). */ + this->current_chip = chipnr; gpmi_begin(this); - else if ((this->current_chip >= 0) && (chipnr < 0)) + } else if ((this->current_chip >= 0) && (chipnr < 0)) { gpmi_end(this); - - this->current_chip = chipnr; + this->current_chip = chipnr; + } } static void gpmi_read_buf(struct mtd_info *mtd, uint8_t *buf, int len) diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index d3201e4..79f864a 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -199,6 +199,8 @@ enum dma_ctrl_flags { /** * enum dma_ctrl_cmd - DMA operations that can optionally be exercised * on a running channel. + * @DMA_START: do some preprations to start the DMA engine, such as enable the + * necessary clocks. * @DMA_TERMINATE_ALL: terminate all ongoing transfers * @DMA_PAUSE: pause ongoing transfers * @DMA_RESUME: resume paused transfer @@ -209,13 +211,17 @@ enum dma_ctrl_flags { * command. * @FSLDMA_EXTERNAL_START: this command will put the Freescale DMA controller * into external start mode. + * @DMA_END: do some works to end the DMA engine, such as disable the + * necessary clocks. */ enum dma_ctrl_cmd { + DMA_START, DMA_TERMINATE_ALL, DMA_PAUSE, DMA_RESUME, DMA_SLAVE_CONFIG, FSLDMA_EXTERNAL_START, + DMA_END, }; /** diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index f82d766..cfcc30f 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -92,6 +92,7 @@ static int snd_mxs_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mxs_pcm_dma_data *pcm_dma_data; + struct dma_chan *chan; int ret; pcm_dma_data = kzalloc(sizeof(*pcm_dma_data), GFP_KERNEL); @@ -107,6 +108,14 @@ static int snd_mxs_open(struct snd_pcm_substream *substream) return ret; } + chan = snd_dmaengine_pcm_get_chan(substream); + ret = dmaengine_device_control(chan, DMA_START, (unsigned long)0); + if (ret) { + snd_dmaengine_pcm_close(substream); + kfree(pcm_dma_data); + return ret; + } + snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware); snd_dmaengine_pcm_set_data(substream, pcm_dma_data); @@ -117,7 +126,10 @@ static int snd_mxs_open(struct snd_pcm_substream *substream) static int snd_mxs_close(struct snd_pcm_substream *substream) { struct mxs_pcm_dma_data *pcm_dma_data = snd_dmaengine_pcm_get_data(substream); + struct dma_chan *chan; + chan = snd_dmaengine_pcm_get_chan(substream); + dmaengine_device_control(chan, DMA_END, 0); snd_dmaengine_pcm_close(substream); kfree(pcm_dma_data); -- 1.7.0.4

5 13

[alsa-devel] Non-Interleaved access method?
by Prasant J 18 Oct '12

18 Oct '12

Hi, When the application set the access method to non-interleaved, that means that the ALSA application will deliver the non-interleaved buffers to alsa subsystem & the alsa subsystem will interleave the data before sending out the data to the hardware. Am I correct? Or non-interleaved means that the data will be sent out in non-interleaved manner itself? Regards, -Prasant

2 1

[alsa-devel] ASoC: mmap or read/write?
by Prasant J 18 Oct '12

18 Oct '12

Hi, I'm writing an alsa application for recording & playing 16 channel audio. The application is intended to run on an embedded platform & the resources are highly constrained. My application is not performing well with read/write methods. I'm considering using mmap but not able to get the right documentation or start, so I want to know whether mmap really offers any advantages or not. Does the mmap method has any advantages over read/write? Is there any thumb rule to decide when to use mmap or read/write? -Prasant

2 3

[alsa-devel] Interleaved or Non-interleaved?
by Prasant J 18 Oct '12

18 Oct '12

hi, My sound hardware supports only interleaved audio. I'm writing an alsa application. Which access method will be more performing : interleaved or non-interleaved? (Or it does not make a difference?) -Prasant

2 2

[alsa-devel] [PATCH 1/2] ARM: OMAP: Trivial driver changes to remove include plat/cpu.h
by Tony Lindgren 17 Oct '12

17 Oct '12

Drivers should not use cpu_is_omap or cpu_class_is_omap macros, they should be private to the platform init code. And we'll be removing plat/cpu.h and only have a private soc.h for the arch/arm/*omap* code. This patch is intended as preparation for the core omap changes and removes the need to include plat/cpu.h from several drivers. This is needed for the ARM common zImage support. These changes are OK to do because: - omap-rng.c and hsmmc.c don't need plat/cpu.h - omap-aes.c and omap-sham.c get the proper platform_data passed to them so they don't need extra checks in the driver - omap-dma.c and omap-pcm.c can test the arch locally as omap1 and omap2 cannot be compiled together because of conflicting compiler flags Cc: Deepak Saxena <dsaxena(a)plexity.net> Cc: Matt Mackall <mpm(a)selenic.com> Cc: Herbert Xu <herbert(a)gondor.apana.org.au> Cc: David S. Miller <davem(a)davemloft.net> Cc: Venkatraman S <svenkatr(a)ti.com> Cc: Chris Ball <cjb(a)laptop.org> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Dan Williams <djbw(a)fb.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: linux-crypto(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- drivers/char/hw_random/omap-rng.c | 2 -- drivers/crypto/omap-aes.c | 6 ------ drivers/crypto/omap-sham.c | 8 -------- drivers/dma/omap-dma.c | 9 +++++++-- drivers/mmc/host/omap_hsmmc.c | 1 - sound/soc/omap/omap-pcm.c | 9 +++++++-- 6 files changed, 14 insertions(+), 21 deletions(-) diff --git a/drivers/char/hw_random/omap-rng.c b/drivers/char/hw_random/omap-rng.c index a5effd8..45e467d 100644 --- a/drivers/char/hw_random/omap-rng.c +++ b/drivers/char/hw_random/omap-rng.c @@ -27,8 +27,6 @@ #include <asm/io.h> -#include <plat/cpu.h> - #define RNG_OUT_REG 0x00 /* Output register */ #define RNG_STAT_REG 0x04 /* Status register [0] = STAT_BUSY */ diff --git a/drivers/crypto/omap-aes.c b/drivers/crypto/omap-aes.c index 3e61feb..649a146 100644 --- a/drivers/crypto/omap-aes.c +++ b/drivers/crypto/omap-aes.c @@ -29,7 +29,6 @@ #include <crypto/scatterwalk.h> #include <crypto/aes.h> -#include <plat/cpu.h> #include <plat-omap/dma-omap.h> /* OMAP TRM gives bitfields as start:end, where start is the higher bit @@ -941,11 +940,6 @@ static int __init omap_aes_mod_init(void) { pr_info("loading %s driver\n", "omap-aes"); - if (!cpu_class_is_omap2() || omap_type() != OMAP2_DEVICE_TYPE_SEC) { - pr_err("Unsupported cpu\n"); - return -ENODEV; - } - return platform_driver_register(&omap_aes_driver); } diff --git a/drivers/crypto/omap-sham.c b/drivers/crypto/omap-sham.c index 6d79695..d76fe06 100644 --- a/drivers/crypto/omap-sham.c +++ b/drivers/crypto/omap-sham.c @@ -37,7 +37,6 @@ #include <crypto/hash.h> #include <crypto/internal/hash.h> -#include <plat/cpu.h> #include <plat-omap/dma-omap.h> #include <mach/irqs.h> @@ -1289,13 +1288,6 @@ static int __init omap_sham_mod_init(void) { pr_info("loading %s driver\n", "omap-sham"); - if (!cpu_class_is_omap2() || - (omap_type() != OMAP2_DEVICE_TYPE_SEC && - omap_type() != OMAP2_DEVICE_TYPE_EMU)) { - pr_err("Unsupported cpu\n"); - return -ENODEV; - } - return platform_driver_register(&omap_sham_driver); } diff --git a/drivers/dma/omap-dma.c b/drivers/dma/omap-dma.c index 0b4fa50..56d9253 100644 --- a/drivers/dma/omap-dma.c +++ b/drivers/dma/omap-dma.c @@ -19,9 +19,14 @@ #include "virt-dma.h" -#include <plat/cpu.h> #include <plat-omap/dma-omap.h> +#ifdef CONFIG_ARCH_OMAP2PLUS +#define dma_omap2plus() 1 +#else +#define dma_omap2plus() 0 +#endif + struct omap_dmadev { struct dma_device ddev; spinlock_t lock; @@ -438,7 +443,7 @@ static struct dma_async_tx_descriptor *omap_dma_prep_dma_cyclic( omap_disable_dma_irq(c->dma_ch, OMAP_DMA_BLOCK_IRQ); } - if (!cpu_class_is_omap1()) { + if (dma_omap2plus()) { omap_set_dma_src_burst_mode(c->dma_ch, OMAP_DMA_DATA_BURST_16); omap_set_dma_dest_burst_mode(c->dma_ch, OMAP_DMA_DATA_BURST_16); } diff --git a/drivers/mmc/host/omap_hsmmc.c b/drivers/mmc/host/omap_hsmmc.c index 85b164e..24df035 100644 --- a/drivers/mmc/host/omap_hsmmc.c +++ b/drivers/mmc/host/omap_hsmmc.c @@ -40,7 +40,6 @@ #include <linux/pm_runtime.h> #include <linux/platform_data/mmc-omap.h> #include <mach/hardware.h> -#include <plat/cpu.h> /* OMAP HSMMC Host Controller Registers */ #define OMAP_HSMMC_SYSSTATUS 0x0014 diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 340874e..52977aa 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -32,9 +32,14 @@ #include <sound/dmaengine_pcm.h> #include <sound/soc.h> -#include <plat/cpu.h> #include "omap-pcm.h" +#ifdef CONFIG_ARCH_OMAP1 +#define pcm_omap1510() cpu_is_omap1510() +#else +#define pcm_omap1510() 0 +#endif + static const struct snd_pcm_hardware omap_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -159,7 +164,7 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) { snd_pcm_uframes_t offset; - if (cpu_is_omap1510()) + if (pcm_omap1510()) offset = snd_dmaengine_pcm_pointer_no_residue(substream); else offset = snd_dmaengine_pcm_pointer(substream);

3 6

[alsa-devel] [PATCH] emu10k1: add chip details for E-mu 1010 PCIe card
by Maxim Kachur 17 Oct '12

17 Oct '12

emu10k1_main.c: Add chip details for E-mu 1010 PCIe card. It has the same chip as found in E-mu 1010b but it uses different PCI id. Signed-off-by: Maxim Kachur <mcdebugger(a)duganet.ru> --- sound/pci/emu10k1/emu10k1_main.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index bed4485..c21adb6 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1416,6 +1416,15 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0108_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 new revision */ + /* Tested by Maxim Kachur <mcdebugger(a)duganet.ru> 17th Oct 2012. */ + /* This is MAEM8986, 0202 is MAEM8980 */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40071102, + .driver = "Audigy2", .name = "E-mu 1010 PCIe [MAEM8986]", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 PCIe */ /* Tested by James(a)superbug.co.uk 8th July 2005. */ /* This is MAEM8810, 0202 is MAEM8820 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, -- 1.7.8.6

2 2

[alsa-devel] [PATCH] emu10k1: add chip details for E-mu 1010 PCIe card
by Maxim Kachur 17 Oct '12

17 Oct '12

emu10k1_main.c: Add chip details for E-mu 1010 PCIe card. It has the same chip as found in E-mu 1010b but it uses different PCI id. Signed-off-by: Maxim Kachur <mcdebugger(a)duganet.ru> --- sound/pci/emu10k1/emu10k1_main.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index bed4485..c21adb6 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1416,6 +1416,15 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0108_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 new revision */ + /* Tested by Maxim Kachur <mcdebugger(a)duganet.ru> 17th Oct 2012. */ + /* This is MAEM8986, 0202 is MAEM8980 */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40071102, + .driver = "Audigy2", .name = "E-mu 1010 PCIe [MAEM8986]", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 PCIe */ /* Tested by James(a)superbug.co.uk 8th July 2005. */ /* This is MAEM8810, 0202 is MAEM8820 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, -- 1.7.8.6

1 0

[alsa-devel] [PATCH] ASoC: codecs: da9055: DAPMise Headphone and Lineout enable controls
by Ashish Chavan 17 Oct '12

17 Oct '12

This patch DAPMises headphone and lineout output enable controls. Earlier these output enable bits were permanently turned on in probe. In da9055 codec, right outmixer is directly connected with right HP and Line out. This resulted in two side effects, (1) When you only want to use lineout, right HP (and connected charge pump) also gets enabled (2) When you only want to use stereo HP, lineout also gets enabled This patch adds three switches to select which output(s) should be enabled. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/da9055.c | 39 ++++++++++++++++++++++++++------------- 1 files changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index abf3e43..d3a6de2 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -753,6 +753,17 @@ static const struct snd_kcontrol_new da9055_dapm_mixoutr_controls[] = { 6, 1, 0), }; +/* Headphone Output Enable */ +static const struct snd_kcontrol_new da9055_dapm_hp_l_control = +SOC_DAPM_SINGLE("Switch", DA9055_HP_L_CTRL, 3, 1, 0); + +static const struct snd_kcontrol_new da9055_dapm_hp_r_control = +SOC_DAPM_SINGLE("Switch", DA9055_HP_R_CTRL, 3, 1, 0); + +/* Lineout Output Enable */ +static const struct snd_kcontrol_new da9055_dapm_lineout_control = +SOC_DAPM_SINGLE("Switch", DA9055_LINE_CTRL, 3, 1, 0); + /* DAPM widgets */ static const struct snd_soc_dapm_widget da9055_dapm_widgets[] = { /* Input Side */ @@ -817,6 +828,14 @@ static const struct snd_soc_dapm_widget da9055_dapm_widgets[] = { &da9055_dapm_mixoutr_controls[0], ARRAY_SIZE(da9055_dapm_mixoutr_controls)), + /* Output Enable Switches */ + SND_SOC_DAPM_SWITCH("Headphone Left Enable", SND_SOC_NOPM, 0, 0, + &da9055_dapm_hp_l_control), + SND_SOC_DAPM_SWITCH("Headphone Right Enable", SND_SOC_NOPM, 0, 0, + &da9055_dapm_hp_r_control), + SND_SOC_DAPM_SWITCH("Lineout Enable", SND_SOC_NOPM, 0, 0, + &da9055_dapm_lineout_control), + /* Output PGAs */ SND_SOC_DAPM_PGA("MIXOUT Left", DA9055_MIXOUT_L_CTRL, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("MIXOUT Right", DA9055_MIXOUT_R_CTRL, 7, 0, NULL, 0), @@ -902,17 +921,20 @@ static const struct snd_soc_dapm_route da9055_audio_map[] = { {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, {"MIXOUT Left", NULL, "Out Mixer Left"}, - {"Headphone Left", NULL, "MIXOUT Left"}, + {"Headphone Left Enable", "Switch", "MIXOUT Left"}, + {"Headphone Left", NULL, "Headphone Left Enable"}, {"Headphone Left", NULL, "Charge Pump"}, {"HPL", NULL, "Headphone Left"}, {"MIXOUT Right", NULL, "Out Mixer Right"}, - {"Headphone Right", NULL, "MIXOUT Right"}, + {"Headphone Right Enable", "Switch", "MIXOUT Right"}, + {"Headphone Right", NULL, "Headphone Right Enable"}, {"Headphone Right", NULL, "Charge Pump"}, {"HPR", NULL, "Headphone Right"}, {"MIXOUT Right", NULL, "Out Mixer Right"}, - {"Lineout", NULL, "MIXOUT Right"}, + {"Lineout Enable", "Switch", "MIXOUT Right"}, + {"Lineout", NULL, "Lineout Enable"}, {"LINE", NULL, "Lineout"}, }; @@ -1394,8 +1416,7 @@ static int da9055_probe(struct snd_soc_codec *codec) DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); /* - * There are two separate control bits for input and output mixers as - * well as headphone and line outs. + * There are two separate control bits for input and output mixers. * One to enable corresponding amplifier and other to enable its * output. As amplifier bits are related to power control, they are * being managed by DAPM while other (non power related) bits are @@ -1411,14 +1432,6 @@ static int da9055_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, DA9055_MIXOUT_R_CTRL, DA9055_MIXOUT_R_MIX_EN, DA9055_MIXOUT_R_MIX_EN); - snd_soc_update_bits(codec, DA9055_HP_L_CTRL, - DA9055_HP_L_AMP_OE, DA9055_HP_L_AMP_OE); - snd_soc_update_bits(codec, DA9055_HP_R_CTRL, - DA9055_HP_R_AMP_OE, DA9055_HP_R_AMP_OE); - - snd_soc_update_bits(codec, DA9055_LINE_CTRL, - DA9055_LINE_AMP_OE, DA9055_LINE_AMP_OE); - /* Set this as per your system configuration */ snd_soc_write(codec, DA9055_PLL_CTRL, DA9055_PLL_INDIV_10_20_MHZ); -- 1.7.1

2 1

[alsa-devel] [PATCH 0/3] Convert to use snd_soc_register_card()
by Bo Shen 17 Oct '12

17 Oct '12

Convert to use snd_soc_register_card(), and enable it Change the id for the gpio i2c to let it work Bo Shen (3): ASoC: sam9g20-wm8731: convert to use snd_soc_register_card() ASoC: sam9g20: using platform device for audio part i2c: change the id to let the i2c-gpio work arch/arm/mach-at91/at91sam9260_devices.c | 11 ++++++- arch/arm/mach-at91/board-sam9g20ek.c | 17 ++++++++++ sound/soc/atmel/sam9g20_wm8731.c | 51 +++++++++++++----------------- 3 files changed, 49 insertions(+), 30 deletions(-) -- 1.7.9.5

2 7

[alsa-devel] [PATCH] ALSA: hda - Don't enable unsol for jacks we're polling
by David Henningsson 17 Oct '12

17 Oct '12

This will greatly reduce the frequency for detection errors in those cases where the hardware is "flaky", i e, rapidly changing between plugged and unplugged states even without user interaction. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 07d5288..5bdbada 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -206,6 +206,8 @@ int snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, jack->action = action; if (cb) jack->callback = cb; + if (codec->jackpoll_interval > 0) + return 0; /* No unsol if we're polling instead */ return snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | jack->tag); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] hda-emu: Handle snd_BUG()
by David Henningsson 17 Oct '12

17 Oct '12

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/sound/core.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/core.h b/include/sound/core.h index 57774ea..a749460 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -24,6 +24,7 @@ static inline int _snd_BUG_ON(int cond, const char *func, int line) return cond; } #define snd_BUG_ON(cond) _snd_BUG_ON(cond, __func__, __LINE__) +#define snd_BUG() snd_BUG_ON(1) #include "asound.h" #include "codec_config.h" -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Always check array bounds in alc_get_line_out_pfx
by David Henningsson 17 Oct '12

17 Oct '12

Even when CONFIG_SND_DEBUG is not enabled, we don't want to return an arbitrary memory location when the channel count is larger than we expected. Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Notes: This was found while debugging an OOPS on kernel 3.2. See https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1067269 for details. The badness calculation was improved in kernel 3.4; so not sure if the same could happen in later kernels, but at least apply to kernels <= 3.3 would make sense. Also adding to current trees (3.7, 3.8) would make sense IMO, just as extra safety. sound/pci/hda/patch_realtek.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2d2bb66..651822c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2599,8 +2599,10 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch, return "PCM"; break; } - if (snd_BUG_ON(ch >= ARRAY_SIZE(channel_name))) + if (ch >= ARRAY_SIZE(channel_name)) { + snd_BUG(); return "PCM"; + } return channel_name[ch]; } -- 1.7.9.5

2 1

[alsa-devel] ASoC updates for 3.7
by Mark Brown 17 Oct '12

17 Oct '12

The following changes since commit 9911f7f7562a25381eff93fdc660a4a3b4c0f6e0: ASoC: codecs: Add DA9055 codec driver (2012-09-28 14:21:54 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.7 for you to fetch changes up to 9233ef6b559faa3db14144f41dea2e70d1ad0092: ASoC: bells: Correct typo in sub speaker DAI name for WM5110 (2012-10-17 19:01:38 +0900) ---------------------------------------------------------------- ASoC: Fixes for v3.7 Nothing too exciting except for the ams-delta change which is relatively lerge due to the fact that the driver loading had been totally broken as the driver needed a newer API to function. ---------------------------------------------------------------- Ashish Chavan (1): ASoC: codecs: da9055: Minor improvement in ALC calibration process Chris Ball (1): ARM: pxa: Fix build error caused by sram.h rename Guennadi Liakhovetski (1): ASoC: fsi: don't reschedule DMA from an atomic context Janusz Krzysztofik (1): ASoC: ams-delta: Convert to use snd_soc_register_card() Mark Brown (5): Merge branches 'asoc-twl6040' and 'asoc-omap' into for-3.7 ASoC: bells: Correct typo in sub speaker DAI name for WM5110 ASoC: wm2200: Use rev A register patches on rev B ASoC: wm2200: Fix non-inverted OUT2 mute control ASoC: bells: Correct typo in sub speaker DAI name for WM5110 Peter Meerwald (1): ASoC: fix documentation in soc-jack Peter Ujfalusi (4): ASoC: twl6040: Fix Stream DAPM mapping ASoC: omap-abe-twl6040: Fix typo of Vibrator ASoC: omap-mcpdm: Remove OMAP revision check ASoC: dmaengine: Correct Makefile when sound is built as module Tony Lindgren (1): ASoC: Fix wrong include for McPDM arch/arm/mach-omap1/board-ams-delta.c | 12 +++++++ sound/soc/Makefile | 5 +-- sound/soc/codecs/da9055.c | 22 +++++++++++- sound/soc/codecs/twl6040.c | 8 ++--- sound/soc/codecs/wm2200.c | 3 +- sound/soc/omap/ams-delta.c | 63 ++++++++++++++++----------------- sound/soc/omap/omap-abe-twl6040.c | 2 +- sound/soc/omap/omap-mcpdm.c | 9 ++--- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/samsung/bells.c | 4 +-- sound/soc/sh/fsi.c | 15 ++++---- sound/soc/soc-jack.c | 7 ++-- 12 files changed, 90 insertions(+), 62 deletions(-)

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[alsa-devel] ALSA hda_intel.c:834 spurious response ... ?
by Oleksij Rempel 17 Oct '12

17 Oct '12

Hallo all, i get with latest linus master fallowing messages after resume from suspend to ram: [ 9188.636589] ALSA hda_intel.c:834 spurious response 0x0:0x0, last cmd=0x170500 [ 9188.636592] ALSA hda_intel.c:834 spurious response 0x0:0x0, last cmd=0x170500 [ 9188.636595] ALSA hda_intel.c:834 spurious response 0x0:0x3, last cmd=0x301f0500 [ 9188.636598] ALSA hda_intel.c:834 spurious response 0x0:0x3, last cmd=0x301f0500 [ 9188.636601] ALSA hda_intel.c:834 spurious response 0x0:0x3, last cmd=0x301f0500 [ 9188.636604] ALSA hda_intel.c:834 spurious response 0x9000094:0x3, last cmd=0x301f0500 [ 9188.636606] ALSA hda_intel.c:834 spurious response 0x0:0x3, last cmd=0x301f0500 is it some thing you need to know? For more info see attached dmesg. -- Regards, Oleksij

2 3

[alsa-devel] [PATCH] hda-analyzer: Ignore unknown lines in proc file
by David Henningsson 17 Oct '12

17 Oct '12

With newer kernels, the proc file sometimes add more information. This makes hda-analyzer ignore such new additions by default, instead of the current behaviour (which is to crash with a stack trace). This way you can run an older hda-analyzer with a newer kernel. A recent example is the addition of "IEC Coding type" to all HDMI codecs. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-analyzer/hda_proc.py | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/hda-analyzer/hda_proc.py b/hda-analyzer/hda_proc.py index a6dd3dc..7424365 100644 --- a/hda-analyzer/hda_proc.py +++ b/hda-analyzer/hda_proc.py @@ -721,7 +721,7 @@ class HDACodecProc(HDACodec, HDABaseProc): elif line.startswith(' Volume-Knob: '): node.add_volknob(line[15:]) else: - self.wrongfile(line) + print 'Unknown line, ignored: (%s)' % line idx += 1 if idx < len(lines): line = lines[idx] -- 1.7.9.5

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Re: [alsa-devel] Silence playback during XRUN
by Babu, Ramesh 16 Oct '12

16 Oct '12

Hi I am trying to configure my application to use silence playback of ALSA in case XRUN. I am configuring the silence parameters as below: silence_threshold = 0 silence_size = boundary stop_threshold = bounday Using the above configuration I can see that ALSA sends silence during XRUN. But I am experiencing audio is jerky, when application starts sending the data after the XRUN. I noticed someone has met similar situation. http://mailman.alsa-project.org/pipermail/alsa-devel/2011-December/046894.h… Do I need to modify my silence settings? Anything that I need take care apart from the above mentioned parameters? Thanks Ramesh

1 1

[alsa-devel] [PATCH 1/2] ASoC: atmel-ssc: use devm_xxx() managed function
by Bo Shen 16 Oct '12

16 Oct '12

Using the devm_xxx() managed function to stripdown the error and remove code. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- drivers/misc/atmel-ssc.c | 47 +++++++++++++++------------------------------- 1 file changed, 15 insertions(+), 32 deletions(-) diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index 5bb1877..23dcb76 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -70,37 +70,33 @@ EXPORT_SYMBOL(ssc_free); static int __init ssc_probe(struct platform_device *pdev) { - int retval = 0; struct resource *regs; struct ssc_device *ssc; - ssc = kzalloc(sizeof(struct ssc_device), GFP_KERNEL); + ssc = devm_kzalloc(&pdev->dev, sizeof(struct ssc_device), GFP_KERNEL); if (!ssc) { dev_dbg(&pdev->dev, "out of memory\n"); - retval = -ENOMEM; - goto out; + return -ENOMEM; } + ssc->pdev = pdev; + regs = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!regs) { dev_dbg(&pdev->dev, "no mmio resource defined\n"); - retval = -ENXIO; - goto out_free; + return -ENXIO; } - ssc->clk = clk_get(&pdev->dev, "pclk"); - if (IS_ERR(ssc->clk)) { - dev_dbg(&pdev->dev, "no pclk clock defined\n"); - retval = -ENXIO; - goto out_free; - } - - ssc->pdev = pdev; - ssc->regs = ioremap(regs->start, resource_size(regs)); + ssc->regs = devm_request_and_ioremap(&pdev->dev, regs); if (!ssc->regs) { dev_dbg(&pdev->dev, "ioremap failed\n"); - retval = -EINVAL; - goto out_clk; + return -EINVAL; + } + + ssc->clk = devm_clk_get(&pdev->dev, "pclk"); + if (IS_ERR(ssc->clk)) { + dev_dbg(&pdev->dev, "no pclk clock defined\n"); + return -ENXIO; } /* disable all interrupts */ @@ -112,8 +108,7 @@ static int __init ssc_probe(struct platform_device *pdev) ssc->irq = platform_get_irq(pdev, 0); if (!ssc->irq) { dev_dbg(&pdev->dev, "could not get irq\n"); - retval = -ENXIO; - goto out_unmap; + return -ENXIO; } spin_lock(&user_lock); @@ -125,16 +120,7 @@ static int __init ssc_probe(struct platform_device *pdev) dev_info(&pdev->dev, "Atmel SSC device at 0x%p (irq %d)\n", ssc->regs, ssc->irq); - goto out; - -out_unmap: - iounmap(ssc->regs); -out_clk: - clk_put(ssc->clk); -out_free: - kfree(ssc); -out: - return retval; + return 0; } static int __devexit ssc_remove(struct platform_device *pdev) @@ -142,10 +128,7 @@ static int __devexit ssc_remove(struct platform_device *pdev) struct ssc_device *ssc = platform_get_drvdata(pdev); spin_lock(&user_lock); - iounmap(ssc->regs); - clk_put(ssc->clk); list_del(&ssc->list); - kfree(ssc); spin_unlock(&user_lock); return 0; -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: Ux500: Dispose of device nodes correctly
by Lee Jones 16 Oct '12

16 Oct '12

When of_parse_phandle() is used to find a device node, its reference count is incremented by the helper. Once we're finished with them, it's our responsibly to ensure they are freed in the correct manor. Cc: alsa-devel(a)alsa-project.org Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/mop500.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 6f67e6b..651a52a 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -57,6 +57,20 @@ static struct snd_soc_card mop500_card = { .num_links = ARRAY_SIZE(mop500_dai_links), }; +static void mop500_of_node_put(void) +{ + int i; + + for (i = 0; i < 2; i++) { + if (mop500_dai_links[i].cpu_of_node) + of_node_put((struct device_node *) + mop500_dai_links[i].cpu_of_node); + if (mop500_dai_links[i].codec_of_node) + of_node_put((struct device_node *) + mop500_dai_links[i].codec_of_node); + } +} + static int __devinit mop500_of_probe(struct platform_device *pdev, struct device_node *np) { @@ -69,6 +83,7 @@ static int __devinit mop500_of_probe(struct platform_device *pdev, if (!(msp_np[0] && msp_np[1] && codec_np)) { dev_err(&pdev->dev, "Phandle missing or invalid\n"); + mop500_of_node_put(); return -EINVAL; } @@ -83,6 +98,7 @@ static int __devinit mop500_of_probe(struct platform_device *pdev, return 0; } + static int __devinit mop500_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; @@ -128,6 +144,7 @@ static int __devexit mop500_remove(struct platform_device *pdev) snd_soc_unregister_card(mop500_card); mop500_ab8500_remove(mop500_card); + mop500_of_node_put(); return 0; } -- 1.7.9.5

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[alsa-devel] [PATCH] ARM: tegra: remove <mach/dma.h>
by Stephen Warren 15 Oct '12

15 Oct '12

From: Stephen Warren <swarren(a)nvidia.com> Remove includes of <mach/dma.h> from sound/soc; nothing from it is used. Remove include of <mach/dma.h> from mach-tegra/apbio.c; since the DMA transfers made by this file don't need flow-control with any peripheral, there's no need to set any slave ID. Once those changes are made, there are no remaining users of <mach/dma.h> so remove it. Drivers should get this information from device tree. This removal is necessary for single zImage. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, I'd like to take this through the Tegra tree please, unless you want me to split out the ASoC parts of the patch, and have you apply them to 3.7. arch/arm/mach-tegra/apbio.c | 3 -- arch/arm/mach-tegra/include/mach/dma.h | 54 -------------------------------- sound/soc/tegra/tegra30_ahub.c | 1 - sound/soc/tegra/tegra_pcm.h | 2 - 4 files changed, 0 insertions(+), 60 deletions(-) delete mode 100644 arch/arm/mach-tegra/include/mach/dma.h diff --git a/arch/arm/mach-tegra/apbio.c b/arch/arm/mach-tegra/apbio.c index b5015d0..bc949fb 100644 --- a/arch/arm/mach-tegra/apbio.c +++ b/arch/arm/mach-tegra/apbio.c @@ -24,8 +24,6 @@ #include <linux/sched.h> #include <linux/mutex.h> -#include <mach/dma.h> - #include "apbio.h" #if defined(CONFIG_TEGRA20_APB_DMA) @@ -71,7 +69,6 @@ bool tegra_apb_dma_init(void) dma_sconfig.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; dma_sconfig.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - dma_sconfig.slave_id = TEGRA_DMA_REQ_SEL_CNTR; dma_sconfig.src_maxburst = 1; dma_sconfig.dst_maxburst = 1; diff --git a/arch/arm/mach-tegra/include/mach/dma.h b/arch/arm/mach-tegra/include/mach/dma.h deleted file mode 100644 index 3081cc6..0000000 --- a/arch/arm/mach-tegra/include/mach/dma.h +++ /dev/null @@ -1,54 +0,0 @@ -/* - * arch/arm/mach-tegra/include/mach/dma.h - * - * Copyright (c) 2008-2009, NVIDIA Corporation. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, but WITHOUT - * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or - * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for - * more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. - */ - -#ifndef __MACH_TEGRA_DMA_H -#define __MACH_TEGRA_DMA_H - -#include <linux/list.h> - -#define TEGRA_DMA_REQ_SEL_CNTR 0 -#define TEGRA_DMA_REQ_SEL_I2S_2 1 -#define TEGRA_DMA_REQ_SEL_I2S_1 2 -#define TEGRA_DMA_REQ_SEL_SPD_I 3 -#define TEGRA_DMA_REQ_SEL_UI_I 4 -#define TEGRA_DMA_REQ_SEL_MIPI 5 -#define TEGRA_DMA_REQ_SEL_I2S2_2 6 -#define TEGRA_DMA_REQ_SEL_I2S2_1 7 -#define TEGRA_DMA_REQ_SEL_UARTA 8 -#define TEGRA_DMA_REQ_SEL_UARTB 9 -#define TEGRA_DMA_REQ_SEL_UARTC 10 -#define TEGRA_DMA_REQ_SEL_SPI 11 -#define TEGRA_DMA_REQ_SEL_AC97 12 -#define TEGRA_DMA_REQ_SEL_ACMODEM 13 -#define TEGRA_DMA_REQ_SEL_SL4B 14 -#define TEGRA_DMA_REQ_SEL_SL2B1 15 -#define TEGRA_DMA_REQ_SEL_SL2B2 16 -#define TEGRA_DMA_REQ_SEL_SL2B3 17 -#define TEGRA_DMA_REQ_SEL_SL2B4 18 -#define TEGRA_DMA_REQ_SEL_UARTD 19 -#define TEGRA_DMA_REQ_SEL_UARTE 20 -#define TEGRA_DMA_REQ_SEL_I2C 21 -#define TEGRA_DMA_REQ_SEL_I2C2 22 -#define TEGRA_DMA_REQ_SEL_I2C3 23 -#define TEGRA_DMA_REQ_SEL_DVC_I2C 24 -#define TEGRA_DMA_REQ_SEL_OWR 25 -#define TEGRA_DMA_REQ_SEL_INVALID 31 - -#endif diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index bf56101..64b67a3 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -26,7 +26,6 @@ #include <linux/regmap.h> #include <linux/slab.h> #include <mach/clk.h> -#include <mach/dma.h> #include <sound/soc.h> #include "tegra30_ahub.h" diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h index b40279b..bc8b46a 100644 --- a/sound/soc/tegra/tegra_pcm.h +++ b/sound/soc/tegra/tegra_pcm.h @@ -31,8 +31,6 @@ #ifndef __TEGRA_PCM_H__ #define __TEGRA_PCM_H__ -#include <mach/dma.h> - struct tegra_pcm_dma_params { unsigned long addr; unsigned long wrap; -- 1.7.0.4

2 2

[alsa-devel] A question about speaker volume control of Creative Recon3D
by Ming Chen 15 Oct '12

15 Oct '12

Hello All, Recently I looked into the hda code for Creative sound cards, based on linux-3.6-rc3. I found that for Ca0110 (X-Fi), the patch (Patch_ca0110.c) would call add_out_volume() (and add_mono_volume()) for those DACs (refer to the ca0110_build_controls() function). While for ca0132 (Recon3D), its patch (Patch_ca0132.c) does not call add_out_volume() for any DAC (refer to the ca0132_build_controls() function). If I play a wav file under Ubuntu, it can adjust the speaker volume successfully with the Recon3D card. Can somebody kindly tell me how the speaker volume been controlled for this card? Thanks, Ming

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Re: [alsa-devel] [RFC] Fix for conflict of HDMI and SPDIF IEC958 controls
by Raymond Yau 15 Oct '12

15 Oct '12

> > > > > > > > > > > > how about those hdmi jack and iec958 on ad1989b? > > > > > > > > > > On AD codecs, all digital outputs are exposed as a single cloned > > > > > device with HDA_PCM_TYPE_SPDIF. > > > > > > > > how about ad1988b with nvidia codec ? > > > > > > Nvidia codec provides devices only with HDA_PCM_TYPE_HDMI, of course. > > > > > > > from user point of view , how can they differenitate hdmi jack is > > digital > > > > ouput and hdmi output ? > > > > > > The device 1 is from AD and the device 3 is from Nvidia codec, as you > > > can see below. And this is exactly the case the conflict happens as I > > > mentioned in the original mail. > > > > will your fix change the spdif of those motherboard with ad1988b which > > have iec958 but no hdmi jack ? > > It won't change anything unless both SPDIF and HDMI *devices* are > actually created. > > > > > > Which is connected to what output, you can't know exactly unless you > > > compare the obtained ELD. The HD-audio spec isn't good enough to > > > identify the actual output. > > > > it seem that there is no presence detect on this codec > > ELD has nothing to do with the presence detect bit. > ftp://download.nvidia.com/XFree86/gpu-hdmi-audio-document/gpu-hdmi-audio.ht… 6.1.2. Chipset with 8-channel support, single stream This configuration is used in MCP77, MCP78, MCP79, MCP7A, and ION. In the case where multiple HDMI display connectors are present, the audio stream is broadcast to all HDMI connectors at once. A single ALSA device is exposed. ELD and PD information is not available on these chipsets. In these chipsets, the multiple 2-channel converters are aggregated by the ALSA driver and exposed as a single 8-channel device. Some chipsets support 2, or 8 channels (MCP77). Other chipsets support 2, 6, or 8 channels (MCP79). id = 0x10de0002, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, does it mean that this hdmi codec support 8 channels but no ELD info ? > > > Codec: Nvidia MCP77/78 HDMI > > Address: 3 > > AFG Function Id: 0x1 (unsol 0) > > Vendor Id: 0x10de0002 > > Subsystem Id: 0x10de0101 > > Revision Id: 0x100000 > > No Modem Function Group found > > Default PCM: > > rates [0x0]: > > bits [0x0]: > > formats [0x0]: > > Default Amp-In caps: N/A > > Default Amp-Out caps: N/A > > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > > Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital > > Control: name="IEC958 Playback Con Mask", index=1, device=0 > > Control: name="IEC958 Playback Pro Mask", index=1, device=0 > > Control: name="IEC958 Playback Default", index=1, device=0 > > Control: name="IEC958 Playback Switch", index=1, device=0 > > Device: name="HDMI 0", type="HDMI", device=3 > > Converter: stream=0, channel=0 > > Digital: > > Digital category: 0x0 > > PCM: > > rates [0xc0]: 48000 88200 > > bits [0xf]: 8 16 20 24 > > formats [0x1]: PCM > > Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital > > Pincap 0x00000014: OUT Detect > > Pin Default 0x18560110: [Jack] Digital Out at Int HDMI > > Conn = Digital, Color = Unknown > > DefAssociation = 0x1, Sequence = 0x0 > > Misc = NO_PRESENCE > > Pin-ctls: 0x40: OUT > > Unsolicited: tag=00, enabled=0 > > Connection: 1 > > 0x04 > > > > > > > > > > > > https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/881826 > > > >

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[alsa-devel] Invitation to connect on LinkedIn
by Hao Song via LinkedIn 15 Oct '12

15 Oct '12

LinkedIn ------------ Hao Song requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Hao Accept invitation from Hao Song http://www.linkedin.com/e/jpxkk3-h8bk8zsc-2g/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Hao Song http://www.linkedin.com/e/jpxkk3-h8bk8zsc-2g/rso/64047763/aTEQ/name/1380363… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-h8bk8zsc-2g/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

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[alsa-devel] [asoc:topic/wm2200] sound/soc/codecs/wm2200.c:1033:4: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t'
by Yuanhan Liu 15 Oct '12

15 Oct '12

Hi Mark, FYI, there are new compile warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm2200 head: 9874482985fb26f90c3b2d8f1a82bd796d141304 commit: 6e87badd3f38e1a095d6e1b13828246c3e8486b5 ASoC: wm2200: Provide initial coefficient loading config: x86_64-allmodconfig # make ARCH=x86_64 allmodconfig All warnings: sound/soc/codecs/wm2200.c: In function 'wm2200_dsp_load': sound/soc/codecs/wm2200.c:1033:4: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t' [-Wformat] sound/soc/codecs/wm2200.c:1145:5: warning: format '%d' expects argument of type 'int', but argument 5 has type 'long unsigned int' [-Wformat] sound/soc/codecs/wm2200.c: In function 'wm2200_setup_algs': sound/soc/codecs/wm2200.c:1187:2: warning: format '%d' expects argument of type 'int', but argument 7 has type 'size_t' [-Wformat] sound/soc/codecs/wm2200.c:1207:5: warning: format '%x' expects argument of type 'unsigned int', but argument 3 has type 'long unsigned int' [-Wformat] sound/soc/codecs/wm2200.c: In function 'wm2200_load_coeff': sound/soc/codecs/wm2200.c:1270:4: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t' [-Wformat] sound/soc/codecs/wm2200.c:1334:5: warning: format '%d' expects argument of type 'int', but argument 5 has type 'long unsigned int' [-Wformat] vim +1033 sound/soc/codecs/wm2200.c e10f8711 Mark Brown 2012-10-04 1017 break; e10f8711 Mark Brown 2012-10-04 1018 default: e10f8711 Mark Brown 2012-10-04 1019 dev_err(codec->dev, "BASE %x\n", base); e10f8711 Mark Brown 2012-10-04 1020 BUG_ON(1); e10f8711 Mark Brown 2012-10-04 1021 return -EINVAL; e10f8711 Mark Brown 2012-10-04 1022 } e10f8711 Mark Brown 2012-10-04 1023 e10f8711 Mark Brown 2012-10-04 1024 ret = request_firmware(&firmware, file, codec->dev); e10f8711 Mark Brown 2012-10-04 1025 if (ret != 0) { e10f8711 Mark Brown 2012-10-04 1026 dev_err(codec->dev, "Failed to request '%s'\n", file); e10f8711 Mark Brown 2012-10-04 1027 return ret; e10f8711 Mark Brown 2012-10-04 1028 } e10f8711 Mark Brown 2012-10-04 1029 e10f8711 Mark Brown 2012-10-04 1030 pos = sizeof(*header) + sizeof(*adsp1_sizes) + sizeof(*footer); e10f8711 Mark Brown 2012-10-04 1031 if (pos >= firmware->size) { e10f8711 Mark Brown 2012-10-04 1032 dev_err(codec->dev, "%s: file too short, %d bytes\n", e10f8711 Mark Brown 2012-10-04 @1033 file, firmware->size); e10f8711 Mark Brown 2012-10-04 1034 return -EINVAL; e10f8711 Mark Brown 2012-10-04 1035 } e10f8711 Mark Brown 2012-10-04 1036 e10f8711 Mark Brown 2012-10-04 1037 header = (void*)&firmware->data[0]; e10f8711 Mark Brown 2012-10-04 1038 e10f8711 Mark Brown 2012-10-04 1039 if (memcmp(&header->magic[0], "WMFW", 4) != 0) { e10f8711 Mark Brown 2012-10-04 1040 dev_err(codec->dev, "%s: invalid magic\n", file); e10f8711 Mark Brown 2012-10-04 1041 return -EINVAL; --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] snd_ctl_elem_list
by Paul 15 Oct '12

15 Oct '12

Newbie. Trying to discover information about the audio hardware. Specifically about a USB microphone ( Audio-Technia AT2020USB ) and whether it has a volume control. Writing in Ada just to be difficult. Not having any problems with that though. After doing a malloc and a snd_ctl_elem_list I can use ctl_elem_list_count to get the number of elements. This number is non-zero ( 2 for the AT2020, 34 for the on-board Intel audio chip ). Then I can use ctl_elem_to get the number of used. That number is zero and prevents me from trying to access the element list. How can I find out information about the elements in the list? How do these elements become used? Is this even the correct path to try to discover the volume control? Thanks, Paul I

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Re: [alsa-devel] sound/pci/hda/hda_codec.c:1435:46: error: 'AC_VERB_GET_CON' undeclared
by Takashi Iwai 15 Oct '12

15 Oct '12

At Sat, 13 Oct 2012 01:01:09 +0800, Fengguang Wu wrote: > > Hi Takashi, > > FYI, kernel build failed on > > commit: 645cf3b5656e47fa9a46e2fbc8d212ab572ab05b Merge branch 'topic/3.8' > config: x86_64-allyesdebian (attached as .config) > > > All error/warnings: > > sound/pci/hda/hda_codec.c: In function 'update_pcm_stream_id': > sound/pci/hda/hda_codec.c:1435:46: error: 'AC_VERB_GET_CON' undeclared (first use in this function) > sound/pci/hda/hda_codec.c:1435:46: note: each undeclared identifier is reported only once for each function it appears in Thanks, this must be already fixed. It was a temporary merge error. (One shouldn't work during updating the system....) Takashi

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[alsa-devel] Support for Roland VG-99
by Pete Leigh 14 Oct '12

14 Oct '12

Hi, The VG-99 is a guitar processor with a usb interface which offers 2x16bit 44100 pcm in and out streams in its basic usb mode, and 2x24bit 44100 in/out + midi in the advanced mode connection mode. Basic mode works fine with usb snd driver, but the device isn't recognised when set to advanced mode (this is done under USB on page 2 of the "system" config menu) I've used the lsusb -v output (below) to guess at an addition to sound/usb/quirks-table.h to try and get this working. On initial (brief) testing, audio in/out at 24bits (as shown by jackd output) and midi (out, at least) are all working. I'll attach the quirk, and would be pleased if someone could look at it, and the lsusb output, and fix up any deficiencies (it's purely voodoo here :-D) and maybe get it or something like it into the official source... please let me know if changes need to be tested. Cheers, - Pete. Here's the lsusb output: Bus 001 Device 007: ID 0582:00b2 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 8 idVendor 0x0582 Roland Corp. idProduct 0x00b2 bcdDevice 1.00 iManufacturer 1 Roland iProduct 2 VG-99 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 2mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 03 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0120 1x 288 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 03 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0120 1x 288 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 02 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Device Status: 0x0001 Self Powered

1 0

[alsa-devel] snd-usb-audio probe problem (Teac UD-H01)
by bEN 13 Oct '12

13 Oct '12

Hi all, the new USB DAC from Teac seems detected as 'Human Interface Device' instead of 'Audio'. Despite the fact it is based on a standard USB Audio class v2 controller (Tenor 8802). Below is dmesg and lsusb outputs. Does it need to apply a patch in /sound/usb/quirks-table.h ? If yes, someone could help me to write this patch, please? Your help would be very much appreciated, Thanks in advance. bEN (from Ubuntu 11.04 with Linux 2.6.38-13-generic + Alsa driver 1.0.23) $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.23. $ dmesg (...) [34035.000078] usb 2-4: new high speed USB device using ehci_hcd and address 3 [34035.151753] usb 2-4: config 1 has an invalid interface number: 3 but max is 2 [34035.151763] usb 2-4: config 1 has an invalid interface number: 3 but max is 2 [34035.151770] usb 2-4: config 1 has an invalid interface number: 3 but max is 2 [34035.151777] usb 2-4: config 1 has no interface number 1 [34035.155108] input: TEAC TEAC USB AUDIO DEVICE as /devices/pci0000:00/0000:00:1d.7/usb2/2-4/2-4:1.0/input/input13 [34035.155328] generic-usb 0003:0644:8038.0002: input,hidraw1: USB HID v1.00 Device [TEAC TEAC USB AUDIO DEVICE] on usb-0000:00:1d.7-4/input0 [34035.202368] Audio class v2 interfaces need an interface association [34035.202387] snd-usb-audio: probe of 2-4:1.2 failed with error -5 [34035.202426] usbcore: registered new interface driver snd-usb-audio $ lsusb -v --> see attachment.

5 6

[alsa-devel] bit depth issue for Scarlett 2i4
by Dan MacDonald 13 Oct '12

13 Oct '12

Hi ALSA devs! I recently bought a Focusrite Scarlett 2i4 which is advertised as being capable of recording at a max of 24bit/96Khz but when I run alsacap it reports it as 32bit: ./alsacap -R -d hw:1 *** Exploring configuration space of device `hw:1' for recording *** 2 channels Sampling rate 44100..96000 Hz Sample formats: S32_LE Significant bits: 32 It does seem to record OK but that output seems wrong and I'm slightly concerned about the waste of disk space when multitracking if I end up with 32bit files but only 24 of those bits per sample are actually being used. Thanks for your help! Dan

2 1

[alsa-devel] Question about rawmidi USB Tx buffer size
by Paul D. DeRocco 12 Oct '12

12 Oct '12

Simple question: how big is it? ALSA provides snd_rawmidi_status for reading the status structure from a rawmidi handle, and snd_rawmidi_status_get_avail to extract the number of bytes in the ring buffer from that structure, but what I want to know is how many bytes of space are available in the ring buffer, so that I know how many bytes I can write without blocking. Is the buffer size a fixed number for any particular driver? The only driver I'm concerned with is the USB MIDI driver, so that shouldn't be hardware dependent. -- Ciao, Paul D. DeRocco Paul mailto:pderocco@ix.netcom.com

1 0

[alsa-devel] [PATCH 0/2] dmaengine: Fix compillation issues in device_prep_dma_cyclic()
by Peter Ujfalusi 12 Oct '12

12 Oct '12

Hello, The reason for the breakage is the patch 1 got lost somewhere in the process. I also greped the whole tree for device_prep_dma_cyclic() and the second patch is the result of that. PS: I did build tested the series on ARM (OMAP), x86_32, x86_64 but not for ppc for sure. Regards, Peter --- Peter Ujfalusi (2): dmaengine: Add flags parameter to dmaengine_prep_dma_cyclic() ARM: SAMSUNG: dma-ops: Fix dmaengine_prep_dma_cyclic() parameter list arch/arm/plat-samsung/dma-ops.c | 3 ++- include/linux/dmaengine.h | 3 ++- sound/soc/soc-dmaengine-pcm.c | 6 +++++- 3 files changed, 9 insertions(+), 3 deletions(-) -- 1.7.12

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Re: [alsa-devel] [3.6-rc7] switcheroo race with Intel HDA...
by Takashi Iwai 12 Oct '12

12 Oct '12

At Fri, 12 Oct 2012 23:08:14 +0800, Daniel J Blueman wrote: > > On 10 October 2012 20:34, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 9 Oct 2012 22:26:40 +0800, > > Daniel J Blueman wrote: > >> On 9 October 2012 21:04, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Tue, 9 Oct 2012 19:23:56 +0800, > >> > Daniel J Blueman wrote: > >> >> On 9 October 2012 18:07, Takashi Iwai <tiwai(a)suse.de> wrote: > >> >> > At Tue, 09 Oct 2012 12:04:08 +0200, > >> >> > Takashi Iwai wrote: > >> >> >> At Tue, 9 Oct 2012 00:34:09 +0800, > >> >> >> Daniel J Blueman wrote: > >> >> >> > On 8 October 2012 20:58, Takashi Iwai <tiwai(a)suse.de> wrote: > >> >> >> > > At Tue, 25 Sep 2012 13:20:05 +0800, > >> >> >> > > Daniel J Blueman wrote: > >> >> >> > >> On my Macbook with a discrete Nvidia GPU, there is a race between > >> >> >> > >> selecting the integrated GPU and putting the discrete GPU into D3 [1], > >> >> >> > >> reliably causing a kernel oops [2]. > >> >> >> > >> > >> >> >> > >> Introducing a delay of ~1s between the calls prevents this. When the > >> >> >> > >> second 'OFF' write path executes, it looks like struct azx at > >> >> >> > >> card->private_data hasn't yet been allocated yet [3], so there is > >> >> >> > >> likely some locking missing. > >> >> >> > > > >> >> >> > > It's rather pci_get_drvdata() returning NULL (i.e. card is NULL, thus > >> >> >> > > card->private_data causes Oops). Could you check the patch like below > >> >> >> > > and see whether you get a kernel warning (but no Oops) or the problem > >> >> >> > > gets fixed by shifting the assignment of pci drvdata? > >> >> >> > [...] > >> >> >> > > >> >> >> > Good patching. Calling pci_set_drvdata later prevents the oops in HDA, > >> >> >> > though we see unexpected 0x0 responses in the response ring buffer > >> >> >> > [1], which we don't see when there's a >~1.5s delay between IGD and > >> >> >> > OFF. > >> >> >> > >> >> >> If the previous patch fixed, it means that the switching occurred > >> >> >> during the device was being probed. Maybe a better approach to > >> >> >> register the VGA switcheroo after the proper initialization. > >> >> >> > >> >> >> The patch below is a revised one. Please give it a try. > >> >> > > >> >> > Also, it's not clear which card spews the spurious response. > >> >> > Apply the patch below in addition. > >> >> [...] > >> >> > >> >> hda-intel: 0000:01:00.1: spurious response 0x0:0x0, last cmd=0x1f0004 > >> >> $ lspci -s :1:0.1 > >> >> 01:00.1 Audio device: NVIDIA Corporation Device 0e1b (rev ff) > >> >> > >> >> It's the NVIDIA device which presumably hasn't completed it's > >> >> transition to D3 at the time the OFF is executed. > >> > > >> > OK, then could you try the patch below on the top of previous two > >> > patches? > >> > >> The first IGD switcheroo command fails to switch to the integrated GPU: > >> > >> # cat /sys/kernel/debug/vgaswitcheroo/switch > >> 0:DIS:+:Pwr:0000:01:00.0 > >> 1:IGD: :Pwr:0000:00:02.0 > >> 2:DIS-Audio: :Pwr:0000:01:00.1 > >> # echo IGD >/sys/kernel/debug/vgaswitcheroo/switch > >> vga_switcheroo: client 1 refused switch > >> > >> I also instrumented snd_hda_lock_devices, but none of the failure > >> paths are being taken, which would leave inconsistent state, as the > >> return value isn't checked. > > > > Hm, right, the return value of snd_hda_lock_devices() isn't checked, > > but I don't understand how this results like above. > > Basically switching is protected by mutex in vga_switcheroo.c, so the > > whole operation in the client side should be serialized. > > > > In anyway, try the patch below cleanly, and see the spurious message > > error coming up at which timing. > [...] > > The patch _does_ address the issue. A recent update to my Macbook > firmware misleadingly broke i915 switching, but since I can reproduce > the oops without the IGD switching completing with the stock kernel, > and consistently can't without [1], the patch is good. > > Tested-by: Daniel J Blueman <daniel(a)quora.org> OK, then I'm going to apply the patch to 3.7 tree (with Cc to stable). thanks, Takashi > > Thanks Takashi! > Daniel > > --- [1] > > snd_hda_intel 0000:00:1b.0: enabling device (0000 -> 0002) > snd_hda_intel 0000:00:1b.0: irq 54 for MSI/MSI-X > XXX 0000:00:1b.0: azx_codec_create entered > vga_switcheroo: enabled > XXX 0000:00:1b.0: azx_codec_create done > input: HDA Intel PCH Headphone as > /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 > snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) > hda_intel: Disabling MSI > hda-intel: 0000:01:00.1: Handle VGA-switcheroo audio client > XXX 0000:01:00.1: azx_codec_create entered > XXX 0000:01:00.1: azx_codec_create done > input: HDA NVidia HDMI/DP,pcm=8 as > /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input12 > input: HDA NVidia HDMI/DP,pcm=7 as > /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input13 > input: HDA NVidia HDMI/DP,pcm=3 as > /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input14 > vga_switcheroo: client 1 refused switch > i915: switched off > -- > Daniel J Blueman >

1 0

[alsa-devel] [PATCH 0/2] USB-audio disconnection race fixes
by Takashi Iwai 12 Oct '12

12 Oct '12

Hi, these two patches are for fixing races at disconnection of USB audio devices. The first patch tries to close the race at open/close, and the next is for other PCM and mixer ops in usb-audio driver. It seems working as far as I've tested with Matthieu's test cases. thanks, Takashi

1 2

[alsa-devel] [RFC PATCH] ALSA: hda - Implement a poll loop for jacks as a module parameter
by David Henningsson 12 Oct '12

12 Oct '12

Now that we have a generic unsol mechanism, we can implement a generic poll loop, which can be used for debugging, or if a codec's unsol mechanism is broken. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- I'm also considering activating it by default if we go into single_cmd mode due to codec not responding. What do you think? sound/pci/hda/hda_codec.c | 32 ++++++++++++++++++++++++++++---- sound/pci/hda/hda_codec.h | 2 ++ sound/pci/hda/hda_intel.c | 5 +++++ sound/pci/hda/hda_jack.c | 22 ++++++++++++++++++++++ sound/pci/hda/hda_jack.h | 2 ++ 5 files changed, 59 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c0ab72c..626e45b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1135,6 +1135,19 @@ static void restore_shutup_pins(struct hda_codec *codec) } #endif +static void hda_jackpoll_work(struct work_struct *work) +{ + struct hda_codec *codec = + container_of(work, struct hda_codec, jackpoll_work.work); + if (!codec->jackpoll_interval) + return; + + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_poll_all(codec); + queue_delayed_work(codec->bus->workq, &codec->jackpoll_work, + codec->jackpoll_interval); +} + static void init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size); static void free_hda_cache(struct hda_cache_rec *cache); @@ -1190,6 +1203,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) { if (!codec) return; + cancel_delayed_work_sync(&codec->jackpoll_work); snd_hda_jack_tbl_clear(codec); restore_init_pincfgs(codec); #ifdef CONFIG_PM @@ -1273,6 +1287,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64); snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); + INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); #ifdef CONFIG_PM spin_lock_init(&codec->power_lock); @@ -2349,7 +2364,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) return -EBUSY; /* OK, let it free */ - + cancel_delayed_work_sync(&codec->jackpoll_work); #ifdef CONFIG_PM cancel_delayed_work_sync(&codec->power_work); codec->power_on = 0; @@ -3646,7 +3661,6 @@ static void hda_call_codec_resume(struct hda_codec *codec) restore_pincfgs(codec); /* restore all current pin configs */ restore_shutup_pins(codec); hda_exec_init_verbs(codec); - snd_hda_jack_set_dirty_all(codec); if (codec->patch_ops.resume) codec->patch_ops.resume(codec); else { @@ -3655,7 +3669,13 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); } - snd_hda_jack_report_sync(codec); + + if (codec->jackpoll_interval) + hda_jackpoll_work(&codec->jackpoll_work.work); + else { + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_report_sync(codec); + } snd_hda_power_down(codec); /* flag down before returning */ } #endif /* CONFIG_PM */ @@ -3737,7 +3757,10 @@ int snd_hda_codec_build_controls(struct hda_codec *codec) if (err < 0) return err; - snd_hda_jack_report_sync(codec); /* call at the last init point */ + if (codec->jackpoll_interval) + hda_jackpoll_work(&codec->jackpoll_work.work); + else + snd_hda_jack_report_sync(codec); /* call at the last init point */ return 0; } @@ -5128,6 +5151,7 @@ int snd_hda_suspend(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { + cancel_delayed_work_sync(&codec->jackpoll_work); if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec, false); } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 507fe8a..10a03b0 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -884,6 +884,8 @@ struct hda_codec { /* jack detection */ struct snd_array jacktbl; + unsigned long jackpoll_interval; /* In jiffies. Zero means no poll, rely on unsol events */ + struct delayed_work jackpoll_work; #ifdef CONFIG_SND_HDA_INPUT_JACK /* jack detection */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f09ff6c..b4f1ab8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -68,6 +68,7 @@ static int position_fix[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_only[SNDRV_CARDS]; +static int jackpoll_ms[SNDRV_CARDS]; static bool single_cmd; static int enable_msi = -1; #ifdef CONFIG_SND_HDA_PATCH_LOADER @@ -95,6 +96,8 @@ module_param_array(probe_mask, int, NULL, 0444); MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); module_param_array(probe_only, int, NULL, 0444); MODULE_PARM_DESC(probe_only, "Only probing and no codec initialization."); +module_param_array(jackpoll_ms, int, NULL, 0444); +MODULE_PARM_DESC(jackpoll_ms, "Ms between polling for jack events (default = 0, using unsol events only)"); module_param(single_cmd, bool, 0444); MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs " "(for debugging only)."); @@ -1666,6 +1669,8 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode err = snd_hda_codec_new(chip->bus, c, &codec); if (err < 0) continue; + codec->jackpoll_interval = + msecs_to_jiffies(jackpoll_ms[chip->dev_index]); codec->beep_mode = chip->beep_mode; codecs++; } diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 5c690cb..07d5288 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -439,3 +439,25 @@ void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) } EXPORT_SYMBOL_HDA(snd_hda_jack_unsol_event); +void snd_hda_jack_poll_all(struct hda_codec *codec) +{ + struct hda_jack_tbl *jack = codec->jacktbl.list; + int i, changes = 0; + + for (i = 0; i < codec->jacktbl.used; i++, jack++) { + unsigned int old_sense; + if (!jack->nid || !jack->jack_dirty || jack->phantom_jack) + continue; + old_sense = get_jack_plug_state(jack->pin_sense); + jack_detect_update(codec, jack); + if (old_sense == get_jack_plug_state(jack->pin_sense)) + continue; + changes = 1; + if (jack->callback) + jack->callback(codec, jack); + } + if (changes) + snd_hda_jack_report_sync(codec); +} +EXPORT_SYMBOL_HDA(snd_hda_jack_poll_all); + diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index af8dd47..4487785 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -84,4 +84,6 @@ void snd_hda_jack_report_sync(struct hda_codec *codec); void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res); +void snd_hda_jack_poll_all(struct hda_codec *codec); + #endif /* __SOUND_HDA_JACK_H */ -- 1.7.9.5

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Re: [alsa-devel] [PATCH 2.6.32.y, 2.6.34.y] ALSA: hda - ALSA HD Audio patch for Intel Panther Point DeviceIDs
by Ben Hutchings 12 Oct '12

12 Oct '12

On Tue, 2012-10-09 at 10:56 -0400, Paul Gortmaker wrote: > On 12-10-08 09:55 PM, Ben Hutchings wrote: > > On Mon, 2012-10-08 at 18:07 -0700, Jonathan Nieder wrote: > >> From: Seth Heasley <seth.heasley(a)intel.com> > >> Date: Wed, 20 Apr 2011 10:59:57 -0700 > >> > >> commit d2edeb7c6f1dada8ca7d5c23e42d604e92ae0c76 upstream. > >> > >> This patch adds the HD Audio Controller DeviceIDs for the Intel Panther Point PCH. > >> > >> [jn: backported for 2.6.32.y by Ana Guerrero] > >> > >> Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> > >> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > >> Tested-by: Ana Guerrero <ana(a)debian.org> # EliteBook 8570w > >> Signed-off-by: Jonathan Nieder <jrnieder(a)gmail.com> > >> --- > >> Hi Willy and Paul, > >> > >> Please consider > >> > >> d2edeb7c6f1d ALSA: hda - ALSA HD Audio patch for Intel Panther Point > >> DeviceIDs > >> > >> for application to the 2.6.32.y and 2.6.34.y trees. > >> > >> It does what it says on the cover. The patch was merged in the 3.0 > >> cycle, so newer stable kernels don't need it. Backported and > >> tested[1] against Debian's 2.6.32.y-based kernel by Ana (cc-ed) -- > >> thanks! > >> > >> Thoughts of all kinds welcome, as always. > > [...] > > > > I queued up a whole series of device ID updates for Debian stable, as > > they didn't obviously depend on other changes: > > > > cea310e ALSA: hda_intel: ALSA HD Audio patch for Intel Patsburg DeviceIDs > > e35d4b1 ALSA: hda: add Vortex86MX PCI ids > > 0f0714c ALSA: hda - Add support for VMware controller > > d2edeb7 ALSA: hda - ALSA HD Audio patch for Intel Panther Point DeviceIDs > > I found that d2edeb7c6f depends (only trivially, on context) on b686453543f, > which makes a blanket class for intel IDs so that "...the driver will work > with any new control chips in future." In that respect, I guess it too > (b68645) is in the same class as the other "add more IDs" patches? Yes, it seems like this would also be worth adding. (Always assuming that any functional changes it depends on are also small enough for stable.) It's not a clean cherry-pick as the line above has changed. Ben. > Paul. > -- > > > > > (They can be cherry-picked cleanly in the above order.) But if I've > > missed some post-2.6.32 dependencies then I would like to know. > > > > Ben. > > > -- Ben Hutchings Kids! Bringing about Armageddon can be dangerous. Do not attempt it in your own home. - Terry Pratchett and Neil Gaiman, `Good Omens'

1 0

[alsa-devel] Arecord issue
by kartik natarajan 11 Oct '12

11 Oct '12

Hi, I am working on a 2.6.37 kernel and on a DM365 davinci board with a tlv320aic3254 audio codec (i2c control lines, i2s data transfer) I wanted to get the audio up on my board so got the alsa-lib and alsa-utils 1.0.25 compiled them and got the aplay to work fine. When I got to working on the arecord things haven't worked fine. I have traced all the calls and have reached so far; codec gets set properly davinci_pcm_trigger executes with success snd_pcm_sw_params() executes successfully. aplay calls begin_wave() aplay calls pcm_read() that invokes snd_pcm_readi() Finally my app seems to not go beyond the snd_pcm_start() Any assistance shall be highly appreciated! Thanks in advance, N.Kartik

2 3

[alsa-devel] [PATCH] ASoC: codecs: da9055: Minor improvement in ALC calibration process
by Ashish Chavan 11 Oct '12

11 Oct '12

This patch slightly improves ALC calibration process of da9055 codec driver by muting Mic PGAs during calibration. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/da9055.c | 22 +++++++++++++++++++++- 1 files changed, 21 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 27c4454..abf3e43 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -179,6 +179,12 @@ #define DA9055_AIF_WORD_S24_LE (2 << 2) #define DA9055_AIF_WORD_S32_LE (3 << 2) +/* MIC_L_CTRL bit fields */ +#define DA9055_MIC_L_MUTE_EN (1 << 6) + +/* MIC_R_CTRL bit fields */ +#define DA9055_MIC_R_MUTE_EN (1 << 6) + /* MIXIN_L_CTRL bit fields */ #define DA9055_MIXIN_L_MIX_EN (1 << 3) @@ -477,7 +483,7 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u8 reg_val, adc_left, adc_right; + u8 reg_val, adc_left, adc_right, mic_left, mic_right; int avg_left_data, avg_right_data, offset_l, offset_r; if (ucontrol->value.integer.value[0]) { @@ -486,6 +492,16 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, * offsets must be done first */ + /* Save current values from Mic control registers */ + mic_left = snd_soc_read(codec, DA9055_MIC_L_CTRL); + mic_right = snd_soc_read(codec, DA9055_MIC_R_CTRL); + + /* Mute Mic PGA Left and Right */ + snd_soc_update_bits(codec, DA9055_MIC_L_CTRL, + DA9055_MIC_L_MUTE_EN, DA9055_MIC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_MIC_R_CTRL, + DA9055_MIC_R_MUTE_EN, DA9055_MIC_R_MUTE_EN); + /* Save current values from ADC control registers */ adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); @@ -521,6 +537,10 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, /* Restore original values of ADC control registers */ snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + + /* Restore original values of Mic control registers */ + snd_soc_write(codec, DA9055_MIC_L_CTRL, mic_left); + snd_soc_write(codec, DA9055_MIC_R_CTRL, mic_right); } return snd_soc_put_volsw(kcontrol, ucontrol); -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: codecs: da9055: Add support for DSP mode of AIF
by Ashish Chavan 11 Oct '12

11 Oct '12

This patch adds AIF DSP mode support for da9055 codec driver. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/da9055.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 185d8dd..27c4454 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -173,6 +173,7 @@ #define DA9055_AIF_FORMAT_I2S_MODE (0 << 0) #define DA9055_AIF_FORMAT_LEFT_J (1 << 0) #define DA9055_AIF_FORMAT_RIGHT_J (2 << 0) +#define DA9055_AIF_FORMAT_DSP (3 << 0) #define DA9055_AIF_WORD_S16_LE (0 << 2) #define DA9055_AIF_WORD_S20_3LE (1 << 2) #define DA9055_AIF_WORD_S24_LE (2 << 2) @@ -1155,6 +1156,9 @@ static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_RIGHT_J: aif_ctrl = DA9055_AIF_FORMAT_RIGHT_J; break; + case SND_SOC_DAIFMT_DSP_A: + aif_ctrl = DA9055_AIF_FORMAT_DSP; + break; default: return -EINVAL; } -- 1.7.1

2 1

[alsa-devel] [PATCH V7 12/15] ALSA: HDA: Make hda sound card usable for Loongson
by Huacai Chen 11 Oct '12

11 Oct '12

Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec, this patch modify patch_conexant.c to add Lemote specific code. Both A1004 and A1205 use the same pin configurations, but A1004 need to increase the default boost of internal mic. Signed-off-by: Jie Chen <chenj(a)lemote.com> Signed-off-by: Huacai Chen <chenhc(a)lemote.com> Signed-off-by: Hongliang Tao <taohl(a)lemote.com> Signed-off-by: Hua Yan <yanh(a)lemote.com> Cc: alsa-devel(a)alsa-project.org --- sound/pci/hda/patch_conexant.c | 44 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e22a8f..3cc265e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4408,7 +4408,10 @@ static const struct hda_codec_ops cx_auto_patch_ops = { enum { CXT_PINCFG_LENOVO_X200, CXT_PINCFG_LENOVO_TP410, + CXT_PINCFG_LEMOTE_A1004, + CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, + CXT_FIXUP_INC_MIC_BOOST, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -4418,6 +4421,19 @@ static void cxt_fixup_stereo_dmic(struct hda_codec *codec, spec->fixup_stereo_dmic = 1; } +static void cxt5066_increase_mic_boost(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_override_amp_caps(codec, 0x17, HDA_OUTPUT, + (0x3 << AC_AMPCAP_OFFSET_SHIFT) | + (0x4 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); +} + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -4435,6 +4451,18 @@ static const struct hda_pintbl cxt_pincfg_lenovo_tp410[] = { {} }; +/* Lemote A1004/A1205 with cxt5066 */ +static const struct hda_pintbl cxt_pincfg_lemote[] = { + { 0x1a, 0x90a10020 }, /* Internal mic */ + { 0x1b, 0x03a11020 }, /* External mic */ + { 0x1d, 0x400101f0 }, /* Not used */ + { 0x1e, 0x40a701f0 }, /* Not used */ + { 0x20, 0x404501f0 }, /* Not used */ + { 0x22, 0x404401f0 }, /* Not used */ + { 0x23, 0x40a701f0 }, /* Not used */ + {} +}; + static const struct hda_fixup cxt_fixups[] = { [CXT_PINCFG_LENOVO_X200] = { .type = HDA_FIXUP_PINS, @@ -4444,10 +4472,24 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_lenovo_tp410, }, + [CXT_PINCFG_LEMOTE_A1004] = { + .type = HDA_FIXUP_PINS, + .chained = true, + .chain_id = CXT_FIXUP_INC_MIC_BOOST, + .v.pins = cxt_pincfg_lemote, + }, + [CXT_PINCFG_LEMOTE_A1205] = { + .type = HDA_FIXUP_PINS, + .v.pins = cxt_pincfg_lemote, + }, [CXT_FIXUP_STEREO_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_stereo_dmic, }, + [CXT_FIXUP_INC_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt5066_increase_mic_boost, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -4463,6 +4505,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), + SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), {} }; -- 1.7.7.3

2 2

[alsa-devel] [PATCH] ASoC: codecs: da9055: Minor improvements
by Ashish Chavan 11 Oct '12

11 Oct '12

This patch includes two minor improvements for da9055 codec driver (1) Adding support for DSP mode of AIF (2) Muting Mic PGAs during ALC calibration process Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/da9055.c | 26 +++++++++++++++++++++++++- 1 files changed, 25 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 185d8dd..abf3e43 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -173,11 +173,18 @@ #define DA9055_AIF_FORMAT_I2S_MODE (0 << 0) #define DA9055_AIF_FORMAT_LEFT_J (1 << 0) #define DA9055_AIF_FORMAT_RIGHT_J (2 << 0) +#define DA9055_AIF_FORMAT_DSP (3 << 0) #define DA9055_AIF_WORD_S16_LE (0 << 2) #define DA9055_AIF_WORD_S20_3LE (1 << 2) #define DA9055_AIF_WORD_S24_LE (2 << 2) #define DA9055_AIF_WORD_S32_LE (3 << 2) +/* MIC_L_CTRL bit fields */ +#define DA9055_MIC_L_MUTE_EN (1 << 6) + +/* MIC_R_CTRL bit fields */ +#define DA9055_MIC_R_MUTE_EN (1 << 6) + /* MIXIN_L_CTRL bit fields */ #define DA9055_MIXIN_L_MIX_EN (1 << 3) @@ -476,7 +483,7 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u8 reg_val, adc_left, adc_right; + u8 reg_val, adc_left, adc_right, mic_left, mic_right; int avg_left_data, avg_right_data, offset_l, offset_r; if (ucontrol->value.integer.value[0]) { @@ -485,6 +492,16 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, * offsets must be done first */ + /* Save current values from Mic control registers */ + mic_left = snd_soc_read(codec, DA9055_MIC_L_CTRL); + mic_right = snd_soc_read(codec, DA9055_MIC_R_CTRL); + + /* Mute Mic PGA Left and Right */ + snd_soc_update_bits(codec, DA9055_MIC_L_CTRL, + DA9055_MIC_L_MUTE_EN, DA9055_MIC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_MIC_R_CTRL, + DA9055_MIC_R_MUTE_EN, DA9055_MIC_R_MUTE_EN); + /* Save current values from ADC control registers */ adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); @@ -520,6 +537,10 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, /* Restore original values of ADC control registers */ snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + + /* Restore original values of Mic control registers */ + snd_soc_write(codec, DA9055_MIC_L_CTRL, mic_left); + snd_soc_write(codec, DA9055_MIC_R_CTRL, mic_right); } return snd_soc_put_volsw(kcontrol, ucontrol); @@ -1155,6 +1176,9 @@ static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_RIGHT_J: aif_ctrl = DA9055_AIF_FORMAT_RIGHT_J; break; + case SND_SOC_DAIFMT_DSP_A: + aif_ctrl = DA9055_AIF_FORMAT_DSP; + break; default: return -EINVAL; } -- 1.7.1

2 1

[alsa-devel] Bug 48381 - [Regression][Bisected]Sound gets permanently muted a few minutes after booting
by Da Fox 10 Oct '12

10 Oct '12

Hi all, Since the official bug-tracker at https://bugtrack.alsa-project.org/alsa-bug/ seems to be down (and moreover this appears to be a kernel bug), I have filed a bug at the kernel's bug-tracker last week ( https://bugzilla.kernel.org/show_bug.cgi?id=48381 ). However, it seems to have gone unnoticed thus-far. Quick summary of the issue: After a while of inactivity the sound-card seems to go into a sort of power-saving mode (you can hear a faint 'pop' from the speakers). Before the sound would automatically turn back on again as soon as a program attempted to play sound. However now it seems that after resuming from the power-saving mode the driver thinks that there is always a headphone connected, even when it is not. I think this is what happens because toggling the 'Auto-Mute Mode' (setting it to 'Disabled') in alsamixer allows the speakers to produce sound again. This is all on a laptop (please see the linked bug report for more detailed hardware description). Would a developer please take a look at it? I have bisected the issue down to a commit made by David Henningsson and/or Takashi Iwai: ---8<--------- 80c8bfbe76869bfd6bdf3d260d316e7a32f318c3 is the first bad commit commit 80c8bfbe76869bfd6bdf3d260d316e7a32f318c3 Author: David Henningsson <david.henningsson(a)canonical.com> Date: Mon Jun 4 09:33:51 2012 +0200 ALSA: HDA: Create phantom jacks for fixed inputs and outputs PulseAudio sometimes have difficulties knowing that there is a "Speaker" or "Internal Mic", if they have no individual volume controls or selectors. As a result, only e g "Headphone" might be created for a laptop, but no "Speaker". To help out, create phantom jacks (that are always present, at least for now) for "Speaker", "Internal Mic" etc, in case we detect them. The naming convention is e g "Speaker Phantom Jack". In order not to pollute the /dev/input namespace with even more devices, these are added to the kcontrols only, not the input devices. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 7a954b731bef80cee731763521f64a424a7058a0 7d176b02ebb4553b0b0f97d200f7527cb69c122a M sound --->8--------- If any additional information is required please let me know. Best regards, Da Fox

3 6

[alsa-devel] ASoC: special SPI format for that5173 driver in daisy chain mode
by Christian Schörkhuber 10 Oct '12

10 Oct '12

Hi all, I'm working on an ASoC driver for the THAT5173 microphone preamplifier which is controlled via SPI. When operating in daisy chain mode (4 amps) the desired format is a bit odd: I need to write four 16 bit spi commands (one gain setting for each amp) in a row without raising the chipselect line between commands. That is, I need to write one 64 bit SPI command that is composed of 4 different values (and 4 register addresses which are all 0). Usually I use regmap but I don't see how I can construct such a message with regmap. How can this be done (with or without regmap)? Thanks, Chris

1 0

[alsa-devel] [PATCH] ASoC: fsl/pcm030 add unique dai_link names
by Eric Millbrandt 10 Oct '12

10 Oct '12

From: Eric Millbrandt <emillbrandt(a)dekaresearch.com Assign each dai_link a unique name to avoid this run-time error. [ 18.978043] pcm030-audio-fabric sound.2: wm9712-hifi <-> mpc5200-psc-ac97.0 mapping ok [ 19.003179] sysfs: cannot create duplicate filename '/devices/sound.2/AC97' Signed-off-by: Eric Millbrandt <emillbrandt(a)dekaresearch.com> --- sound/soc/fsl/pcm030-audio-fabric.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 4b63ec8..4d26192 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -29,14 +29,14 @@ struct pcm030_audio_data { static struct snd_soc_dai_link pcm030_fabric_dai[] = { { - .name = "AC97", + .name = "AC97.0", .stream_name = "AC97 Analog", .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "mpc5200-psc-ac97.0", .codec_name = "wm9712-codec", }, { - .name = "AC97", + .name = "AC97.1", .stream_name = "AC97 IEC958", .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", -- 1.7.8

1 1

[alsa-devel] [PATCH] ALSA: ASoC: davinci-pcm: stub out gen_pool functions for !CONFIG_GENERIC_ALLOCATOR
by Daniel Mack 10 Oct '12

10 Oct '12

On platforms without CONFIG_GENERIC_ALLOCATOR, these functions can be no-ops. The code that uses the memory buffers does the right thing if they are NULL. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> Cc: Matt Porter <mporter(a)ti.com> Cc: Anil Kumar <anilkumar.v(a)ti.com> --- Alternatively, the gen_pool functions could stub out themselves for !CONFIG_GENERIC_ALLOCATOR. sound/soc/davinci/davinci-pcm.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 01b3f17..afab81f 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -251,6 +251,7 @@ static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) } } +#ifdef CONFIG_GENERIC_ALLOCATOR static int allocate_sram(struct snd_pcm_substream *substream, struct gen_pool *sram_pool, unsigned size, struct snd_pcm_hardware *ppcm) @@ -284,6 +285,28 @@ exit1: return -ENOMEM; } +static void davinci_free_sram(struct snd_pcm_substream *substream, + struct snd_dma_buffer *iram_dma) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct gen_pool *sram_pool = prtd->params->sram_pool; + + gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes); +} +#else +static int allocate_sram(struct snd_pcm_substream *substream, + struct gen_pool *sram_pool, unsigned size, + struct snd_pcm_hardware *ppcm) +{ + return 0; +} + +static void davinci_free_sram(struct snd_pcm_substream *substream, + struct snd_dma_buffer *iram_dma) +{ +} +#endif + /* * Only used with ping/pong. * This is called after runtime->dma_addr, period_bytes and data_type are valid @@ -813,11 +836,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) buf->area = NULL; iram_dma = buf->private_data; if (iram_dma) { - struct davinci_runtime_data *prtd = - substream->runtime->private_data; - struct gen_pool *sram_pool = prtd->params->sram_pool; - gen_pool_free(sram_pool, (unsigned)iram_dma->area, - iram_dma->bytes); + davinci_free_sram(substream, iram_dma); kfree(iram_dma); } } -- 1.7.11.4

1 1

[alsa-devel] [PATCH] ASoC: wm8958: Load firmwares simultaneously
by Mark Brown 10 Oct '12

10 Oct '12

As of commit 99c2aa (firmware loader: fix creation failure of fw loader device) we can have more than one firmware request outstanding at once so there is no need to daisychain our requests any more. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8958-dsp2.c | 31 +++++++++++-------------------- 1 file changed, 11 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 2f191a0..b0710d8 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -883,13 +883,6 @@ static void wm8958_mbc_vss_loaded(const struct firmware *fw, void *context) wm8994->mbc_vss = fw; mutex_unlock(&codec->mutex); } - - /* We can't have more than one request outstanding at once so - * we daisy chain. - */ - request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, - "wm8958_enh_eq.wfw", codec->dev, GFP_KERNEL, - codec, wm8958_enh_eq_loaded); } static void wm8958_mbc_loaded(const struct firmware *fw, void *context) @@ -897,19 +890,11 @@ static void wm8958_mbc_loaded(const struct firmware *fw, void *context) struct snd_soc_codec *codec = context; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - if (wm8958_dsp2_fw(codec, "MBC", fw, true) != 0) - return; - - mutex_lock(&codec->mutex); - wm8994->mbc = fw; - mutex_unlock(&codec->mutex); - - /* We can't have more than one request outstanding at once so - * we daisy chain. - */ - request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, - "wm8958_mbc_vss.wfw", codec->dev, GFP_KERNEL, - codec, wm8958_mbc_vss_loaded); + if (fw && (wm8958_dsp2_fw(codec, "MBC", fw, true) == 0)) { + mutex_lock(&codec->mutex); + wm8994->mbc = fw; + mutex_unlock(&codec->mutex); + } } void wm8958_dsp2_init(struct snd_soc_codec *codec) @@ -933,6 +918,12 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, "wm8958_mbc.wfw", codec->dev, GFP_KERNEL, codec, wm8958_mbc_loaded); + request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + "wm8958_mbc_vss.wfw", codec->dev, GFP_KERNEL, + codec, wm8958_mbc_vss_loaded); + request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + "wm8958_enh_eq.wfw", codec->dev, GFP_KERNEL, + codec, wm8958_enh_eq_loaded); if (pdata->num_mbc_cfgs) { struct snd_kcontrol_new control[] = { -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm8750: Convert to direct regmap API usage
by Mark Brown 10 Oct '12

10 Oct '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8750.c | 86 ++++++++++++++++++++++++++++++++++----------- 1 file changed, 66 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 89151ca..7665d68 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -18,6 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -34,24 +35,55 @@ * We can't read the WM8750 register space when we * are using 2 wire for device control, so we cache them instead. */ -static const u16 wm8750_reg[] = { - 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ - 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ - 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ - 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ - 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ - 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ - 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ - 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ - 0x0079, 0x0079, 0x0079, /* 40 */ +static const struct reg_default wm8750_reg_defaults[] = { + { 0, 0x0097 }, + { 1, 0x0097 }, + { 2, 0x0079 }, + { 3, 0x0079 }, + { 4, 0x0000 }, + { 5, 0x0008 }, + { 6, 0x0000 }, + { 7, 0x000a }, + { 8, 0x0000 }, + { 9, 0x0000 }, + { 10, 0x00ff }, + { 11, 0x00ff }, + { 12, 0x000f }, + { 13, 0x000f }, + { 14, 0x0000 }, + { 15, 0x0000 }, + { 16, 0x0000 }, + { 17, 0x007b }, + { 18, 0x0000 }, + { 19, 0x0032 }, + { 20, 0x0000 }, + { 21, 0x00c3 }, + { 22, 0x00c3 }, + { 23, 0x00c0 }, + { 24, 0x0000 }, + { 25, 0x0000 }, + { 26, 0x0000 }, + { 27, 0x0000 }, + { 28, 0x0000 }, + { 29, 0x0000 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0050 }, + { 35, 0x0050 }, + { 36, 0x0050 }, + { 37, 0x0050 }, + { 38, 0x0050 }, + { 39, 0x0050 }, + { 40, 0x0079 }, + { 41, 0x0079 }, + { 42, 0x0079 }, }; /* codec private data */ struct wm8750_priv { unsigned int sysclk; - enum snd_soc_control_type control_type; }; #define wm8750_reset(c) snd_soc_write(c, WM8750_RESET, 0) @@ -668,10 +700,9 @@ static int wm8750_resume(struct snd_soc_codec *codec) static int wm8750_probe(struct snd_soc_codec *codec) { - struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8750->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret); return ret; @@ -711,9 +742,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8750 = { .suspend = wm8750_suspend, .resume = wm8750_resume, .set_bias_level = wm8750_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8750_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8750_reg, .controls = wm8750_snd_controls, .num_controls = ARRAY_SIZE(wm8750_snd_controls), @@ -730,10 +758,21 @@ static const struct of_device_id wm8750_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8750_of_match); +static const struct regmap_config wm8750_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8750_MOUTV, + + .reg_defaults = wm8750_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8750_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8750_spi_probe(struct spi_device *spi) { struct wm8750_priv *wm8750; + struct regmap *regmap; int ret; wm8750 = devm_kzalloc(&spi->dev, sizeof(struct wm8750_priv), @@ -741,7 +780,10 @@ static int __devinit wm8750_spi_probe(struct spi_device *spi) if (wm8750 == NULL) return -ENOMEM; - wm8750->control_type = SND_SOC_SPI; + regmap = devm_regmap_init_spi(spi, &wm8750_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + spi_set_drvdata(spi, wm8750); ret = snd_soc_register_codec(&spi->dev, @@ -779,6 +821,7 @@ static __devinit int wm8750_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8750_priv *wm8750; + struct regmap *regmap; int ret; wm8750 = devm_kzalloc(&i2c->dev, sizeof(struct wm8750_priv), @@ -787,7 +830,10 @@ static __devinit int wm8750_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm8750); - wm8750->control_type = SND_SOC_I2C; + + regmap = devm_regmap_init_i2c(i2c, &wm8750_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8750, &wm8750_dai, 1); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm8971: Convert to direct regmap API usage
by Mark Brown 10 Oct '12

10 Oct '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8971.c | 80 ++++++++++++++++++++++++++++++++++----------- 1 file changed, 61 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 5ce6477..70dd27e 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -19,6 +19,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -34,7 +35,6 @@ static struct workqueue_struct *wm8971_workq = NULL; /* codec private data */ struct wm8971_priv { - enum snd_soc_control_type control_type; unsigned int sysclk; }; @@ -43,18 +43,50 @@ struct wm8971_priv { * We can't read the WM8971 register space when we * are using 2 wire for device control, so we cache them instead. */ -static const u16 wm8971_reg[] = { - 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ - 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ - 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ - 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ - 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ - 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ - 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ - 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ - 0x0079, 0x0079, 0x0079, /* 40 */ +static const struct reg_default wm8971_reg_defaults[] = { + { 0, 0x0097 }, + { 1, 0x0097 }, + { 2, 0x0079 }, + { 3, 0x0079 }, + { 4, 0x0000 }, + { 5, 0x0008 }, + { 6, 0x0000 }, + { 7, 0x000a }, + { 8, 0x0000 }, + { 9, 0x0000 }, + { 10, 0x00ff }, + { 11, 0x00ff }, + { 12, 0x000f }, + { 13, 0x000f }, + { 14, 0x0000 }, + { 15, 0x0000 }, + { 16, 0x0000 }, + { 17, 0x007b }, + { 18, 0x0000 }, + { 19, 0x0032 }, + { 20, 0x0000 }, + { 21, 0x00c3 }, + { 22, 0x00c3 }, + { 23, 0x00c0 }, + { 24, 0x0000 }, + { 25, 0x0000 }, + { 26, 0x0000 }, + { 27, 0x0000 }, + { 28, 0x0000 }, + { 29, 0x0000 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0050 }, + { 35, 0x0050 }, + { 36, 0x0050 }, + { 37, 0x0050 }, + { 38, 0x0050 }, + { 39, 0x0050 }, + { 40, 0x0079 }, + { 41, 0x0079 }, + { 42, 0x0079 }, }; #define wm8971_reset(c) snd_soc_write(c, WM8971_RESET, 0) @@ -613,11 +645,10 @@ static int wm8971_resume(struct snd_soc_codec *codec) static int wm8971_probe(struct snd_soc_codec *codec) { - struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8971->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret); return ret; @@ -667,9 +698,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8971 = { .suspend = wm8971_suspend, .resume = wm8971_resume, .set_bias_level = wm8971_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8971_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8971_reg, .controls = wm8971_snd_controls, .num_controls = ARRAY_SIZE(wm8971_snd_controls), @@ -679,10 +707,21 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8971 = { .num_dapm_routes = ARRAY_SIZE(wm8971_dapm_routes), }; +static const struct regmap_config wm8971_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8971_MOUTV, + + .reg_defaults = wm8971_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8971_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + static __devinit int wm8971_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8971_priv *wm8971; + struct regmap *regmap; int ret; wm8971 = devm_kzalloc(&i2c->dev, sizeof(struct wm8971_priv), @@ -690,7 +729,10 @@ static __devinit int wm8971_i2c_probe(struct i2c_client *i2c, if (wm8971 == NULL) return -ENOMEM; - wm8971->control_type = SND_SOC_I2C; + regmap = devm_regmap_init_i2c(i2c, &wm8971_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + i2c_set_clientdata(i2c, wm8971); ret = snd_soc_register_codec(&i2c->dev, -- 1.7.10.4

1 0

Re: [alsa-devel] [3.6-rc7] switcheroo race with Intel HDA...
by Takashi Iwai 10 Oct '12

10 Oct '12

At Tue, 9 Oct 2012 22:26:40 +0800, Daniel J Blueman wrote: > > On 9 October 2012 21:04, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 9 Oct 2012 19:23:56 +0800, > > Daniel J Blueman wrote: > >> On 9 October 2012 18:07, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Tue, 09 Oct 2012 12:04:08 +0200, > >> > Takashi Iwai wrote: > >> >> At Tue, 9 Oct 2012 00:34:09 +0800, > >> >> Daniel J Blueman wrote: > >> >> > On 8 October 2012 20:58, Takashi Iwai <tiwai(a)suse.de> wrote: > >> >> > > At Tue, 25 Sep 2012 13:20:05 +0800, > >> >> > > Daniel J Blueman wrote: > >> >> > >> On my Macbook with a discrete Nvidia GPU, there is a race between > >> >> > >> selecting the integrated GPU and putting the discrete GPU into D3 [1], > >> >> > >> reliably causing a kernel oops [2]. > >> >> > >> > >> >> > >> Introducing a delay of ~1s between the calls prevents this. When the > >> >> > >> second 'OFF' write path executes, it looks like struct azx at > >> >> > >> card->private_data hasn't yet been allocated yet [3], so there is > >> >> > >> likely some locking missing. > >> >> > > > >> >> > > It's rather pci_get_drvdata() returning NULL (i.e. card is NULL, thus > >> >> > > card->private_data causes Oops). Could you check the patch like below > >> >> > > and see whether you get a kernel warning (but no Oops) or the problem > >> >> > > gets fixed by shifting the assignment of pci drvdata? > >> >> > [...] > >> >> > > >> >> > Good patching. Calling pci_set_drvdata later prevents the oops in HDA, > >> >> > though we see unexpected 0x0 responses in the response ring buffer > >> >> > [1], which we don't see when there's a >~1.5s delay between IGD and > >> >> > OFF. > >> >> > >> >> If the previous patch fixed, it means that the switching occurred > >> >> during the device was being probed. Maybe a better approach to > >> >> register the VGA switcheroo after the proper initialization. > >> >> > >> >> The patch below is a revised one. Please give it a try. > >> > > >> > Also, it's not clear which card spews the spurious response. > >> > Apply the patch below in addition. > >> [...] > >> > >> hda-intel: 0000:01:00.1: spurious response 0x0:0x0, last cmd=0x1f0004 > >> $ lspci -s :1:0.1 > >> 01:00.1 Audio device: NVIDIA Corporation Device 0e1b (rev ff) > >> > >> It's the NVIDIA device which presumably hasn't completed it's > >> transition to D3 at the time the OFF is executed. > > > > OK, then could you try the patch below on the top of previous two > > patches? > > The first IGD switcheroo command fails to switch to the integrated GPU: > > # cat /sys/kernel/debug/vgaswitcheroo/switch > 0:DIS:+:Pwr:0000:01:00.0 > 1:IGD: :Pwr:0000:00:02.0 > 2:DIS-Audio: :Pwr:0000:01:00.1 > # echo IGD >/sys/kernel/debug/vgaswitcheroo/switch > vga_switcheroo: client 1 refused switch > > I also instrumented snd_hda_lock_devices, but none of the failure > paths are being taken, which would leave inconsistent state, as the > return value isn't checked. Hm, right, the return value of snd_hda_lock_devices() isn't checked, but I don't understand how this results like above. Basically switching is protected by mutex in vga_switcheroo.c, so the whole operation in the client side should be serialized. In anyway, try the patch below cleanly, and see the spurious message error coming up at which timing. thanks, Takashi --- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6833835..4518b05 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -501,6 +501,7 @@ struct azx { /* VGA-switcheroo setup */ unsigned int use_vga_switcheroo:1; + unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ unsigned int disabled:1; /* disabled by VGA-switcher */ @@ -1597,6 +1598,8 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode int c, codecs, err; int max_slots; + printk(KERN_DEBUG "XXX %s: azx_codec_create entered\n", + pci_name(chip->pci)); memset(&bus_temp, 0, sizeof(bus_temp)); bus_temp.private_data = chip; bus_temp.modelname = model; @@ -1673,6 +1676,8 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode snd_printk(KERN_ERR SFX "no codecs initialized\n"); return -ENXIO; } + printk(KERN_DEBUG "XXX %s: azx_codec_create done\n", + pci_name(chip->pci)); return 0; } @@ -2615,6 +2620,8 @@ static void azx_vs_set_state(struct pci_dev *pci, return; disabled = (state == VGA_SWITCHEROO_OFF); + printk(KERN_DEBUG "XXX %s: chip->disabled=%d, disabled=%d\n", + pci_name(chip->pci), chip->disabled, disabled); if (chip->disabled == disabled) return; @@ -2638,14 +2645,26 @@ static void azx_vs_set_state(struct pci_dev *pci, disabled ? "Disabling" : "Enabling", pci_name(chip->pci)); if (disabled) { + printk(KERN_DEBUG "XXX %s: azx_suspend...\n", + pci_name(chip->pci)); azx_suspend(&pci->dev); + printk(KERN_DEBUG "XXX %s: azx_suspend done...\n", + pci_name(chip->pci)); chip->disabled = true; - snd_hda_lock_devices(chip->bus); + if (snd_hda_lock_devices(chip->bus)) + snd_printk(KERN_WARNING SFX + "Cannot lock devices!\n"); } else { snd_hda_unlock_devices(chip->bus); chip->disabled = false; + printk(KERN_DEBUG "XXX %s: azx_resume...\n", + pci_name(chip->pci)); azx_resume(&pci->dev); + printk(KERN_DEBUG "XXX %s: azx_resume done...\n", + pci_name(chip->pci)); } + printk(KERN_DEBUG "XXX %s: switching finished: disabled=%d\n", + pci_name(chip->pci), chip->disabled); } } @@ -2683,14 +2702,20 @@ static const struct vga_switcheroo_client_ops azx_vs_ops = { static int __devinit register_vga_switcheroo(struct azx *chip) { + int err; + if (!chip->use_vga_switcheroo) return 0; /* FIXME: currently only handling DIS controller * is there any machine with two switchable HDMI audio controllers? */ - return vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, + err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, VGA_SWITCHEROO_DIS, chip->bus != NULL); + if (err < 0) + return err; + chip->vga_switcheroo_registered = 1; + return 0; } #else #define init_vga_switcheroo(chip) /* NOP */ @@ -2712,7 +2737,8 @@ static int azx_free(struct azx *chip) if (use_vga_switcheroo(chip)) { if (chip->disabled && chip->bus) snd_hda_unlock_devices(chip->bus); - vga_switcheroo_unregister_client(chip->pci); + if (chip->vga_switcheroo_registered) + vga_switcheroo_unregister_client(chip->pci); } if (chip->initialized) { @@ -3062,14 +3088,6 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } ok: - err = register_vga_switcheroo(chip); - if (err < 0) { - snd_printk(KERN_ERR SFX - "Error registering VGA-switcheroo client\n"); - azx_free(chip); - return err; - } - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); @@ -3340,6 +3358,13 @@ static int __devinit azx_probe(struct pci_dev *pci, if (pci_dev_run_wake(pci)) pm_runtime_put_noidle(&pci->dev); + err = register_vga_switcheroo(chip); + if (err < 0) { + snd_printk(KERN_ERR SFX + "Error registering VGA-switcheroo client\n"); + goto out_free; + } + dev++; return 0;

1 0

[alsa-devel] [PATCH 1/4] ASoC: wm8770: Convert to direct regmap API usage
by Mark Brown 10 Oct '12

10 Oct '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8770.c | 100 +++++++++++++++++++++++++++++---------------- 1 file changed, 65 insertions(+), 35 deletions(-) diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index c7c0034..21ec563 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -17,6 +17,7 @@ #include <linux/of_device.h> #include <linux/pm.h> #include <linux/spi/spi.h> +#include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <sound/core.h> @@ -35,19 +36,52 @@ static const char *wm8770_supply_names[WM8770_NUM_SUPPLIES] = { "DVDD" }; -static const u16 wm8770_reg_defs[WM8770_CACHEREGNUM] = { - 0x7f, 0x7f, 0x7f, 0x7f, - 0x7f, 0x7f, 0x7f, 0x7f, - 0x7f, 0xff, 0xff, 0xff, - 0xff, 0xff, 0xff, 0xff, - 0xff, 0xff, 0, 0x90, 0, - 0, 0x22, 0x22, 0x3e, - 0xc, 0xc, 0x100, 0x189, - 0x189, 0x8770 +static const struct reg_default wm8770_reg_defaults[] = { + { 0, 0x7f }, + { 1, 0x7f }, + { 2, 0x7f }, + { 3, 0x7f }, + { 4, 0x7f }, + { 5, 0x7f }, + { 6, 0x7f }, + { 7, 0x7f }, + { 8, 0x7f }, + { 9, 0xff }, + { 10, 0xff }, + { 11, 0xff }, + { 12, 0xff }, + { 13, 0xff }, + { 14, 0xff }, + { 15, 0xff }, + { 16, 0xff }, + { 17, 0xff }, + { 18, 0 }, + { 19, 0x90 }, + { 20, 0 }, + { 21, 0 }, + { 22, 0x22 }, + { 23, 0x22 }, + { 24, 0x3e }, + { 25, 0xc }, + { 26, 0xc }, + { 27, 0x100 }, + { 28, 0x189 }, + { 29, 0x189 }, + { 30, 0x8770 }, }; +static bool wm8770_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8770_RESET: + return true; + default: + return false; + } +} + struct wm8770_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8770_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8770_NUM_SUPPLIES]; struct snd_soc_codec *codec; @@ -71,7 +105,7 @@ static int wm8770_regulator_event_##n(struct notifier_block *nb, \ struct wm8770_priv *wm8770 = container_of(nb, struct wm8770_priv, \ disable_nb[n]); \ if (event & REGULATOR_EVENT_DISABLE) { \ - wm8770->codec->cache_sync = 1; \ + regcache_mark_dirty(wm8770->regmap); \ } \ return 0; \ } @@ -466,24 +500,6 @@ static int wm8770_set_sysclk(struct snd_soc_dai *dai, return 0; } -static void wm8770_sync_cache(struct snd_soc_codec *codec) -{ - int i; - u16 *cache; - - if (!codec->cache_sync) - return; - - codec->cache_only = 0; - cache = codec->reg_cache; - for (i = 0; i < codec->driver->reg_cache_size; i++) { - if (i == WM8770_RESET || cache[i] == wm8770_reg_defs[i]) - continue; - snd_soc_write(codec, i, cache[i]); - } - codec->cache_sync = 0; -} - static int wm8770_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -507,7 +523,9 @@ static int wm8770_set_bias_level(struct snd_soc_codec *codec, ret); return ret; } - wm8770_sync_cache(codec); + + regcache_sync(wm8770->regmap); + /* global powerup */ snd_soc_write(codec, WM8770_PWDNCTRL, 0); } @@ -580,7 +598,7 @@ static int wm8770_probe(struct snd_soc_codec *codec) wm8770 = snd_soc_codec_get_drvdata(codec); wm8770->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8770->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -678,9 +696,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8770 = { .resume = wm8770_resume, .set_bias_level = wm8770_set_bias_level, .idle_bias_off = true, - .reg_cache_size = ARRAY_SIZE(wm8770_reg_defs), - .reg_word_size = sizeof (u16), - .reg_cache_default = wm8770_reg_defs }; static const struct of_device_id wm8770_of_match[] = { @@ -689,6 +704,18 @@ static const struct of_device_id wm8770_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8770_of_match); +static const struct regmap_config wm8770_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8770_RESET, + + .reg_defaults = wm8770_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8770_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8770_volatile_reg, +}; + static int __devinit wm8770_spi_probe(struct spi_device *spi) { struct wm8770_priv *wm8770; @@ -699,7 +726,10 @@ static int __devinit wm8770_spi_probe(struct spi_device *spi) if (!wm8770) return -ENOMEM; - wm8770->control_type = SND_SOC_SPI; + wm8770->regmap = devm_regmap_init_spi(spi, &wm8770_regmap); + if (IS_ERR(wm8770->regmap)) + return PTR_ERR(wm8770->regmap); + spi_set_drvdata(spi, wm8770); ret = snd_soc_register_codec(&spi->dev, -- 1.7.10.4

1 3

Re: [alsa-devel] [PATCH 0/2] Convert davinci ASoC to genalloc SRAM
by Daniel Mack 10 Oct '12

10 Oct '12

On 10.10.2012 11:59, Kumar, Anil wrote: > On Wed, Oct 10, 2012 at 12:34:47, Daniel Mack wrote: >> On 10.10.2012 08:54, Kumar, Anil wrote: >>> On Thu, Oct 04, 2012 at 15:45:43, Porter, Matt wrote: >>>> This series converts davinci ASoC to use genalloc and enables >>>> that support on DA850. It applies on top of the uio_pruss/genalloc >>>> series [1] which allows DaVinci to provide a gen_pool via pdata >>>> for driver use. >>>> >>>> I've tested this on the AM180x EVM. Note that prior to this, >>>> the SRAM paths in the driver were completely unused. I've only >>>> enabled ping-pong buffering on the platform I can test as it's >>>> best to allow those with DM644x and similar platforms to set the >>>> playback/capture sram size to something that's known to work >>>> for them. >>>> >>>> [1] http://www.spinics.net/lists/arm-kernel/msg198854.html >>> >>> Tested on DA850 EVM after applying patches from [1] above. >>> Tested on Linux-torvalds 3.6-rc6 master branch. >>> Tested ASoC module (aplay & arecord for various frequencies) >>> >>> Acked-by: Kumar, Anil <anilkumar.v(a)ti.com> >> >> As you're on it, could you also quickly merge this branch >> >> git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git >> topic/davinci >> >> And see if my recent patches for more McASP flexibility cause any >> trouble on your hardware? >> >> >> Daniel >> >> > > I have merged it with 3.6-rc6 and I have got error at compile time. > I have fixed this locally with the following patch [1]. > Kindly confirm if the patch is correct? I did the same thing locally, but I considered this a merge conflict. > With that > Tested on DA850 EVM > Tested on Linux-torvalds 3.6-rc6 master branch merged + [1] > Tested ASoC module (aplay & arecord for various frequencies) > > Tested-By: Kumar, Anil <anilkumar.v(a)ti.com> Ok, that's good to hear. Thanks a lot :) Daniel > diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h > index 09da4e5..9c02a45 100644 > --- a/include/linux/dmaengine.h > +++ b/include/linux/dmaengine.h > @@ -591,7 +591,7 @@ struct dma_device { > struct dma_async_tx_descriptor *(*device_prep_dma_cyclic)( > struct dma_chan *chan, dma_addr_t buf_addr, size_t buf_len, > size_t period_len, enum dma_transfer_direction direction, > - unsigned long flags, void *context); > + void *context); > struct dma_async_tx_descriptor *(*device_prep_interleaved_dma)( > struct dma_chan *chan, struct dma_interleaved_template *xt, > unsigned long flags); > @@ -656,7 +656,7 @@ static inline struct dma_async_tx_descriptor *dmaengine_prep_dma_cyclic( > size_t period_len, enum dma_transfer_direction dir) > { > return chan->device->device_prep_dma_cyclic(chan, buf_addr, buf_len, > - period_len, dir, flags, NULL); > + period_len, dir, NULL); > } >

1 0

[alsa-devel] [PATCH] ALSA: hda - Add missing hda_gen_spec to struct via_spec
by Takashi Iwai 10 Oct '12

10 Oct '12

The commit [4b527b65 ALSA: hda - limit internal mic boost for Asus X202E] introduced the use of auto-parser code, but it forgot to add struct hda_gen_spec at the head of codec->spec which the auto-parser assumes silently. Without this record, it may result in memory corruption. This patch adds the missing piece. Cc: <stable(a)vger.kernel.org> [v3.5+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- David, I queued this fix now. I'm going to send a pull request tomorrow or on Friday. sound/pci/hda/patch_via.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index eae466a..72a2f60 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -118,6 +118,8 @@ enum { }; struct via_spec { + struct hda_gen_spec gen; + /* codec parameterization */ const struct snd_kcontrol_new *mixers[6]; unsigned int num_mixers; @@ -246,6 +248,7 @@ static struct via_spec * via_new_spec(struct hda_codec *codec) /* VT1708BCE & VT1708S are almost same */ if (spec->codec_type == VT1708BCE) spec->codec_type = VT1708S; + snd_hda_gen_init(&spec->gen); return spec; } @@ -1627,6 +1630,7 @@ static void via_free(struct hda_codec *codec) vt1708_stop_hp_work(spec); kfree(spec->bind_cap_vol); kfree(spec->bind_cap_sw); + snd_hda_gen_free(&spec->gen); kfree(spec); } -- 1.7.12.2

2 2

Re: [alsa-devel] [PATCH 0/2] Convert davinci ASoC to genalloc SRAM
by Daniel Mack 10 Oct '12

10 Oct '12

On 10.10.2012 08:54, Kumar, Anil wrote: > On Thu, Oct 04, 2012 at 15:45:43, Porter, Matt wrote: >> This series converts davinci ASoC to use genalloc and enables >> that support on DA850. It applies on top of the uio_pruss/genalloc >> series [1] which allows DaVinci to provide a gen_pool via pdata >> for driver use. >> >> I've tested this on the AM180x EVM. Note that prior to this, >> the SRAM paths in the driver were completely unused. I've only >> enabled ping-pong buffering on the platform I can test as it's >> best to allow those with DM644x and similar platforms to set the >> playback/capture sram size to something that's known to work >> for them. >> >> [1] http://www.spinics.net/lists/arm-kernel/msg198854.html > > Tested on DA850 EVM after applying patches from [1] above. > Tested on Linux-torvalds 3.6-rc6 master branch. > Tested ASoC module (aplay & arecord for various frequencies) > > Acked-by: Kumar, Anil <anilkumar.v(a)ti.com> As you're on it, could you also quickly merge this branch git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/davinci And see if my recent patches for more McASP flexibility cause any trouble on your hardware? Daniel

1 0

Re: [alsa-devel] sound/pci/pcxhr/pcxhr.c:916:2: warning: 'flush_work_sync' is deprecated (declared at include/linux/workqueue.h:448)
by Takashi Iwai 10 Oct '12

10 Oct '12

At Wed, 10 Oct 2012 07:36:43 +0800, Fengguang Wu wrote: > > Hi Takashi, > > FYI, there *will* be new compile warnings in > > 672a112 ALSA: pcxhr - Kill tasklets > > sound/pci/pcxhr/pcxhr.c: In function 'pcxhr_prepare': > sound/pci/pcxhr/pcxhr.c:916:2: warning: 'flush_work_sync' is deprecated (declared at include/linux/workqueue.h:448) [-Wdeprecated-declarations] > sound/pci/pcxhr/pcxhr.c: In function 'pcxhr_hw_free': > sound/pci/pcxhr/pcxhr.c:978:2: warning: 'flush_work_sync' is deprecated (declared at include/linux/workqueue.h:448) [-Wdeprecated-declarations] Thanks, fixed now. Takashi > > vim +916 sound/pci/pcxhr/pcxhr.c > > e12229b4 Markus Bollinger 2005-12-06 901 > e12229b4 Markus Bollinger 2005-12-06 902 /* > e12229b4 Markus Bollinger 2005-12-06 903 * prepare callback for all pcms > e12229b4 Markus Bollinger 2005-12-06 904 */ > e12229b4 Markus Bollinger 2005-12-06 905 static int pcxhr_prepare(struct snd_pcm_substream *subs) > e12229b4 Markus Bollinger 2005-12-06 906 { > e12229b4 Markus Bollinger 2005-12-06 907 struct snd_pcxhr *chip = snd_pcm_substream_chip(subs); > e12229b4 Markus Bollinger 2005-12-06 908 struct pcxhr_mgr *mgr = chip->mgr; > e12229b4 Markus Bollinger 2005-12-06 909 int err = 0; > e12229b4 Markus Bollinger 2005-12-06 910 > e12229b4 Markus Bollinger 2005-12-06 911 snd_printdd("pcxhr_prepare : period_size(%lx) periods(%x) buffer_size(%lx)\n", > e12229b4 Markus Bollinger 2005-12-06 912 subs->runtime->period_size, subs->runtime->periods, > e12229b4 Markus Bollinger 2005-12-06 913 subs->runtime->buffer_size); > e12229b4 Markus Bollinger 2005-12-06 914 > 672a1128 Takashi Iwai 2011-06-14 915 cancel_work_sync(&mgr->timer_work); > 672a1128 Takashi Iwai 2011-06-14 @916 flush_work_sync(&mgr->trigger_work); > 672a1128 Takashi Iwai 2011-06-14 917 > 62932df8 Ingo Molnar 2006-01-16 918 mutex_lock(&mgr->setup_mutex); > e12229b4 Markus Bollinger 2005-12-06 919 > e12229b4 Markus Bollinger 2005-12-06 920 do { > e12229b4 Markus Bollinger 2005-12-06 921 /* only the first stream can choose the sample rate */ > e12229b4 Markus Bollinger 2005-12-06 922 /* set the clock only once (first stream) */ > 8937fd88 Takashi Iwai 2006-01-10 923 if (mgr->sample_rate != subs->runtime->rate) { > e12229b4 Markus Bollinger 2005-12-06 924 err = pcxhr_set_clock(mgr, subs->runtime->rate); > > --- > 0-DAY kernel build testing backend Open Source Technology Center > Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [PATCH v2 1/3] ALSA: core: keep track of boundary wrap-around
by Pierre-Louis Bossart 10 Oct '12

10 Oct '12

Keep track of boundary crossing when hw_ptr exceeds boundary limit and wraps-around. This will help keep track of total number of frames played/received at the kernel level Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 26 +++++++++++++++++++++----- 2 files changed, 22 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index cdca2ab..ba13e0b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -281,6 +281,7 @@ struct snd_pcm_runtime { unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ unsigned long hw_ptr_buffer_jiffies; /* buffer time in jiffies */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ + u64 hw_ptr_wrap; /* offset for hw_ptr due to boundary wrap-around */ /* -- HW params -- */ snd_pcm_access_t access; /* access mode */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7ae6719..ec996ca 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -315,6 +315,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, unsigned long jdelta; unsigned long curr_jiffies; struct timespec curr_tstamp; + int crossed_boundary = 0; old_hw_ptr = runtime->status->hw_ptr; @@ -359,8 +360,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, hdelta = curr_jiffies - runtime->hw_ptr_jiffies; if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) + if (hw_base >= runtime->boundary) { hw_base = 0; + crossed_boundary++; + } new_hw_ptr = hw_base + pos; goto __delta; } @@ -370,8 +373,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* pointer crosses the end of the ring buffer */ if (new_hw_ptr < old_hw_ptr) { hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) + if (hw_base >= runtime->boundary) { hw_base = 0; + crossed_boundary++; + } new_hw_ptr = hw_base + pos; } __delta: @@ -409,8 +414,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, while (hdelta > xrun_threshold) { delta += runtime->buffer_size; hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) + if (hw_base >= runtime->boundary) { hw_base = 0; + crossed_boundary++; + } new_hw_ptr = hw_base + pos; hdelta -= runtime->hw_ptr_buffer_jiffies; } @@ -455,8 +462,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* the delta value is small or zero in most cases */ while (delta > 0) { new_hw_ptr += runtime->period_size; - if (new_hw_ptr >= runtime->boundary) + if (new_hw_ptr >= runtime->boundary) { new_hw_ptr -= runtime->boundary; + crossed_boundary--; + } delta--; } /* align hw_base to buffer_size */ @@ -506,6 +515,11 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = curr_jiffies; + if (crossed_boundary) { + BUG_ON(crossed_boundary > 1); + BUG_ON(crossed_boundary < 0); + runtime->hw_ptr_wrap += runtime->boundary; + } if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) runtime->status->tstamp = curr_tstamp; @@ -1660,8 +1674,10 @@ static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream, if (snd_pcm_running(substream) && snd_pcm_update_hw_ptr(substream) >= 0) runtime->status->hw_ptr %= runtime->buffer_size; - else + else { runtime->status->hw_ptr = 0; + runtime->hw_ptr_wrap = 0; + } snd_pcm_stream_unlock_irqrestore(substream, flags); return 0; } -- 1.7.9.5

2 7

[alsa-devel] [PATCH] ALSA: ASoC: McASP: treat SNDRV_PCM_FORMAT_U24_* as 32 bit words
by Daniel Mack 10 Oct '12

10 Oct '12

Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: Mike Looijmans <mike.looijmans(a)topic.nl> Cc: Mike Looijmans <mike.looijmans(a)topic.nl> --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 501f8b4..82877a9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -868,12 +868,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_S24_3LE: - case SNDRV_PCM_FORMAT_U24_LE: - case SNDRV_PCM_FORMAT_S24_LE: dma_params->data_type = 3; word_length = DAVINCI_AUDIO_WORD_24; break; + case SNDRV_PCM_FORMAT_U24_LE: + case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; -- 1.7.11.4

2 1

[alsa-devel] [PATCH] ASoC: dmaengine: Correct Makefile when sound is built as module
by Peter Ujfalusi 10 Oct '12

10 Oct '12

soc-dmaengine-pcm library need to be part of the snd-soc-core in order to be able to compile ASoC as modules when dmaengine is enabled on the platform. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/Makefile | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/Makefile b/sound/soc/Makefile index bcbf1d0..99f32f7 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,8 +1,9 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o -snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o -obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o +ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),) +snd-soc-core-objs += soc-dmaengine-pcm.o +endif obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ -- 1.7.12

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Re: [alsa-devel] [GIT PULL] Disintegrate UAPI for sound
by Takashi Iwai 09 Oct '12

09 Oct '12

At Tue, 09 Oct 2012 14:58:41 +0100, David Howells wrote: > > Takashi Iwai <tiwai(a)suse.de> wrote: > > > > Can you merge the following branch into the sound tree please. > > > > The changes look OK, but is it targeted for 3.7 or 3.8 in general? > > I assumed the latter. > > Either will do. There are no dependencies on the branch I gave you. OK, then I'm going to take it once when 3.7-rc1 is released. thanks, Takashi

1 0

[alsa-devel] Complex hw buffer with MMAP_COMPLEX?
by Adrian Knoth 09 Oct '12

09 Oct '12

Hi! I'm about to start working on the driver for the new RME MADI FX card with its 194 input and 196 output channels. I've noticed that the device has a rather uncommon hw buffer layout. Let's stick to the input side for the sake of simplicity: 2ch AES, interleaved 8ch MADI 001-008, interleaved 8ch MADI 009-015, interleaved [..] 8ch MADI 184-192, interleaved On the output side, there is another stereo pair for headphones. That is, we're dealing with 25 (or 26 for output) interleaved streams with two or eight channels each. How could we represent such a layout in ALSA, that is, how are we talking to userspace? Would one create subdevices and rejoin them via plughw? Or use MMAP_COMPLEX and then let every application deal with it on its own? It's certainly not a problem to special-case the MADI FX in jackd1 and jackd2, but OTOH, this would require other ALSA apps to do the same (I'm thinking of proprietary custom code). I wonder if we could afford to rearrange the buffers in the driver and expose a single non-interleaved buffer to userspace. WDYT? TIA

3 3

Re: [alsa-devel] [GIT PULL] Disintegrate UAPI for sound
by Takashi Iwai 09 Oct '12

09 Oct '12

At Tue, 09 Oct 2012 14:30:42 +0100, David Howells wrote: > > Can you merge the following branch into the sound tree please. The changes look OK, but is it targeted for 3.7 or 3.8 in general? I assumed the latter. thanks, Takashi > > This is to complete part of the Userspace API (UAPI) disintegration for which > the preparatory patches were pulled recently. After these patches, userspace > headers will be segregated into: > > include/uapi/linux/.../foo.h > > for the userspace interface stuff, and: > > include/linux/.../foo.h > > for the strictly kernel internal stuff. > > --- > The following changes since commit 9e2d8656f5e8aa214e66b462680cf86b210b74a8: > > Merge branch 'akpm' (Andrew's patch-bomb) (2012-10-09 16:23:15 +0900) > > are available in the git repository at: > > > git://git.infradead.org/users/dhowells/linux-headers.git tags/disintegrate-sound-20121009 > > for you to fetch changes up to 674e95ca446369e9ca40e4e92b214a616a52bcd6: > > UAPI: (Scripted) Disintegrate include/sound (2012-10-09 09:49:13 +0100) > > ---------------------------------------------------------------- > UAPI Disintegration 2012-10-09 > > ---------------------------------------------------------------- > David Howells (1): > UAPI: (Scripted) Disintegrate include/sound > > include/sound/Kbuild | 10 - > include/sound/asequencer.h | 594 +---------------- > include/sound/asound.h | 935 +-------------------------- > include/sound/emu10k1.h | 355 +---------- > include/sound/sb16_csp.h | 104 +-- > include/uapi/sound/Kbuild | 10 + > include/uapi/sound/asequencer.h | 614 ++++++++++++++++++ > include/uapi/sound/asound.h | 957 ++++++++++++++++++++++++++++ > include/{ => uapi}/sound/asound_fm.h | 0 > include/{ => uapi}/sound/compress_offload.h | 0 > include/{ => uapi}/sound/compress_params.h | 0 > include/uapi/sound/emu10k1.h | 373 +++++++++++ > include/{ => uapi}/sound/hdsp.h | 0 > include/{ => uapi}/sound/hdspm.h | 0 > include/uapi/sound/sb16_csp.h | 122 ++++ > include/{ => uapi}/sound/sfnt_info.h | 0 > 16 files changed, 2084 insertions(+), 1990 deletions(-) > create mode 100644 include/uapi/sound/asequencer.h > create mode 100644 include/uapi/sound/asound.h > rename include/{ => uapi}/sound/asound_fm.h (100%) > rename include/{ => uapi}/sound/compress_offload.h (100%) > rename include/{ => uapi}/sound/compress_params.h (100%) > create mode 100644 include/uapi/sound/emu10k1.h > rename include/{ => uapi}/sound/hdsp.h (100%) > rename include/{ => uapi}/sound/hdspm.h (100%) > create mode 100644 include/uapi/sound/sb16_csp.h > rename include/{ => uapi}/sound/sfnt_info.h (100%) > . >

1 0

[alsa-devel] [PATCH v2] ALSA: ASoC: McASP: add support for 24 bit samples
by Mike Looijmans 09 Oct '12

09 Oct '12

Sorry for the lack of quoting, but I onle get the digest. These are wrong: + case SNDRV_PCM_FORMAT_U24_LE: + case SNDRV_PCM_FORMAT_S24_LE: These pack a 24-bit sample value in a 32-bit word. The codec will send 32 bits to the McASP, and you should transfer 32 bits to the user, not just 24. Hence, SNDRV_PCM_FORMAT_S24_LE must be treated just like SNDRV_PCM_FORMAT_S32_LE. I've tested that on a DA850-alike board with several TLV320AIC3256 codecs, treating them as 3-byte samples will reasult in invalid data. Met vriendelijke groet / kind regards, Mike Looijmans TOPIC Embedded Systems Eindhovenseweg 32-C, NL-5683 KH Best Postbus 440, NL-5680 AK Best Telefoon: (+31) - (0)499 - 33.69.79 Telefax: (+31) - (0)499 - 33.69.70 E-mail: mike.looijmans(a)topic.nl Website: www.topic.nl Dit e-mail bericht en de eventueel daarbij behorende bijlagen zijn uitsluitend bestemd voor de geadresseerde, zoals die blijkt uit het e-mail bericht en/of de bijlagen. Er kunnen gegevens met betrekking tot een derde instaan. Indien u als niet-geadresseerde dit bericht en de bijlagen ontvangt, terwijl u niet bevoegd of gemachtigd bent om dit bericht namens de geadresseerde te ontvangen, wordt u verzocht de afzender hierover direct te informeren en het e-mail bericht met de bijlagen te vernietigen. Ieder gebruik van de inhoud van het e-mail bericht, waaronder de daarbij behorende bijlagen, door een ander dan de geadresseerde is onrechtmatig jegens ons dan wel de eventueel in het e-mail bericht of de bijlagen voorkomende andere personen. TOPIC Embedded Systems is niet aansprakelijk voor enigerlei schade voortvloeiend uit het gebruik en/of acceptatie van dit e-mail bericht of de daarbij behorende bijlagen. The contents of this message, as well as any enclosures, are addressed personally to, and thus solely intended for the addressee. They may contain information regarding a third party. A recipient who is neither the addressee, nor empowered to receive this message on behalf of the addressee, is kindly requested to immediately inform the sender of receipt, and to destroy the message and the enclosures. Any use of the contents of this message and/or the enclosures by any other person than the addressee or person who is empowered to receive this message, is illegal towards the sender and/or the aforementioned third party. TOPIC Embedded Systems is not liable for any damage as a result of the use and/or acceptance of this message and as well as any enclosures.

4 7

Re: [alsa-devel] [3.6-rc7] switcheroo race with Intel HDA...
by Takashi Iwai 09 Oct '12

09 Oct '12

At Tue, 9 Oct 2012 19:23:56 +0800, Daniel J Blueman wrote: > > On 9 October 2012 18:07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 09 Oct 2012 12:04:08 +0200, > > Takashi Iwai wrote: > >> > >> At Tue, 9 Oct 2012 00:34:09 +0800, > >> Daniel J Blueman wrote: > >> > > >> > On 8 October 2012 20:58, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > > At Tue, 25 Sep 2012 13:20:05 +0800, > >> > > Daniel J Blueman wrote: > >> > >> On my Macbook with a discrete Nvidia GPU, there is a race between > >> > >> selecting the integrated GPU and putting the discrete GPU into D3 [1], > >> > >> reliably causing a kernel oops [2]. > >> > >> > >> > >> Introducing a delay of ~1s between the calls prevents this. When the > >> > >> second 'OFF' write path executes, it looks like struct azx at > >> > >> card->private_data hasn't yet been allocated yet [3], so there is > >> > >> likely some locking missing. > >> > > > >> > > It's rather pci_get_drvdata() returning NULL (i.e. card is NULL, thus > >> > > card->private_data causes Oops). Could you check the patch like below > >> > > and see whether you get a kernel warning (but no Oops) or the problem > >> > > gets fixed by shifting the assignment of pci drvdata? > >> > [...] > >> > > >> > Good patching. Calling pci_set_drvdata later prevents the oops in HDA, > >> > though we see unexpected 0x0 responses in the response ring buffer > >> > [1], which we don't see when there's a >~1.5s delay between IGD and > >> > OFF. > >> > >> If the previous patch fixed, it means that the switching occurred > >> during the device was being probed. Maybe a better approach to > >> register the VGA switcheroo after the proper initialization. > >> > >> The patch below is a revised one. Please give it a try. > > > > Also, it's not clear which card spews the spurious response. > > Apply the patch below in addition. > [...] > > hda-intel: 0000:01:00.1: spurious response 0x0:0x0, last cmd=0x1f0004 > $ lspci -s :1:0.1 > 01:00.1 Audio device: NVIDIA Corporation Device 0e1b (rev ff) > > It's the NVIDIA device which presumably hasn't completed it's > transition to D3 at the time the OFF is executed. OK, then could you try the patch below on the top of previous two patches? Takashi --- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f09ff6c..676c64d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2639,13 +2640,13 @@ static void azx_vs_set_state(struct pci_dev *pci, disabled ? "Disabling" : "Enabling", pci_name(chip->pci)); if (disabled) { + snd_hda_lock_devices(chip->bus); azx_suspend(&pci->dev); chip->disabled = true; - snd_hda_lock_devices(chip->bus); } else { - snd_hda_unlock_devices(chip->bus); chip->disabled = false; azx_resume(&pci->dev); + snd_hda_unlock_devices(chip->bus); } } }

1 0

[alsa-devel] [PATCH] ALSA: hda - remove "Mic Jack Mode" for headset jacks (Latitude Exx30)
by David Henningsson 09 Oct '12

09 Oct '12

Dell Latitude 5x30 and 6x30 series of machines all have a single 4-pin headset jack. Enabling line in mode for such jack is very confusing (you would only get mono input, and would have to use non-standard adapters), so remove the option by default. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index bb6c50e..f60b353 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -104,6 +104,7 @@ enum { STAC_92HD83XXX_HP_LED, STAC_92HD83XXX_HP_INV_LED, STAC_92HD83XXX_HP_MIC_LED, + STAC_92HD83XXX_HEADSET_JACK, STAC_92HD83XXX_MODELS }; @@ -204,6 +205,7 @@ struct sigmatel_spec { unsigned int check_volume_offset:1; unsigned int auto_mic:1; unsigned int linear_tone_beep:1; + unsigned int headset_jack:1; /* 4-pin headset jack (hp + mono mic) */ /* gpio lines */ unsigned int eapd_mask; @@ -1684,6 +1686,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_HP_LED] = "hp-led", [STAC_92HD83XXX_HP_INV_LED] = "hp-inv-led", [STAC_92HD83XXX_HP_MIC_LED] = "hp-mic-led", + [STAC_92HD83XXX_HEADSET_JACK] = "headset-jack", }; static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1694,6 +1697,24 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD83XXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba, "unknown Dell", STAC_DELL_S14), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0532, + "Dell Latitude E6230", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0533, + "Dell Latitude E6330", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0534, + "Dell Latitude E6430", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0535, + "Dell Latitude E6530", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x053c, + "Dell Latitude E5430", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x053d, + "Dell Latitude E5530", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0549, + "Dell Latitude E5430", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x057d, + "Dell Latitude E6430s", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0584, + "Dell Latitude E6430U", STAC_92HD83XXX_HEADSET_JACK), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x1028, "Dell Vostro 3500", STAC_DELL_VOSTRO_3500), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1656, @@ -2855,6 +2876,9 @@ static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, char name[22]; if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) { + if (spec->headset_jack && snd_hda_get_input_pin_attr(def_conf) + != INPUT_PIN_ATTR_DOCK) + return 0; if (snd_hda_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD && nid == spec->line_switch) control = STAC_CTL_WIDGET_IO_SWITCH; @@ -5636,6 +5660,9 @@ again: case STAC_92HD83XXX_HP_MIC_LED: spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ break; + case STAC_92HD83XXX_HEADSET_JACK: + spec->headset_jack = 1; + break; } if (find_mute_led_cfg(codec, default_polarity)) -- 1.7.9.5

2 1

Re: [alsa-devel] [3.6-rc7] switcheroo race with Intel HDA...
by Takashi Iwai 09 Oct '12

09 Oct '12

At Tue, 9 Oct 2012 00:34:09 +0800, Daniel J Blueman wrote: > > On 8 October 2012 20:58, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 25 Sep 2012 13:20:05 +0800, > > Daniel J Blueman wrote: > >> On my Macbook with a discrete Nvidia GPU, there is a race between > >> selecting the integrated GPU and putting the discrete GPU into D3 [1], > >> reliably causing a kernel oops [2]. > >> > >> Introducing a delay of ~1s between the calls prevents this. When the > >> second 'OFF' write path executes, it looks like struct azx at > >> card->private_data hasn't yet been allocated yet [3], so there is > >> likely some locking missing. > > > > It's rather pci_get_drvdata() returning NULL (i.e. card is NULL, thus > > card->private_data causes Oops). Could you check the patch like below > > and see whether you get a kernel warning (but no Oops) or the problem > > gets fixed by shifting the assignment of pci drvdata? > [...] > > Good patching. Calling pci_set_drvdata later prevents the oops in HDA, > though we see unexpected 0x0 responses in the response ring buffer > [1], which we don't see when there's a >~1.5s delay between IGD and > OFF. If the previous patch fixed, it means that the switching occurred during the device was being probed. Maybe a better approach to register the VGA switcheroo after the proper initialization. The patch below is a revised one. Please give it a try. Takashi --- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f09ff6c..dcbfd29 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -501,6 +501,7 @@ struct azx { /* VGA-switcheroo setup */ unsigned int use_vga_switcheroo:1; + unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ unsigned int disabled:1; /* disabled by VGA-switcher */ @@ -2684,14 +2685,20 @@ static const struct vga_switcheroo_client_ops azx_vs_ops = { static int __devinit register_vga_switcheroo(struct azx *chip) { + int err; + if (!chip->use_vga_switcheroo) return 0; /* FIXME: currently only handling DIS controller * is there any machine with two switchable HDMI audio controllers? */ - return vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, + err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, VGA_SWITCHEROO_DIS, chip->bus != NULL); + if (err < 0) + return err; + chip->vga_switcheroo_registered = 1; + return 0; } #else #define init_vga_switcheroo(chip) /* NOP */ @@ -2713,7 +2720,8 @@ static int azx_free(struct azx *chip) if (use_vga_switcheroo(chip)) { if (chip->disabled && chip->bus) snd_hda_unlock_devices(chip->bus); - vga_switcheroo_unregister_client(chip->pci); + if (chip->vga_switcheroo_registered) + vga_switcheroo_unregister_client(chip->pci); } if (chip->initialized) { @@ -3067,14 +3075,6 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } ok: - err = register_vga_switcheroo(chip); - if (err < 0) { - snd_printk(KERN_ERR SFX - "Error registering VGA-switcheroo client\n"); - azx_free(chip); - return err; - } - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); @@ -3345,6 +3345,13 @@ static int __devinit azx_probe(struct pci_dev *pci, if (pci_dev_run_wake(pci)) pm_runtime_put_noidle(&pci->dev); + err = register_vga_switcheroo(chip); + if (err < 0) { + snd_printk(KERN_ERR SFX + "Error registering VGA-switcheroo client\n"); + goto out_free; + } + dev++; return 0;

1 1

[alsa-devel] [PATCH v2] ALSA: ASoC: McASP: add support for 24 bit samples
by Daniel Mack 09 Oct '12

09 Oct '12

Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> Cc: Matt Porter <mporter(a)ti.com> --- v2: also enable 24bit in the DAI sound/soc/davinci/davinci-mcasp.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1252bad..76728be 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -839,6 +839,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, word_length = DAVINCI_AUDIO_WORD_16; break; + case SNDRV_PCM_FORMAT_U24_3LE: + case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_U24_LE: + case SNDRV_PCM_FORMAT_S24_LE: + dma_params->data_type = 3; + word_length = DAVINCI_AUDIO_WORD_24; + break; + case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; @@ -917,6 +925,10 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { SNDRV_PCM_FMTBIT_U8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | \ SNDRV_PCM_FMTBIT_U32_LE) -- 1.7.11.4

2 1

[alsa-devel] [PATCH 0/5] McASP changes for more flexibility
by Daniel Mack 09 Oct '12

09 Oct '12

Hi, I've been working on the McASP driver lately to support a board on which an AM33xx SoC is connected to a CS4271 via I2S. The CPU is clock master for both LRCLK and BITCLK, but receives its master clock externally. Some changes were necessary in order to get this playing, as the driver seems to have been used with DSP_B formats only. I couldn't the pateches on any Davinci platform, so I rely on other people to test and see if they break anything. Thanks, Daniel Daniel Mack (5): ALSA: ASoC: enable 192KHz support for McASP ALSA: ASoC: McASP: add support for clock dividers ALSA: ASoC: McASP: add support for 24 bit samples ALSA: ASoC: McASP: make AHCLK direction configurable ALSA: ASoC: McASP: set format parameters in dependence of the DAI format sound/soc/davinci/davinci-evm.c | 5 +++ sound/soc/davinci/davinci-mcasp.c | 87 ++++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-mcasp.h | 2 +- sound/soc/davinci/davinci-pcm.c | 16 ++----- 4 files changed, 86 insertions(+), 24 deletions(-) -- 1.7.11.4

2 10

[alsa-devel] [PATCH v2 1/2] ARM: imx: clk: Split SSI clock into 'ipg' and 'per'
by Fabio Estevam 09 Oct '12

09 Oct '12

Currently imx ssi driver has been using only the ipg clock as only slave mode is supported. For mx27 it is necessay to provide both 'ipg' and 'per' clocks in order to get ssi functional. This issue was found on mx27 by unselecting the mmc driver from the kernel ,which resulted on a system that could not play audio. When the mmc driver is selected the required 'per2' clock is provided and it allows the ssi to work. Add the required 'per' clock to the mx27 ssi and a dummy one for the other SoCs. Reported-by: Gaëtan Carlier <gcembed(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Drop per4 from the camera clok, which belonged to a different patch arch/arm/mach-imx/clk-imx21.c | 11 +++++++---- arch/arm/mach-imx/clk-imx25.c | 6 ++++-- arch/arm/mach-imx/clk-imx27.c | 8 ++++---- arch/arm/mach-imx/clk-imx31.c | 11 +++++++---- arch/arm/mach-imx/clk-imx35.c | 10 +++++++--- arch/arm/mach-imx/clk-imx51-imx53.c | 19 +++++++++++++------ 6 files changed, 42 insertions(+), 23 deletions(-) diff --git a/arch/arm/mach-imx/clk-imx21.c b/arch/arm/mach-imx/clk-imx21.c index cf65148..2ed49c8 100644 --- a/arch/arm/mach-imx/clk-imx21.c +++ b/arch/arm/mach-imx/clk-imx21.c @@ -51,8 +51,8 @@ static const char *mpll_sel_clks[] = { "fpm", "ckih", }; static const char *spll_sel_clks[] = { "fpm", "ckih", }; enum imx21_clks { - ckil, ckih, fpm, mpll_sel, spll_sel, mpll, spll, fclk, hclk, ipg, per1, - per2, per3, per4, uart1_ipg_gate, uart2_ipg_gate, uart3_ipg_gate, + dummy, ckil, ckih, fpm, mpll_sel, spll_sel, mpll, spll, fclk, hclk, ipg, + per1, per2, per3, per4, uart1_ipg_gate, uart2_ipg_gate, uart3_ipg_gate, uart4_ipg_gate, gpt1_ipg_gate, gpt2_ipg_gate, gpt3_ipg_gate, pwm_ipg_gate, sdhc1_ipg_gate, sdhc2_ipg_gate, lcdc_ipg_gate, lcdc_hclk_gate, cspi3_ipg_gate, cspi2_ipg_gate, cspi1_ipg_gate, @@ -72,6 +72,7 @@ int __init mx21_clocks_init(unsigned long lref, unsigned long href) { int i; + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckil] = imx_clk_fixed("ckil", lref); clk[ckih] = imx_clk_fixed("ckih", href); clk[fpm] = imx_clk_fixed_factor("fpm", "ckil", 512, 1); @@ -174,8 +175,10 @@ int __init mx21_clocks_init(unsigned long lref, unsigned long href) clk_register_clkdev(clk[gpio_gate], "gpio", NULL); clk_register_clkdev(clk[rtc_gate], "rtc", NULL); clk_register_clkdev(clk[csi_hclk_gate], "csi", NULL); - clk_register_clkdev(clk[ssi1_gate], "ssi1", NULL); - clk_register_clkdev(clk[ssi2_gate], "ssi2", NULL); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[sdhc1_ipg_gate], "sdhc1", NULL); clk_register_clkdev(clk[sdhc2_ipg_gate], "sdhc2", NULL); diff --git a/arch/arm/mach-imx/clk-imx25.c b/arch/arm/mach-imx/clk-imx25.c index d20d479..c81864d 100644 --- a/arch/arm/mach-imx/clk-imx25.c +++ b/arch/arm/mach-imx/clk-imx25.c @@ -222,8 +222,10 @@ int __init mx25_clocks_init(void) clk_register_clkdev(clk[lcdc_ipg], "ipg", "imx-fb.0"); clk_register_clkdev(clk[lcdc_ahb], "ahb", "imx-fb.0"); clk_register_clkdev(clk[wdt_ipg], NULL, "imx2-wdt.0"); - clk_register_clkdev(clk[ssi1_ipg], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_ipg], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[esdhc1_ipg_per], "per", "sdhci-esdhc-imx25.0"); clk_register_clkdev(clk[esdhc1_ipg], "ipg", "sdhci-esdhc-imx25.0"); clk_register_clkdev(clk[esdhc1_ahb], "ahb", "sdhci-esdhc-imx25.0"); diff --git a/arch/arm/mach-imx/clk-imx27.c b/arch/arm/mach-imx/clk-imx27.c index 3b6b640..8cbb6b5 100644 --- a/arch/arm/mach-imx/clk-imx27.c +++ b/arch/arm/mach-imx/clk-imx27.c @@ -236,8 +236,10 @@ int __init mx27_clocks_init(unsigned long fref) clk_register_clkdev(clk[usb_div], "per", "mxc-ehci.2"); clk_register_clkdev(clk[usb_ipg_gate], "ipg", "mxc-ehci.2"); clk_register_clkdev(clk[usb_ahb_gate], "ahb", "mxc-ehci.2"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[per2_gate], "per", "imx-ssi.0"); + clk_register_clkdev(clk[per2_gate], "per", "imx-ssi.1"); clk_register_clkdev(clk[nfc_baud_gate], NULL, "mxc_nand.0"); clk_register_clkdev(clk[vpu_baud_gate], "per", "coda-imx27.0"); clk_register_clkdev(clk[vpu_ahb_gate], "ahb", "coda-imx27.0"); @@ -262,8 +264,6 @@ int __init mx27_clocks_init(unsigned long fref) clk_register_clkdev(clk[scc_ipg_gate], "scc", NULL); clk_register_clkdev(clk[cpu_div], "cpu", NULL); clk_register_clkdev(clk[emi_ahb_gate], "emi_ahb" , NULL); - clk_register_clkdev(clk[ssi1_baud_gate], "bitrate" , "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_baud_gate], "bitrate" , "imx-ssi.1"); mxc_timer_init(MX27_IO_ADDRESS(MX27_GPT1_BASE_ADDR), MX27_INT_GPT1); diff --git a/arch/arm/mach-imx/clk-imx31.c b/arch/arm/mach-imx/clk-imx31.c index 1253af2..61567dc 100644 --- a/arch/arm/mach-imx/clk-imx31.c +++ b/arch/arm/mach-imx/clk-imx31.c @@ -35,8 +35,8 @@ static const char *csi_sel[] = { "upll", "spll", }; static const char *fir_sel[] = { "mcu_main", "upll", "spll" }; enum mx31_clks { - ckih, ckil, mpll, spll, upll, mcu_main, hsp, ahb, nfc, ipg, per_div, - per, csi, fir, csi_div, usb_div_pre, usb_div_post, fir_div_pre, + dummy, ckih, ckil, mpll, spll, upll, mcu_main, hsp, ahb, nfc, ipg, + per_div, per, csi, fir, csi_div, usb_div_pre, usb_div_post, fir_div_pre, fir_div_post, sdhc1_gate, sdhc2_gate, gpt_gate, epit1_gate, epit2_gate, iim_gate, ata_gate, sdma_gate, cspi3_gate, rng_gate, uart1_gate, uart2_gate, ssi1_gate, i2c1_gate, i2c2_gate, i2c3_gate, hantro_gate, @@ -53,6 +53,7 @@ int __init mx31_clocks_init(unsigned long fref) void __iomem *base = MX31_IO_ADDRESS(MX31_CCM_BASE_ADDR); int i; + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckih] = imx_clk_fixed("ckih", fref); clk[ckil] = imx_clk_fixed("ckil", 32768); clk[mpll] = imx_clk_pllv1("mpll", "ckih", base + MXC_CCM_MPCTL); @@ -161,8 +162,10 @@ int __init mx31_clocks_init(unsigned long fref) clk_register_clkdev(clk[owire_gate], NULL, "mxc_w1.0"); clk_register_clkdev(clk[sdhc1_gate], NULL, "mxc-mmc.0"); clk_register_clkdev(clk[sdhc2_gate], NULL, "mxc-mmc.1"); - clk_register_clkdev(clk[ssi1_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[firi_gate], "firi", NULL); clk_register_clkdev(clk[ata_gate], NULL, "pata_imx"); clk_register_clkdev(clk[rtic_gate], "rtic", NULL); diff --git a/arch/arm/mach-imx/clk-imx35.c b/arch/arm/mach-imx/clk-imx35.c index 177259b..1ab7467 100644 --- a/arch/arm/mach-imx/clk-imx35.c +++ b/arch/arm/mach-imx/clk-imx35.c @@ -50,7 +50,7 @@ static const char *std_sel[] = {"ppll", "arm"}; static const char *ipg_per_sel[] = {"ahb_per_div", "arm_per_div"}; enum mx35_clks { - ckih, mpll, ppll, mpll_075, arm, hsp, hsp_div, hsp_sel, ahb, ipg, + dummy, ckih, mpll, ppll, mpll_075, arm, hsp, hsp_div, hsp_sel, ahb, ipg, arm_per_div, ahb_per_div, ipg_per, uart_sel, uart_div, esdhc_sel, esdhc1_div, esdhc2_div, esdhc3_div, spdif_sel, spdif_div_pre, spdif_div_post, ssi_sel, ssi1_div_pre, ssi1_div_post, ssi2_div_pre, @@ -88,6 +88,7 @@ int __init mx35_clocks_init() aad = &clk_consumer[0]; } + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckih] = imx_clk_fixed("ckih", 24000000); clk[mpll] = imx_clk_pllv1("mpll", "ckih", base + MX35_CCM_MPCTL); clk[ppll] = imx_clk_pllv1("ppll", "ckih", base + MX35_CCM_PPCTL); @@ -234,8 +235,11 @@ int __init mx35_clocks_init() clk_register_clkdev(clk[kpp_gate], NULL, "imx-keypad"); clk_register_clkdev(clk[owire_gate], NULL, "mxc_w1"); clk_register_clkdev(clk[sdma_gate], NULL, "imx35-sdma"); - clk_register_clkdev(clk[ssi1_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); + /* i.mx35 has the i.mx21 type uart */ clk_register_clkdev(clk[uart1_gate], "per", "imx21-uart.0"); clk_register_clkdev(clk[ipg], "ipg", "imx21-uart.0"); diff --git a/arch/arm/mach-imx/clk-imx51-imx53.c b/arch/arm/mach-imx/clk-imx51-imx53.c index a0bf848..46d679c 100644 --- a/arch/arm/mach-imx/clk-imx51-imx53.c +++ b/arch/arm/mach-imx/clk-imx51-imx53.c @@ -273,9 +273,12 @@ static void __init mx5_clocks_common_init(unsigned long rate_ckil, clk_register_clkdev(clk[usboh3_gate], "ipg", "fsl-usb2-udc"); clk_register_clkdev(clk[usboh3_gate], "ahb", "fsl-usb2-udc"); clk_register_clkdev(clk[nfc_gate], NULL, "mxc_nand"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "imx-ssi.1"); - clk_register_clkdev(clk[ssi3_ipg_gate], NULL, "imx-ssi.2"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[ssi3_ipg_gate], "ipg", "imx-ssi.2"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.2"); clk_register_clkdev(clk[ssi_ext1_gate], "ssi_ext1", NULL); clk_register_clkdev(clk[ssi_ext2_gate], "ssi_ext2", NULL); clk_register_clkdev(clk[sdma_gate], NULL, "imx35-sdma"); @@ -366,9 +369,13 @@ int __init mx51_clocks_init(unsigned long rate_ckil, unsigned long rate_osc, clk_register_clkdev(clk[esdhc4_ipg_gate], "ipg", "sdhci-esdhc-imx51.3"); clk_register_clkdev(clk[dummy], "ahb", "sdhci-esdhc-imx51.3"); clk_register_clkdev(clk[esdhc4_per_gate], "per", "sdhci-esdhc-imx51.3"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "83fcc000.ssi"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "70014000.ssi"); - clk_register_clkdev(clk[ssi3_ipg_gate], NULL, "83fe8000.ssi"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "83fcc000.ssi"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "70014000.ssi"); + clk_register_clkdev(clk[ssi3_ipg_gate], "ipg", "83fe8000.ssi"); + clk_register_clkdev(clk[dummy], "per", "83fcc000.ssi"); + clk_register_clkdev(clk[dummy], "per", "70014000.ssi"); + clk_register_clkdev(clk[dummy], "per", "83fe8000.ssi"); + clk_register_clkdev(clk[nfc_gate], NULL, "83fdb000.nand"); /* set the usboh3 parent to pll2_sw */ -- 1.7.9.5

3 4

Re: [alsa-devel] issue of shdma-base library
by Guennadi Liakhovetski 09 Oct '12

09 Oct '12

Subject: [PATCH] ASoC: fsi: don't reschedule DMA from an atomic context shdma doesn't support transfer re-scheduling or triggering from callbacks or from atomic context. The fsi driver issues DMA transfers from a tasklet context, which is a bug. To fix it convert tasklet to a work. Reported-by: Do Q.Thang <dq-thang(a)jinso.co.jp> Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- Thang-san, Please, check, whether the attached patch fixes your issues. It seems to fix them for me. Mark, please, wait with committing this patch until the guys confirm, it really helps. It then will also have to go to 3.6-stable. Thanks Guennadi On Tue, 2 Oct 2012, Do Q.Thang wrote: > Hi Guennadi-san, > > I'm testing uptream kernel (v3.6-rc3 ~) on the kzm-a9-gt & armadillo-800eva > board. > I found 2 issues with the sh-dma driver while testing audio playback. > > issue 1: this log appeared while playback with the aplay, then aplay was > stopped. > > ------- log ----------------------------- > # aplay audio_s16le_44100.wav > Playing WAVE 'audio_s16le_44100.wav' : Signed 16 bit Little Endian, Rate > 44100 Hz, Stereo > aplay: pcm_write:1710: write error: Input/output error > ----------------------------------------- > > issue 2: this log appeared while playback with the aplay, then aplay was > stopped. > ( this issue isn't appear on the kzm-a9-gt board only ) > > ------- log ----------------------------- > # aplay audio_s16le_44100.wav > Playing WAVE 'audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo > BUG: scheduling while atomic: kworker/1:1/273/0x00000103 > aplay: pcm_write:1710: write error: Input/output error > ----------------------------------------- > > I think these issues are introduced by "dmaengine:add an shdma-base library" > commit. > > ---------- commit ----------------------- > commit 9a7b8e002e331d0599127f16613c32f425a14f2c > Author: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > Date: Wed May 9 17:09:13 2012 +0200 > > dmaengine: add an shdma-base library > > This patch extracts code from shdma.c, that does not directly deal with > hardware implementation details and can be re-used with diverse DMA > controller variants, found on SH-based SoCs. > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > Cc: Sascha Hauer <s.hauer(a)pengutronix.de> > Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> > ------------------------------------------ > > About issue 1: I found that > The fsi.c uses fsi_dma_do_tasklet() to setup dma-transfer.This is a function > of a tasklet. > When one dma-transfer is finished shdma_chan_ld_cleanup() is called.In this > function > __ld_cleanup() is called to cleanup the schan->ld_queue. In behavior of > __ld_cleanup() > if descriptor is the last chunk its callback is called, then its node is > cleanup in the > next call of __ld_cleanup().In callback of the last descriptor, > fsi_dma_do_tasklet() is > scheduled then is called after the shdma_chan_ld_cleanup() is finished. > But when this issue appear, shdma_chan_ld_cleanup() is called while > shdma_chan_ld_cleanup() > is not finished.At that time, schan->ld_queue is not empty, so in > shdma_tx_submit() > schan->pm_state is set to SHDMA_PM_PENDING, and then dma-transfer isn't > started in > shdma_issue_pending(). > > About issue 2: > I found that this issue is appear when pm_runtime_barrier() is running. > This issue is not appear when disable CONFIG_PM_RUNTIME config. > > What do you think about these issues? > > Best regards > --- > Thang diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 0540408..1bb0d58c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -20,6 +20,7 @@ #include <linux/sh_dma.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/workqueue.h> #include <sound/soc.h> #include <sound/sh_fsi.h> @@ -223,7 +224,7 @@ struct fsi_stream { */ struct dma_chan *chan; struct sh_dmae_slave slave; /* see fsi_handler_init() */ - struct tasklet_struct tasklet; + struct work_struct work; dma_addr_t dma; }; @@ -1085,9 +1086,9 @@ static void fsi_dma_complete(void *data) snd_pcm_period_elapsed(io->substream); } -static void fsi_dma_do_tasklet(unsigned long data) +static void fsi_dma_do_work(struct work_struct *work) { - struct fsi_stream *io = (struct fsi_stream *)data; + struct fsi_stream *io = container_of(work, struct fsi_stream, work); struct fsi_priv *fsi = fsi_stream_to_priv(io); struct snd_soc_dai *dai; struct dma_async_tx_descriptor *desc; @@ -1129,7 +1130,7 @@ static void fsi_dma_do_tasklet(unsigned long data) * FIXME * * In DMAEngine case, codec and FSI cannot be started simultaneously - * since FSI is using tasklet. + * since FSI is using the scheduler work queue. * Therefore, in capture case, probably FSI FIFO will have got * overflow error in this point. * in that case, DMA cannot start transfer until error was cleared. @@ -1153,7 +1154,7 @@ static bool fsi_dma_filter(struct dma_chan *chan, void *param) static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) { - tasklet_schedule(&io->tasklet); + schedule_work(&io->work); return 0; } @@ -1195,14 +1196,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev return fsi_stream_probe(fsi, dev); } - tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); + INIT_WORK(&io->work, fsi_dma_do_work); return 0; } static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) { - tasklet_kill(&io->tasklet); + cancel_work_sync(&io->work); fsi_stream_stop(fsi, io);

2 1

[alsa-devel] [PATCH] ASoC: fix documentation in soc-jack
by Peter Meerwald 09 Oct '12

09 Oct '12

From: Peter Meerwald <p.meerwald(a)bct-electronic.com> Signed-off-by: Peter Meerwald <pmeerw(a)pmeerw.net> --- sound/soc/soc-jack.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0c17293..b24cd53 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -22,7 +22,7 @@ /** * snd_soc_jack_new - Create a new jack - * @card: ASoC card + * @codec: ASoC codec * @id: an identifying string for this jack * @type: a bitmask of enum snd_jack_type values that can be detected by * this jack @@ -139,12 +139,13 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones); /** * snd_soc_jack_get_type - Based on the mic bias value, this function returns - * the type of jack from the zones delcared in the jack type + * the type of jack from the zones declared in the jack type * + * @jack: ASoC jack * @micbias_voltage: mic bias voltage at adc channel when jack is plugged in * * Based on the mic bias value passed, this function helps identify - * the type of jack from the already delcared jack zones + * the type of jack from the already declared jack zones */ int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage) { -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: ASoC: ak4104: handle more sample rates
by Daniel Mack 09 Oct '12

09 Oct '12

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/ak4104.c | 22 ++++++++++++++++++++-- 1 file changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 31d4483..d14163f 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -98,14 +98,32 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, val = 0; switch (params_rate(params)) { + case 22050: + val |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + val |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + val |= IEC958_AES3_CON_FS_32000; + break; case 44100: val |= IEC958_AES3_CON_FS_44100; break; case 48000: val |= IEC958_AES3_CON_FS_48000; break; - case 32000: - val |= IEC958_AES3_CON_FS_32000; + case 88200: + val |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + val |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + val |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + val |= IEC958_AES3_CON_FS_192000; break; default: dev_err(codec->dev, "unsupported sampling rate\n"); -- 1.7.11.4

2 2

[alsa-devel] [PATCH] ASoC: wm5102: Write register value corrections after SYSCLK is enabled
by Mark Brown 09 Oct '12

09 Oct '12

Evalation of the WM5102 has identified a number of register values which should be written after SYSCLK is enabled on revision A in order to improve performance. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 552 ++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 551 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index fdd7f27..d89652e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -43,6 +43,556 @@ static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static const struct reg_default wm5102_sysclk_reva_patch[] = { + { 0x3000, 0x2225 }, + { 0x3001, 0x3a03 }, + { 0x3002, 0x0225 }, + { 0x3003, 0x0801 }, + { 0x3004, 0x6249 }, + { 0x3005, 0x0c04 }, + { 0x3006, 0x0225 }, + { 0x3007, 0x5901 }, + { 0x3008, 0xe249 }, + { 0x3009, 0x030d }, + { 0x300a, 0x0249 }, + { 0x300b, 0x2c01 }, + { 0x300c, 0xe249 }, + { 0x300d, 0x4342 }, + { 0x300e, 0xe249 }, + { 0x300f, 0x73c0 }, + { 0x3010, 0x4249 }, + { 0x3011, 0x0c00 }, + { 0x3012, 0x0225 }, + { 0x3013, 0x1f01 }, + { 0x3014, 0x0225 }, + { 0x3015, 0x1e01 }, + { 0x3016, 0x0225 }, + { 0x3017, 0xfa00 }, + { 0x3018, 0x0000 }, + { 0x3019, 0xf000 }, + { 0x301a, 0x0000 }, + { 0x301b, 0xf000 }, + { 0x301c, 0x0000 }, + { 0x301d, 0xf000 }, + { 0x301e, 0x0000 }, + { 0x301f, 0xf000 }, + { 0x3020, 0x0000 }, + { 0x3021, 0xf000 }, + { 0x3022, 0x0000 }, + { 0x3023, 0xf000 }, + { 0x3024, 0x0000 }, + { 0x3025, 0xf000 }, + { 0x3026, 0x0000 }, + { 0x3027, 0xf000 }, + { 0x3028, 0x0000 }, + { 0x3029, 0xf000 }, + { 0x302a, 0x0000 }, + { 0x302b, 0xf000 }, + { 0x302c, 0x0000 }, + { 0x302d, 0xf000 }, + { 0x302e, 0x0000 }, + { 0x302f, 0xf000 }, + { 0x3030, 0x0225 }, + { 0x3031, 0x1a01 }, + { 0x3032, 0x0225 }, + { 0x3033, 0x1e00 }, + { 0x3034, 0x0225 }, + { 0x3035, 0x1f00 }, + { 0x3036, 0x6225 }, + { 0x3037, 0xf800 }, + { 0x3038, 0x0000 }, + { 0x3039, 0xf000 }, + { 0x303a, 0x0000 }, + { 0x303b, 0xf000 }, + { 0x303c, 0x0000 }, + { 0x303d, 0xf000 }, + { 0x303e, 0x0000 }, + { 0x303f, 0xf000 }, + { 0x3040, 0x2226 }, + { 0x3041, 0x3a03 }, + { 0x3042, 0x0226 }, + { 0x3043, 0x0801 }, + { 0x3044, 0x6249 }, + { 0x3045, 0x0c06 }, + { 0x3046, 0x0226 }, + { 0x3047, 0x5901 }, + { 0x3048, 0xe249 }, + { 0x3049, 0x030d }, + { 0x304a, 0x0249 }, + { 0x304b, 0x2c01 }, + { 0x304c, 0xe249 }, + { 0x304d, 0x4342 }, + { 0x304e, 0xe249 }, + { 0x304f, 0x73c0 }, + { 0x3050, 0x4249 }, + { 0x3051, 0x0c00 }, + { 0x3052, 0x0226 }, + { 0x3053, 0x1f01 }, + { 0x3054, 0x0226 }, + { 0x3055, 0x1e01 }, + { 0x3056, 0x0226 }, + { 0x3057, 0xfa00 }, + { 0x3058, 0x0000 }, + { 0x3059, 0xf000 }, + { 0x305a, 0x0000 }, + { 0x305b, 0xf000 }, + { 0x305c, 0x0000 }, + { 0x305d, 0xf000 }, + { 0x305e, 0x0000 }, + { 0x305f, 0xf000 }, + { 0x3060, 0x0000 }, + { 0x3061, 0xf000 }, + { 0x3062, 0x0000 }, + { 0x3063, 0xf000 }, + { 0x3064, 0x0000 }, + { 0x3065, 0xf000 }, + { 0x3066, 0x0000 }, + { 0x3067, 0xf000 }, + { 0x3068, 0x0000 }, + { 0x3069, 0xf000 }, + { 0x306a, 0x0000 }, + { 0x306b, 0xf000 }, + { 0x306c, 0x0000 }, + { 0x306d, 0xf000 }, + { 0x306e, 0x0000 }, + { 0x306f, 0xf000 }, + { 0x3070, 0x0226 }, + { 0x3071, 0x1a01 }, + { 0x3072, 0x0226 }, + { 0x3073, 0x1e00 }, + { 0x3074, 0x0226 }, + { 0x3075, 0x1f00 }, + { 0x3076, 0x6226 }, + { 0x3077, 0xf800 }, + { 0x3078, 0x0000 }, + { 0x3079, 0xf000 }, + { 0x307a, 0x0000 }, + { 0x307b, 0xf000 }, + { 0x307c, 0x0000 }, + { 0x307d, 0xf000 }, + { 0x307e, 0x0000 }, + { 0x307f, 0xf000 }, + { 0x3080, 0x2227 }, + { 0x3081, 0x3a03 }, + { 0x3082, 0x0227 }, + { 0x3083, 0x0801 }, + { 0x3084, 0x6255 }, + { 0x3085, 0x0c04 }, + { 0x3086, 0x0227 }, + { 0x3087, 0x5901 }, + { 0x3088, 0xe255 }, + { 0x3089, 0x030d }, + { 0x308a, 0x0255 }, + { 0x308b, 0x2c01 }, + { 0x308c, 0xe255 }, + { 0x308d, 0x4342 }, + { 0x308e, 0xe255 }, + { 0x308f, 0x73c0 }, + { 0x3090, 0x4255 }, + { 0x3091, 0x0c00 }, + { 0x3092, 0x0227 }, + { 0x3093, 0x1f01 }, + { 0x3094, 0x0227 }, + { 0x3095, 0x1e01 }, + { 0x3096, 0x0227 }, + { 0x3097, 0xfa00 }, + { 0x3098, 0x0000 }, + { 0x3099, 0xf000 }, + { 0x309a, 0x0000 }, + { 0x309b, 0xf000 }, + { 0x309c, 0x0000 }, + { 0x309d, 0xf000 }, + { 0x309e, 0x0000 }, + { 0x309f, 0xf000 }, + { 0x30a0, 0x0000 }, + { 0x30a1, 0xf000 }, + { 0x30a2, 0x0000 }, + { 0x30a3, 0xf000 }, + { 0x30a4, 0x0000 }, + { 0x30a5, 0xf000 }, + { 0x30a6, 0x0000 }, + { 0x30a7, 0xf000 }, + { 0x30a8, 0x0000 }, + { 0x30a9, 0xf000 }, + { 0x30aa, 0x0000 }, + { 0x30ab, 0xf000 }, + { 0x30ac, 0x0000 }, + { 0x30ad, 0xf000 }, + { 0x30ae, 0x0000 }, + { 0x30af, 0xf000 }, + { 0x30b0, 0x0227 }, + { 0x30b1, 0x1a01 }, + { 0x30b2, 0x0227 }, + { 0x30b3, 0x1e00 }, + { 0x30b4, 0x0227 }, + { 0x30b5, 0x1f00 }, + { 0x30b6, 0x6227 }, + { 0x30b7, 0xf800 }, + { 0x30b8, 0x0000 }, + { 0x30b9, 0xf000 }, + { 0x30ba, 0x0000 }, + { 0x30bb, 0xf000 }, + { 0x30bc, 0x0000 }, + { 0x30bd, 0xf000 }, + { 0x30be, 0x0000 }, + { 0x30bf, 0xf000 }, + { 0x30c0, 0x2228 }, + { 0x30c1, 0x3a03 }, + { 0x30c2, 0x0228 }, + { 0x30c3, 0x0801 }, + { 0x30c4, 0x6255 }, + { 0x30c5, 0x0c06 }, + { 0x30c6, 0x0228 }, + { 0x30c7, 0x5901 }, + { 0x30c8, 0xe255 }, + { 0x30c9, 0x030d }, + { 0x30ca, 0x0255 }, + { 0x30cb, 0x2c01 }, + { 0x30cc, 0xe255 }, + { 0x30cd, 0x4342 }, + { 0x30ce, 0xe255 }, + { 0x30cf, 0x73c0 }, + { 0x30d0, 0x4255 }, + { 0x30d1, 0x0c00 }, + { 0x30d2, 0x0228 }, + { 0x30d3, 0x1f01 }, + { 0x30d4, 0x0228 }, + { 0x30d5, 0x1e01 }, + { 0x30d6, 0x0228 }, + { 0x30d7, 0xfa00 }, + { 0x30d8, 0x0000 }, + { 0x30d9, 0xf000 }, + { 0x30da, 0x0000 }, + { 0x30db, 0xf000 }, + { 0x30dc, 0x0000 }, + { 0x30dd, 0xf000 }, + { 0x30de, 0x0000 }, + { 0x30df, 0xf000 }, + { 0x30e0, 0x0000 }, + { 0x30e1, 0xf000 }, + { 0x30e2, 0x0000 }, + { 0x30e3, 0xf000 }, + { 0x30e4, 0x0000 }, + { 0x30e5, 0xf000 }, + { 0x30e6, 0x0000 }, + { 0x30e7, 0xf000 }, + { 0x30e8, 0x0000 }, + { 0x30e9, 0xf000 }, + { 0x30ea, 0x0000 }, + { 0x30eb, 0xf000 }, + { 0x30ec, 0x0000 }, + { 0x30ed, 0xf000 }, + { 0x30ee, 0x0000 }, + { 0x30ef, 0xf000 }, + { 0x30f0, 0x0228 }, + { 0x30f1, 0x1a01 }, + { 0x30f2, 0x0228 }, + { 0x30f3, 0x1e00 }, + { 0x30f4, 0x0228 }, + { 0x30f5, 0x1f00 }, + { 0x30f6, 0x6228 }, + { 0x30f7, 0xf800 }, + { 0x30f8, 0x0000 }, + { 0x30f9, 0xf000 }, + { 0x30fa, 0x0000 }, + { 0x30fb, 0xf000 }, + { 0x30fc, 0x0000 }, + { 0x30fd, 0xf000 }, + { 0x30fe, 0x0000 }, + { 0x30ff, 0xf000 }, + { 0x3100, 0x222b }, + { 0x3101, 0x3a03 }, + { 0x3102, 0x222b }, + { 0x3103, 0x5803 }, + { 0x3104, 0xe26f }, + { 0x3105, 0x030d }, + { 0x3106, 0x626f }, + { 0x3107, 0x2c01 }, + { 0x3108, 0xe26f }, + { 0x3109, 0x4342 }, + { 0x310a, 0xe26f }, + { 0x310b, 0x73c0 }, + { 0x310c, 0x026f }, + { 0x310d, 0x0c00 }, + { 0x310e, 0x022b }, + { 0x310f, 0x1f01 }, + { 0x3110, 0x022b }, + { 0x3111, 0x1e01 }, + { 0x3112, 0x022b }, + { 0x3113, 0xfa00 }, + { 0x3114, 0x0000 }, + { 0x3115, 0xf000 }, + { 0x3116, 0x0000 }, + { 0x3117, 0xf000 }, + { 0x3118, 0x0000 }, + { 0x3119, 0xf000 }, + { 0x311a, 0x0000 }, + { 0x311b, 0xf000 }, + { 0x311c, 0x0000 }, + { 0x311d, 0xf000 }, + { 0x311e, 0x0000 }, + { 0x311f, 0xf000 }, + { 0x3120, 0x022b }, + { 0x3121, 0x0a01 }, + { 0x3122, 0x022b }, + { 0x3123, 0x1e00 }, + { 0x3124, 0x022b }, + { 0x3125, 0x1f00 }, + { 0x3126, 0x622b }, + { 0x3127, 0xf800 }, + { 0x3128, 0x0000 }, + { 0x3129, 0xf000 }, + { 0x312a, 0x0000 }, + { 0x312b, 0xf000 }, + { 0x312c, 0x0000 }, + { 0x312d, 0xf000 }, + { 0x312e, 0x0000 }, + { 0x312f, 0xf000 }, + { 0x3130, 0x0000 }, + { 0x3131, 0xf000 }, + { 0x3132, 0x0000 }, + { 0x3133, 0xf000 }, + { 0x3134, 0x0000 }, + { 0x3135, 0xf000 }, + { 0x3136, 0x0000 }, + { 0x3137, 0xf000 }, + { 0x3138, 0x0000 }, + { 0x3139, 0xf000 }, + { 0x313a, 0x0000 }, + { 0x313b, 0xf000 }, + { 0x313c, 0x0000 }, + { 0x313d, 0xf000 }, + { 0x313e, 0x0000 }, + { 0x313f, 0xf000 }, + { 0x3140, 0x0000 }, + { 0x3141, 0xf000 }, + { 0x3142, 0x0000 }, + { 0x3143, 0xf000 }, + { 0x3144, 0x0000 }, + { 0x3145, 0xf000 }, + { 0x3146, 0x0000 }, + { 0x3147, 0xf000 }, + { 0x3148, 0x0000 }, + { 0x3149, 0xf000 }, + { 0x314a, 0x0000 }, + { 0x314b, 0xf000 }, + { 0x314c, 0x0000 }, + { 0x314d, 0xf000 }, + { 0x314e, 0x0000 }, + { 0x314f, 0xf000 }, + { 0x3150, 0x0000 }, + { 0x3151, 0xf000 }, + { 0x3152, 0x0000 }, + { 0x3153, 0xf000 }, + { 0x3154, 0x0000 }, + { 0x3155, 0xf000 }, + { 0x3156, 0x0000 }, + { 0x3157, 0xf000 }, + { 0x3158, 0x0000 }, + { 0x3159, 0xf000 }, + { 0x315a, 0x0000 }, + { 0x315b, 0xf000 }, + { 0x315c, 0x0000 }, + { 0x315d, 0xf000 }, + { 0x315e, 0x0000 }, + { 0x315f, 0xf000 }, + { 0x3160, 0x0000 }, + { 0x3161, 0xf000 }, + { 0x3162, 0x0000 }, + { 0x3163, 0xf000 }, + { 0x3164, 0x0000 }, + { 0x3165, 0xf000 }, + { 0x3166, 0x0000 }, + { 0x3167, 0xf000 }, + { 0x3168, 0x0000 }, + { 0x3169, 0xf000 }, + { 0x316a, 0x0000 }, + { 0x316b, 0xf000 }, + { 0x316c, 0x0000 }, + { 0x316d, 0xf000 }, + { 0x316e, 0x0000 }, + { 0x316f, 0xf000 }, + { 0x3170, 0x0000 }, + { 0x3171, 0xf000 }, + { 0x3172, 0x0000 }, + { 0x3173, 0xf000 }, + { 0x3174, 0x0000 }, + { 0x3175, 0xf000 }, + { 0x3176, 0x0000 }, + { 0x3177, 0xf000 }, + { 0x3178, 0x0000 }, + { 0x3179, 0xf000 }, + { 0x317a, 0x0000 }, + { 0x317b, 0xf000 }, + { 0x317c, 0x0000 }, + { 0x317d, 0xf000 }, + { 0x317e, 0x0000 }, + { 0x317f, 0xf000 }, + { 0x3180, 0x2001 }, + { 0x3181, 0xf101 }, + { 0x3182, 0x0000 }, + { 0x3183, 0xf000 }, + { 0x3184, 0x0000 }, + { 0x3185, 0xf000 }, + { 0x3186, 0x0000 }, + { 0x3187, 0xf000 }, + { 0x3188, 0x0000 }, + { 0x3189, 0xf000 }, + { 0x318a, 0x0000 }, + { 0x318b, 0xf000 }, + { 0x318c, 0x0000 }, + { 0x318d, 0xf000 }, + { 0x318e, 0x0000 }, + { 0x318f, 0xf000 }, + { 0x3190, 0x0000 }, + { 0x3191, 0xf000 }, + { 0x3192, 0x0000 }, + { 0x3193, 0xf000 }, + { 0x3194, 0x0000 }, + { 0x3195, 0xf000 }, + { 0x3196, 0x0000 }, + { 0x3197, 0xf000 }, + { 0x3198, 0x0000 }, + { 0x3199, 0xf000 }, + { 0x319a, 0x0000 }, + { 0x319b, 0xf000 }, + { 0x319c, 0x0000 }, + { 0x319d, 0xf000 }, + { 0x319e, 0x0000 }, + { 0x319f, 0xf000 }, + { 0x31a0, 0x0000 }, + { 0x31a1, 0xf000 }, + { 0x31a2, 0x0000 }, + { 0x31a3, 0xf000 }, + { 0x31a4, 0x0000 }, + { 0x31a5, 0xf000 }, + { 0x31a6, 0x0000 }, + { 0x31a7, 0xf000 }, + { 0x31a8, 0x0000 }, + { 0x31a9, 0xf000 }, + { 0x31aa, 0x0000 }, + { 0x31ab, 0xf000 }, + { 0x31ac, 0x0000 }, + { 0x31ad, 0xf000 }, + { 0x31ae, 0x0000 }, + { 0x31af, 0xf000 }, + { 0x31b0, 0x0000 }, + { 0x31b1, 0xf000 }, + { 0x31b2, 0x0000 }, + { 0x31b3, 0xf000 }, + { 0x31b4, 0x0000 }, + { 0x31b5, 0xf000 }, + { 0x31b6, 0x0000 }, + { 0x31b7, 0xf000 }, + { 0x31b8, 0x0000 }, + { 0x31b9, 0xf000 }, + { 0x31ba, 0x0000 }, + { 0x31bb, 0xf000 }, + { 0x31bc, 0x0000 }, + { 0x31bd, 0xf000 }, + { 0x31be, 0x0000 }, + { 0x31bf, 0xf000 }, + { 0x31c0, 0x0000 }, + { 0x31c1, 0xf000 }, + { 0x31c2, 0x0000 }, + { 0x31c3, 0xf000 }, + { 0x31c4, 0x0000 }, + { 0x31c5, 0xf000 }, + { 0x31c6, 0x0000 }, + { 0x31c7, 0xf000 }, + { 0x31c8, 0x0000 }, + { 0x31c9, 0xf000 }, + { 0x31ca, 0x0000 }, + { 0x31cb, 0xf000 }, + { 0x31cc, 0x0000 }, + { 0x31cd, 0xf000 }, + { 0x31ce, 0x0000 }, + { 0x31cf, 0xf000 }, + { 0x31d0, 0x0000 }, + { 0x31d1, 0xf000 }, + { 0x31d2, 0x0000 }, + { 0x31d3, 0xf000 }, + { 0x31d4, 0x0000 }, + { 0x31d5, 0xf000 }, + { 0x31d6, 0x0000 }, + { 0x31d7, 0xf000 }, + { 0x31d8, 0x0000 }, + { 0x31d9, 0xf000 }, + { 0x31da, 0x0000 }, + { 0x31db, 0xf000 }, + { 0x31dc, 0x0000 }, + { 0x31dd, 0xf000 }, + { 0x31de, 0x0000 }, + { 0x31df, 0xf000 }, + { 0x31e0, 0x0000 }, + { 0x31e1, 0xf000 }, + { 0x31e2, 0x0000 }, + { 0x31e3, 0xf000 }, + { 0x31e4, 0x0000 }, + { 0x31e5, 0xf000 }, + { 0x31e6, 0x0000 }, + { 0x31e7, 0xf000 }, + { 0x31e8, 0x0000 }, + { 0x31e9, 0xf000 }, + { 0x31ea, 0x0000 }, + { 0x31eb, 0xf000 }, + { 0x31ec, 0x0000 }, + { 0x31ed, 0xf000 }, + { 0x31ee, 0x0000 }, + { 0x31ef, 0xf000 }, + { 0x31f0, 0x0000 }, + { 0x31f1, 0xf000 }, + { 0x31f2, 0x0000 }, + { 0x31f3, 0xf000 }, + { 0x31f4, 0x0000 }, + { 0x31f5, 0xf000 }, + { 0x31f6, 0x0000 }, + { 0x31f7, 0xf000 }, + { 0x31f8, 0x0000 }, + { 0x31f9, 0xf000 }, + { 0x31fa, 0x0000 }, + { 0x31fb, 0xf000 }, + { 0x31fc, 0x0000 }, + { 0x31fd, 0xf000 }, + { 0x31fe, 0x0000 }, + { 0x31ff, 0xf000 }, + { 0x024d, 0xff50 }, + { 0x0252, 0xff50 }, + { 0x0259, 0x0112 }, + { 0x025e, 0x0112 }, +}; + +static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 0: + patch = wm5102_sysclk_reva_patch; + patch_size = ARRAY_SIZE(wm5102_sysclk_reva_patch); + break; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + + default: + break; + } + + return 0; +} + static const struct snd_kcontrol_new wm5102_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, ARIZONA_IN1_OSR_SHIFT, 1, 0), @@ -305,7 +855,7 @@ static const struct snd_kcontrol_new wm5102_aec_loopback_mux = static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, - 0, NULL, 0), + 0, wm5102_sysclk_ev, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, -- 1.7.10.4

1 0

Re: [alsa-devel] [PATCH 2.6.39-rc3] hda_intel: ALSA HD Audio patch for Intel Panther Point DeviceIDs
by Takashi Iwai 09 Oct '12

09 Oct '12

At Wed, 20 Apr 2011 10:59:57 -0700, Seth Heasley wrote: > > This patch adds the HD Audio Controller DeviceIDs for the Intel Panther Point PCH. > > Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> Applied now. Thanks. Takashi > --- > --- linux-2.6.39-rc3/sound/pci/hda/hda_intel.c.orig 2011-04-11 17:21:51.000000000 -0700 > +++ linux-2.6.39-rc3/sound/pci/hda/hda_intel.c 2011-04-14 11:44:40.000000000 -0700 > @@ -126,6 +126,7 @@ > "{Intel, ICH10}," > "{Intel, PCH}," > "{Intel, CPT}," > + "{Intel, PPT}," > "{Intel, PBG}," > "{Intel, SCH}," > "{ATI, SB450}," > @@ -2759,6 +2760,8 @@ > { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH }, > + /* Panther Point */ > + { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, > /* Generic Intel */ >

3 2

[alsa-devel] [PATCH 1/2] ARM: imx: clk: Split SSI clock into 'ipg' and 'per'
by Fabio Estevam 08 Oct '12

08 Oct '12

Currently imx ssi driver has been using only the ipg clock as only slave mode is supported. For mx27 it is necessay to provide both 'ipg' and 'per' clocks in order to get ssi functional. This issue was found on mx27 by unselecting the mmc driver from the kernel ,which resulted on a system that could not play audio. When the mmc driver is selected the required 'per2' clock is provided and it allows the ssi to work. Add the required 'per' clock to the mx27 ssi and a dummy one for the other SoCs. Reported-by: Gaëtan Carlier <gcembed(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- arch/arm/mach-imx/clk-imx21.c | 11 +++++++---- arch/arm/mach-imx/clk-imx25.c | 6 ++++-- arch/arm/mach-imx/clk-imx27.c | 9 +++++---- arch/arm/mach-imx/clk-imx31.c | 11 +++++++---- arch/arm/mach-imx/clk-imx35.c | 12 ++++++++---- arch/arm/mach-imx/clk-imx51-imx53.c | 19 +++++++++++++------ 6 files changed, 44 insertions(+), 24 deletions(-) diff --git a/arch/arm/mach-imx/clk-imx21.c b/arch/arm/mach-imx/clk-imx21.c index cf65148..2ed49c8 100644 --- a/arch/arm/mach-imx/clk-imx21.c +++ b/arch/arm/mach-imx/clk-imx21.c @@ -51,8 +51,8 @@ static const char *mpll_sel_clks[] = { "fpm", "ckih", }; static const char *spll_sel_clks[] = { "fpm", "ckih", }; enum imx21_clks { - ckil, ckih, fpm, mpll_sel, spll_sel, mpll, spll, fclk, hclk, ipg, per1, - per2, per3, per4, uart1_ipg_gate, uart2_ipg_gate, uart3_ipg_gate, + dummy, ckil, ckih, fpm, mpll_sel, spll_sel, mpll, spll, fclk, hclk, ipg, + per1, per2, per3, per4, uart1_ipg_gate, uart2_ipg_gate, uart3_ipg_gate, uart4_ipg_gate, gpt1_ipg_gate, gpt2_ipg_gate, gpt3_ipg_gate, pwm_ipg_gate, sdhc1_ipg_gate, sdhc2_ipg_gate, lcdc_ipg_gate, lcdc_hclk_gate, cspi3_ipg_gate, cspi2_ipg_gate, cspi1_ipg_gate, @@ -72,6 +72,7 @@ int __init mx21_clocks_init(unsigned long lref, unsigned long href) { int i; + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckil] = imx_clk_fixed("ckil", lref); clk[ckih] = imx_clk_fixed("ckih", href); clk[fpm] = imx_clk_fixed_factor("fpm", "ckil", 512, 1); @@ -174,8 +175,10 @@ int __init mx21_clocks_init(unsigned long lref, unsigned long href) clk_register_clkdev(clk[gpio_gate], "gpio", NULL); clk_register_clkdev(clk[rtc_gate], "rtc", NULL); clk_register_clkdev(clk[csi_hclk_gate], "csi", NULL); - clk_register_clkdev(clk[ssi1_gate], "ssi1", NULL); - clk_register_clkdev(clk[ssi2_gate], "ssi2", NULL); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[sdhc1_ipg_gate], "sdhc1", NULL); clk_register_clkdev(clk[sdhc2_ipg_gate], "sdhc2", NULL); diff --git a/arch/arm/mach-imx/clk-imx25.c b/arch/arm/mach-imx/clk-imx25.c index d20d479..c81864d 100644 --- a/arch/arm/mach-imx/clk-imx25.c +++ b/arch/arm/mach-imx/clk-imx25.c @@ -222,8 +222,10 @@ int __init mx25_clocks_init(void) clk_register_clkdev(clk[lcdc_ipg], "ipg", "imx-fb.0"); clk_register_clkdev(clk[lcdc_ahb], "ahb", "imx-fb.0"); clk_register_clkdev(clk[wdt_ipg], NULL, "imx2-wdt.0"); - clk_register_clkdev(clk[ssi1_ipg], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_ipg], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[esdhc1_ipg_per], "per", "sdhci-esdhc-imx25.0"); clk_register_clkdev(clk[esdhc1_ipg], "ipg", "sdhci-esdhc-imx25.0"); clk_register_clkdev(clk[esdhc1_ahb], "ahb", "sdhci-esdhc-imx25.0"); diff --git a/arch/arm/mach-imx/clk-imx27.c b/arch/arm/mach-imx/clk-imx27.c index 3b6b640..453b4f3 100644 --- a/arch/arm/mach-imx/clk-imx27.c +++ b/arch/arm/mach-imx/clk-imx27.c @@ -224,6 +224,7 @@ int __init mx27_clocks_init(unsigned long fref) clk_register_clkdev(clk[lcdc_ipg_gate], "ipg", "imx-fb.0"); clk_register_clkdev(clk[lcdc_ahb_gate], "ahb", "imx-fb.0"); clk_register_clkdev(clk[csi_ahb_gate], "ahb", "mx2-camera.0"); + clk_register_clkdev(clk[per4_gate], "per", "mx2-camera.0"); clk_register_clkdev(clk[usb_div], "per", "fsl-usb2-udc"); clk_register_clkdev(clk[usb_ipg_gate], "ipg", "fsl-usb2-udc"); clk_register_clkdev(clk[usb_ahb_gate], "ahb", "fsl-usb2-udc"); @@ -236,8 +237,10 @@ int __init mx27_clocks_init(unsigned long fref) clk_register_clkdev(clk[usb_div], "per", "mxc-ehci.2"); clk_register_clkdev(clk[usb_ipg_gate], "ipg", "mxc-ehci.2"); clk_register_clkdev(clk[usb_ahb_gate], "ahb", "mxc-ehci.2"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[per2_gate], "per", "imx-ssi.0"); + clk_register_clkdev(clk[per2_gate], "per", "imx-ssi.1"); clk_register_clkdev(clk[nfc_baud_gate], NULL, "mxc_nand.0"); clk_register_clkdev(clk[vpu_baud_gate], "per", "coda-imx27.0"); clk_register_clkdev(clk[vpu_ahb_gate], "ahb", "coda-imx27.0"); @@ -262,8 +265,6 @@ int __init mx27_clocks_init(unsigned long fref) clk_register_clkdev(clk[scc_ipg_gate], "scc", NULL); clk_register_clkdev(clk[cpu_div], "cpu", NULL); clk_register_clkdev(clk[emi_ahb_gate], "emi_ahb" , NULL); - clk_register_clkdev(clk[ssi1_baud_gate], "bitrate" , "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_baud_gate], "bitrate" , "imx-ssi.1"); mxc_timer_init(MX27_IO_ADDRESS(MX27_GPT1_BASE_ADDR), MX27_INT_GPT1); diff --git a/arch/arm/mach-imx/clk-imx31.c b/arch/arm/mach-imx/clk-imx31.c index 1253af2..a653ecf 100644 --- a/arch/arm/mach-imx/clk-imx31.c +++ b/arch/arm/mach-imx/clk-imx31.c @@ -35,8 +35,8 @@ static const char *csi_sel[] = { "upll", "spll", }; static const char *fir_sel[] = { "mcu_main", "upll", "spll" }; enum mx31_clks { - ckih, ckil, mpll, spll, upll, mcu_main, hsp, ahb, nfc, ipg, per_div, - per, csi, fir, csi_div, usb_div_pre, usb_div_post, fir_div_pre, + dummy, ckih, ckil, mpll, spll, upll, mcu_main, hsp, ahb, nfc, ipg, + per_div, per, csi, fir, csi_div, usb_div_pre, usb_div_post, fir_div_pre, fir_div_post, sdhc1_gate, sdhc2_gate, gpt_gate, epit1_gate, epit2_gate, iim_gate, ata_gate, sdma_gate, cspi3_gate, rng_gate, uart1_gate, uart2_gate, ssi1_gate, i2c1_gate, i2c2_gate, i2c3_gate, hantro_gate, @@ -53,6 +53,7 @@ int __init mx31_clocks_init(unsigned long fref) void __iomem *base = MX31_IO_ADDRESS(MX31_CCM_BASE_ADDR); int i; + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckih] = imx_clk_fixed("ckih", fref); clk[ckil] = imx_clk_fixed("ckil", 32768); clk[mpll] = imx_clk_pllv1("mpll", "ckih", base + MXC_CCM_MPCTL); @@ -161,8 +162,10 @@ int __init mx31_clocks_init(unsigned long fref) clk_register_clkdev(clk[owire_gate], NULL, "mxc_w1.0"); clk_register_clkdev(clk[sdhc1_gate], NULL, "mxc-mmc.0"); clk_register_clkdev(clk[sdhc2_gate], NULL, "mxc-mmc.1"); - clk_register_clkdev(clk[ssi1_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[firi_gate], "firi", NULL); clk_register_clkdev(clk[ata_gate], NULL, "pata_imx"); clk_register_clkdev(clk[rtic_gate], "rtic", NULL); diff --git a/arch/arm/mach-imx/clk-imx35.c b/arch/arm/mach-imx/clk-imx35.c index 177259b..0d952c7 100644 --- a/arch/arm/mach-imx/clk-imx35.c +++ b/arch/arm/mach-imx/clk-imx35.c @@ -50,7 +50,7 @@ static const char *std_sel[] = {"ppll", "arm"}; static const char *ipg_per_sel[] = {"ahb_per_div", "arm_per_div"}; enum mx35_clks { - ckih, mpll, ppll, mpll_075, arm, hsp, hsp_div, hsp_sel, ahb, ipg, + dummy, ckih, mpll, ppll, mpll_075, arm, hsp, hsp_div, hsp_sel, ahb, ipg, arm_per_div, ahb_per_div, ipg_per, uart_sel, uart_div, esdhc_sel, esdhc1_div, esdhc2_div, esdhc3_div, spdif_sel, spdif_div_pre, spdif_div_post, ssi_sel, ssi1_div_pre, ssi1_div_post, ssi2_div_pre, @@ -87,7 +87,8 @@ int __init mx35_clocks_init() */ aad = &clk_consumer[0]; } - + + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckih] = imx_clk_fixed("ckih", 24000000); clk[mpll] = imx_clk_pllv1("mpll", "ckih", base + MX35_CCM_MPCTL); clk[ppll] = imx_clk_pllv1("ppll", "ckih", base + MX35_CCM_PPCTL); @@ -234,8 +235,11 @@ int __init mx35_clocks_init() clk_register_clkdev(clk[kpp_gate], NULL, "imx-keypad"); clk_register_clkdev(clk[owire_gate], NULL, "mxc_w1"); clk_register_clkdev(clk[sdma_gate], NULL, "imx35-sdma"); - clk_register_clkdev(clk[ssi1_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); + /* i.mx35 has the i.mx21 type uart */ clk_register_clkdev(clk[uart1_gate], "per", "imx21-uart.0"); clk_register_clkdev(clk[ipg], "ipg", "imx21-uart.0"); diff --git a/arch/arm/mach-imx/clk-imx51-imx53.c b/arch/arm/mach-imx/clk-imx51-imx53.c index a0bf848..46d679c 100644 --- a/arch/arm/mach-imx/clk-imx51-imx53.c +++ b/arch/arm/mach-imx/clk-imx51-imx53.c @@ -273,9 +273,12 @@ static void __init mx5_clocks_common_init(unsigned long rate_ckil, clk_register_clkdev(clk[usboh3_gate], "ipg", "fsl-usb2-udc"); clk_register_clkdev(clk[usboh3_gate], "ahb", "fsl-usb2-udc"); clk_register_clkdev(clk[nfc_gate], NULL, "mxc_nand"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "imx-ssi.1"); - clk_register_clkdev(clk[ssi3_ipg_gate], NULL, "imx-ssi.2"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[ssi3_ipg_gate], "ipg", "imx-ssi.2"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.2"); clk_register_clkdev(clk[ssi_ext1_gate], "ssi_ext1", NULL); clk_register_clkdev(clk[ssi_ext2_gate], "ssi_ext2", NULL); clk_register_clkdev(clk[sdma_gate], NULL, "imx35-sdma"); @@ -366,9 +369,13 @@ int __init mx51_clocks_init(unsigned long rate_ckil, unsigned long rate_osc, clk_register_clkdev(clk[esdhc4_ipg_gate], "ipg", "sdhci-esdhc-imx51.3"); clk_register_clkdev(clk[dummy], "ahb", "sdhci-esdhc-imx51.3"); clk_register_clkdev(clk[esdhc4_per_gate], "per", "sdhci-esdhc-imx51.3"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "83fcc000.ssi"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "70014000.ssi"); - clk_register_clkdev(clk[ssi3_ipg_gate], NULL, "83fe8000.ssi"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "83fcc000.ssi"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "70014000.ssi"); + clk_register_clkdev(clk[ssi3_ipg_gate], "ipg", "83fe8000.ssi"); + clk_register_clkdev(clk[dummy], "per", "83fcc000.ssi"); + clk_register_clkdev(clk[dummy], "per", "70014000.ssi"); + clk_register_clkdev(clk[dummy], "per", "83fe8000.ssi"); + clk_register_clkdev(clk[nfc_gate], NULL, "83fdb000.nand"); /* set the usboh3 parent to pll2_sw */ -- 1.7.9.5

1 1

[alsa-devel] [PATCH] pcm: support for audio timestamps
by Pierre-Louis Bossart 08 Oct '12

08 Oct '12

add new snd_pcm_status_get_audio_htstamp() routine to query the audio timestamps provided by the kernel. This change provides applications with better ways to track elapsed time. Before this patch, applications would subtract queued samples (delay) from written samples, resulting in a 1-2 sample error. Also add snd_pcm_hw_params_supports_audio_wallclock_ts() to query what the hardware supports. TODO: check protocol compatibility? Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/pcm.h | 2 ++ include/sound/asound.h | 7 +++-- src/pcm/pcm.c | 31 ++++++++++++++++++++++ test/audio_time.c | 68 +++++++++++++++++++++++++++++++++++++++++------- 4 files changed, 96 insertions(+), 12 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 290593b..1ce91e7 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -631,6 +631,7 @@ int snd_pcm_hw_params_is_half_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_is_joint_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_sync_start(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *params); +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_get_rate_numden(const snd_pcm_hw_params_t *params, unsigned int *rate_num, unsigned int *rate_den); @@ -939,6 +940,7 @@ void snd_pcm_status_get_trigger_tstamp(const snd_pcm_status_t *obj, snd_timestam void snd_pcm_status_get_trigger_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); void snd_pcm_status_get_tstamp(const snd_pcm_status_t *obj, snd_timestamp_t *ptr); void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); +void snd_pcm_status_get_audio_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); diff --git a/include/sound/asound.h b/include/sound/asound.h index e24d144..16d03e8 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -160,7 +160,7 @@ enum { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) typedef unsigned long sndrv_pcm_uframes_t; typedef long sndrv_pcm_sframes_t; @@ -285,6 +285,7 @@ enum sndrv_pcm_subformat { #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ enum sndrv_pcm_state { SNDRV_PCM_STATE_OPEN = 0, /* stream is open */ @@ -426,7 +427,8 @@ struct sndrv_pcm_status { sndrv_pcm_uframes_t avail_max; /* max frames available on hw since last status */ sndrv_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ int suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ + unsigned char reserved[60-sizeof(struct timespec)]; /* must be filled with zero */ }; struct sndrv_pcm_mmap_status { @@ -435,6 +437,7 @@ struct sndrv_pcm_mmap_status { sndrv_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ int suspended_state; /* RO: suspended stream state */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ }; struct sndrv_pcm_mmap_control { diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 65c7646..5880057 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -3124,6 +3124,26 @@ int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *param } /** + * \brief Check if hardware supports audio wallclock timestamps + * \param params Configuration space + * \retval 0 Hardware doesn't support audio wallclock timestamps + * \retval 1 Hardware supports audio wallclock timestamps + * + * This function should only be called when the configuration space + * contains a single configuration. Call #snd_pcm_hw_params to choose + * a single configuration from the configuration space. + */ +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params) +{ + assert(params); + if (CHECK_SANITY(params->info == ~0U)) { + SNDMSG("invalid PCM info field"); + return 0; /* FIXME: should be a negative error? */ + } + return !!(params->info & SNDRV_PCM_INFO_HAS_WALL_CLOCK); +} + +/** * \brief Get rate exact info from a configuration space * \param params Configuration space * \param rate_num Pointer to returned rate numerator @@ -6214,6 +6234,17 @@ void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *p use_default_symbol_version(__snd_pcm_status_get_htstamp, snd_pcm_status_get_htstamp, ALSA_0.9.0rc8); /** + * \brief Get "now" hi-res audio timestamp from a PCM status container + * \param obj pointer to #snd_pcm_status_t + * \param ptr Pointer to returned timestamp + */ +void snd_pcm_status_get_audio_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +{ + assert(obj && ptr); + *ptr = obj->audio_tstamp; +} + +/** * \brief Get delay from a PCM status container (see #snd_pcm_delay) * \return Delay in frames * diff --git a/test/audio_time.c b/test/audio_time.c index a910783..03817c7 100644 --- a/test/audio_time.c +++ b/test/audio_time.c @@ -33,6 +33,7 @@ long long timediff(snd_htimestamp_t t1, snd_htimestamp_t t2) void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, snd_htimestamp_t *trigger_timestamp, + snd_htimestamp_t *audio_timestamp, snd_pcm_uframes_t *avail, snd_pcm_sframes_t *delay) { int err; @@ -45,6 +46,7 @@ void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, } snd_pcm_status_get_trigger_htstamp(status, trigger_timestamp); snd_pcm_status_get_htstamp(status, timestamp); + snd_pcm_status_get_audio_htstamp(status, audio_timestamp); *avail = snd_pcm_status_get_avail(status); *delay = snd_pcm_status_get_delay(status); } @@ -53,6 +55,7 @@ void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, #define PCM_LINK /* sync start for playback and capture */ #define TRACK_CAPTURE /* dump capture timing info */ #define TRACK_PLAYBACK /* dump playback timing info */ +#define TRACK_SAMPLE_COUNTS /* show difference between sample counters and audiotimestamps returned by driver */ #define PLAYBACK_BUFFERS 4 @@ -65,9 +68,13 @@ int main(void) snd_pcm_sframes_t frames; snd_htimestamp_t tstamp_c, tstamp_p; snd_htimestamp_t trigger_tstamp_c, trigger_tstamp_p; + snd_htimestamp_t audio_tstamp_c, audio_tstamp_p; unsigned char buffer_p[PERIOD*4*4]; unsigned char buffer_c[PERIOD*4*4]; + snd_pcm_hw_params_t *hwparams_p; + snd_pcm_hw_params_t *hwparams_c; + snd_pcm_sw_params_t *swparams_p; snd_pcm_sw_params_t *swparams_c; @@ -94,6 +101,18 @@ int main(void) goto _exit; } + snd_pcm_hw_params_alloca(&hwparams_p); + /* get the current hwparams */ + err = snd_pcm_hw_params_current(handle_p, hwparams_p); + if (err < 0) { + printf("Unable to determine current hwparams_p: %s\n", snd_strerror(err)); + goto _exit; + } + if (snd_pcm_hw_params_supports_audio_wallclock_ts(hwparams_p)) + printf("Playback relies on audio wallclock timestamps\n"); + else + printf("Playback relies on audio sample counter timestamps\n"); + snd_pcm_sw_params_alloca(&swparams_p); /* get the current swparams */ err = snd_pcm_sw_params_current(handle_p, swparams_p); @@ -131,6 +150,18 @@ int main(void) goto _exit; } + snd_pcm_hw_params_alloca(&hwparams_c); + /* get the current hwparams */ + err = snd_pcm_hw_params_current(handle_c, hwparams_c); + if (err < 0) { + printf("Unable to determine current hwparams_c: %s\n", snd_strerror(err)); + goto _exit; + } + if (snd_pcm_hw_params_supports_audio_wallclock_ts(hwparams_c)) + printf("Capture relies on audio wallclock timestamps\n"); + else + printf("Capture relies on audio sample counter timestamps\n"); + snd_pcm_sw_params_alloca(&swparams_c); /* get the current swparams */ err = snd_pcm_sw_params_current(handle_c, swparams_c); @@ -202,26 +233,43 @@ int main(void) frame_count_p += frames; #if defined(TRACK_PLAYBACK) - gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &avail_p, &delay_p); + gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &audio_tstamp_p, &avail_p, &delay_p); +#if defined(TRACK_SAMPLE_COUNTS) curr_count_p = frame_count_p - delay_p; /* written minus queued */ - printf("playback: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", - timediff(tstamp_p,trigger_tstamp_p), - (long long)round(((float)curr_count_p * 1000000000.0 / 48000.0)), - timediff(tstamp_p, trigger_tstamp_p) - (long long)round((double)curr_count_p * 1000000000.0 / 48000.0) + printf("playback: curr_count %lli driver count %lli, delta %lli\n", + (long long)curr_count_p * 1000000000LL / 48000 , + timestamp2ns(audio_tstamp_p), + (long long)curr_count_p * 1000000000LL / 48000 - timestamp2ns(audio_tstamp_p) + ); +#endif + + printf("playback: systime: %lli nsec, audio time %lli nsec, \tsystime delta %lli\n", + timediff(tstamp_p, trigger_tstamp_p), + timestamp2ns(audio_tstamp_p), + timediff(tstamp_p, trigger_tstamp_p) - timestamp2ns(audio_tstamp_p) ); #endif #if defined(TRACK_CAPTURE) - gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &avail_c, &delay_c); + gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &audio_tstamp_c, &avail_c, &delay_c); +#if defined(TRACK_SAMPLE_COUNTS) curr_count_c = frame_count_c + delay_c; /* read plus queued */ - printf("\t capture: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", - timediff(tstamp_c,trigger_tstamp_c), - (long long)round(((float)curr_count_c * 1000000000.0 / 48000.0)), - timediff(tstamp_c, trigger_tstamp_c) - (long long)round((double)curr_count_c * 1000000000.0 / 48000.0) + + printf("capture: curr_count %lli driver count %lli, delta %lli\n", + (long long)curr_count_c * 1000000000LL / 48000 , + timestamp2ns(audio_tstamp_c), + (long long)curr_count_c * 1000000000LL / 48000 - timestamp2ns(audio_tstamp_c) + ); +#endif + + printf("\t capture: systime: %lli nsec, audio time %lli nsec, \tsystime delta %lli\n", + timediff(tstamp_c, trigger_tstamp_c), + timestamp2ns(audio_tstamp_c), + timediff(tstamp_c, trigger_tstamp_c) - timestamp2ns(audio_tstamp_c) ); #endif -- 1.7.9.5

1 0

[alsa-devel] [PATCH 1/3] ALSA: hda - Remove dead GPIO code for VIA codec
by David Henningsson 08 Oct '12

08 Oct '12

>From what I can conclude all GPIO handling was removed in 2009. Remove dead code remnants. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_via.c | 47 --------------------------------------------- 1 file changed, 47 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 5a45a91..430cb33 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -299,7 +299,6 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) #define VIA_JACK_EVENT 0x20 #define VIA_HP_EVENT 0x01 -#define VIA_GPIO_EVENT 0x02 #define VIA_LINE_EVENT 0x03 enum { @@ -1685,50 +1684,6 @@ static void via_hp_automute(struct hda_codec *codec) via_line_automute(codec, present); } -static void via_gpio_control(struct hda_codec *codec) -{ - unsigned int gpio_data; - unsigned int vol_counter; - unsigned int vol; - unsigned int master_vol; - - struct via_spec *spec = codec->spec; - - gpio_data = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DATA, 0) & 0x03; - - vol_counter = (snd_hda_codec_read(codec, codec->afg, 0, - 0xF84, 0) & 0x3F0000) >> 16; - - vol = vol_counter & 0x1F; - master_vol = snd_hda_codec_read(codec, 0x1A, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_INPUT); - - if (gpio_data == 0x02) { - /* unmute line out */ - snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0], - PIN_OUT); - if (vol_counter & 0x20) { - /* decrease volume */ - if (vol > master_vol) - vol = master_vol; - snd_hda_codec_amp_stereo(codec, 0x1A, HDA_INPUT, - 0, HDA_AMP_VOLMASK, - master_vol-vol); - } else { - /* increase volume */ - snd_hda_codec_amp_stereo(codec, 0x1A, HDA_INPUT, 0, - HDA_AMP_VOLMASK, - ((master_vol+vol) > 0x2A) ? 0x2A : - (master_vol+vol)); - } - } else if (!(gpio_data & 0x02)) { - /* mute line out */ - snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0], 0); - } -} - /* unsolicited event for jack sensing */ static void via_unsol_event(struct hda_codec *codec, unsigned int res) @@ -1743,8 +1698,6 @@ static void via_unsol_event(struct hda_codec *codec, if (res == VIA_HP_EVENT || res == VIA_LINE_EVENT) via_hp_automute(codec); - else if (res == VIA_GPIO_EVENT) - via_gpio_control(codec); snd_hda_jack_report_sync(codec); } -- 1.7.9.5

2 3

[alsa-devel] [3.6-rc7] switcheroo race with Intel HDA...
by Daniel J Blueman 08 Oct '12

08 Oct '12

On my Macbook with a discrete Nvidia GPU, there is a race between selecting the integrated GPU and putting the discrete GPU into D3 [1], reliably causing a kernel oops [2]. Introducing a delay of ~1s between the calls prevents this. When the second 'OFF' write path executes, it looks like struct azx at card->private_data hasn't yet been allocated yet [3], so there is likely some locking missing. I'm happy to perform further testing and debug of course... Thanks, Daniel --- [1] echo IGD > /sys/kernel/debug/vgaswitcheroo/switch echo OFF > /sys/kernel/debug/vgaswitcheroo/switch --- [2] BUG: unable to handle kernel NULL pointer dereference at 0000000000000170 IP: [<ffffffffa01ba936>] azx_vs_set_state+0x26/0x178 [snd_hda_intel] PGD 259c26067 PUD 25a0fd067 PMD 0 Oops: 0000 [#1] SMP DEBUG_PAGEALLOC Modules linked in: snd_hda_codec_hdmi bnep rfcomm b43 joydev nfsd ssb nfs_acl auth_rpcgss binfmt_misc nfs lockd sunrpc uvcvideo bcm5974 videobuf2_core videobuf2_vmalloc videobuf2_memops coretemp kvm_intel snd_hda_codec_cirrus kvm applesmc input_polldev microcode bcma lpc_ich mfd_core mei snd_hda_intel(+) snd_hda_codec snd_hwdep snd_pcm snd_timer snd snd_page_alloc nls_iso8859_1 apple_gmux mac_hid apple_bl btrfs hid_apple sdhci_pci ghash_clmulni_intel tg3 sdhci i915 nouveau ttm drm_kms_helper hwmon mxm_wmi video CPU 2 Pid: 961, comm: sh Not tainted 3.6.0-rc7 #2 Apple Inc. MacBookPro10,1/Mac-C3EC7CD22292981F RIP: 0010:[<ffffffffa01ba936>] [<ffffffffa01ba936>] azx_vs_set_state+0x26/0x178 [snd_hda_intel] RSP: 0018:ffff880264271e48 EFLAGS: 00010286 RAX: 0000000000000000 RBX: ffff88025a2f5280 RCX: 0000000000000000 RDX: 0000000000000006 RSI: 0000000000000000 RDI: ffff880265479098 RBP: ffff880264271e68 R08: 2222222222222222 R09: 2222222222222222 R10: 0000000000000000 R11: 0000000000000000 R12: ffff880265479098 R13: 0000000000000000 R14: ffff880264271f50 R15: 0000000000000000 FS: 00007fa4fe183700(0000) GS:ffff88026f280000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000170 CR3: 00000002641a7000 CR4: 00000000001407e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 Process sh (pid: 961, threadinfo ffff880264270000, task ffff880264503a00) Stack: 2222222222222222 ffff88025a2f5280 0000000000000000 ffff880264271e98 ffff880264271e88 ffffffff812e83a7 ffff8802622835c0 0000000000000004 ffff880264271ef8 ffffffff812e89ac ffff88020a46464f ffff880264503a00 Call Trace: [<ffffffff812e83a7>] set_audio_state+0x67/0x70 [<ffffffff812e89ac>] vga_switcheroo_debugfs_write+0xbc/0x380 [<ffffffff81108773>] vfs_write+0xa3/0x160 [<ffffffff81108a75>] sys_write+0x45/0xa0 [<ffffffff815231a6>] system_call_fastpath+0x1a/0x1f Code: 00 00 00 00 00 55 48 89 e5 48 83 ec 20 4c 89 65 f0 4c 8d a7 98 00 00 00 4c 89 e7 48 89 5d e8 4c 89 6d f8 41 89 f5 e8 2a 35 13 e1 <48> 8b 98 70 01 00 00 0f b6 83 55 02 00 00 a8 08 75 34 45 85 ed RIP [<ffffffffa01ba936>] azx_vs_set_state+0x26/0x178 [snd_hda_intel] RSP <ffff880264271e48> CR2: 0000000000000170 --- [3] (gdb) list *(azx_vs_set_state+0x26) 0x2936 is in azx_vs_set_state (sound/pci/hda/hda_intel.c:2505). 2500 2501 static void azx_vs_set_state(struct pci_dev *pci, 2502 enum vga_switcheroo_state state) 2503 { 2504 struct snd_card *card = pci_get_drvdata(pci); 2505 struct azx *chip = card->private_data; 2506 bool disabled; 2507 2508 if (chip->init_failed) 2509 return; -- Daniel J Blueman

2 1

Re: [alsa-devel] Constant noise on HDA ALC275
by Julien Danjou 08 Oct '12

08 Oct '12

On Mon, Oct 08 2012, Takashi Iwai wrote: > At best, please give alsa-info.sh outputs at both working and > non-working cases. In that way, we can see the internal states more > exactly than ambiguous descriptions. Sure! Both attached [in gzip this time to pass the list]. The diff between background-noise and no-background-noise is: --- /tmp/alsa-info-background-noise.txt 2012-10-08 14:41:05.880816525 +0200 +++ /tmp/alsa-info-no-background-noise.txt 2012-10-08 14:42:24.700814459 +0200 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.61 !!################################ -!!Script ran on: Mon Oct 8 12:41:05 UTC 2012 +!!Script ran on: Mon Oct 8 12:42:24 UTC 2012 !!Linux Distribution @@ -198,7 +198,7 @@ Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 - Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] + Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x00 0x00] Connection: 5 0x18 0x19 0x1a 0x1b 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In -- Julien Danjou // Free Software hacker & freelance // http://julien.danjou.info

1 0

[alsa-devel] Constant noise on HDA ALC275
by Julien Danjou 08 Oct '12

08 Oct '12

Hi there, I've an HDA based chipset in a new Sony Vaio Z (Ivy Bridge). As sound as the system starts and the sound module are loaded, there's a constant noise (shhhhhhht) in the speakers. The noise change a bit if I toggle the mute button, but it's still there. This noise stops as sound as I plug something in the jack port. I've tried to play with HDA Analyzer¹, but I've no clue what to look for, so any hint appreciated. Some information: 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04) % aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC275 Analog [ALC275 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 % lsmod | grep snd snd_hda_codec_hdmi 30783 1 snd_hda_codec_realtek 50906 1 snd_hda_intel 26504 3 snd_hda_codec 83533 3 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel snd_hwdep 13186 1 snd_hda_codec snd_pcm 64080 3 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 12969 2 snd_pcm,snd_hda_intel snd_seq 45130 0 snd_seq_device 13176 1 snd_seq snd_timer 22917 2 snd_pcm,snd_seq snd 53077 15 snd_hda_codec_realtek,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_hda_codec,snd_hda_intel,snd_seq_device soundcore 13026 1 snd /proc/asound/card0/codec#0 ========================== /proc/asound/card0/codec#3 ========================== ¹ http://www.alsa-project.org/main/index.php/HDA_Analyzer -- Julien Danjou # Free Software hacker & freelance # http://julien.danjou.info

2 16

Re: [alsa-devel] 3.6.0 regression, distorted sound failure while using jack and hd-audio
by Daniel Mack 08 Oct '12

08 Oct '12

(cc alsa-devel) On 07.10.2012 21:34, Helge Hafting wrote: > My sound setup works fine with 3.5.4, but have serious problems with 3.6.0 > > It is sound through jack that breaks, simply using "aplay" directly with the > alsa device works. > > When I use jack, my setup works fine with 3.5.4. But boot into 3.6.0, > and I get badly distorted sound as if using an extremely overdone reverb effect. > And jack reports tons of xruns, several per second. > > This can be worked around by going from 256 to 512 frames/period in the jack > settings. But that also double the latency, which makes keyboard playing less > pleasant. > > It is not a case of 256 frames/period being (almost) too extreme for the machine. With kernel 3.5.4, > I can use 128 frames/period and get mostly ok sound, with only an occational > xrun (and a correseponding pop in the sound). So I use 256 because that's solid. > But with 3.6.0, not even 256 is enough. And it is not an occational pop, but > awful distortion all the time. > > I don't really know what has changed from 3.5.4 to 3.6.0. The hd-audio driver? > Something in alsa? Or the scheduling? > > Machine: Dell precision M6600 laptop, 8GB, 2.2GHz quad-core i7. > Usually at 800MHz to save power, this is enough for making sound. > Kernels: 64-bit 3.5.4 or 3.6.0 > Distro: funtoo > > Audio according to lspci: > 00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 04) > > Please tell if there is anything I should try. I can test other kernel versions > or try patches. > > Helge Hafting > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

2 1

[alsa-devel] [PATCH] ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
by David Henningsson 08 Oct '12

08 Oct '12

In case there is one "Headphone Jack" and one "Dock Headphone Jack", one of them will get an index, even though that is not needed. This patch fixes that issue. BugLink: https://bugs.launchpad.net/bugs/1060729 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 49 +++++++++++++++++++++++++++++---------- 1 file changed, 37 insertions(+), 12 deletions(-) Alsa-info at: https://launchpadlibrarian.net/118211803/AlsaInfo diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index a98e25e..4ec6dc8 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -498,6 +498,38 @@ static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins, return channel_sfx[i]; } +static const char *check_output_pfx(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + int attr = snd_hda_get_input_pin_attr(def_conf); + + /* check the location */ + switch (attr) { + case INPUT_PIN_ATTR_DOCK: + return "Dock "; + case INPUT_PIN_ATTR_FRONT: + return "Front "; + } + return ""; +} + +static int get_hp_label_index(struct hda_codec *codec, hda_nid_t nid, + const hda_nid_t *pins, int num_pins) +{ + int i, j, idx = 0; + + const char *pfx = check_output_pfx(codec, nid); + + i = find_idx_in_nid_list(nid, pins, num_pins); + if (i < 0) + return -1; + for (j = 0; j < i; j++) + if (pfx == check_output_pfx(codec, pins[j])) + idx++; + + return idx; +} + static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, const struct auto_pin_cfg *cfg, const char *name, char *label, int maxlen, @@ -505,20 +537,13 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, { unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); int attr = snd_hda_get_input_pin_attr(def_conf); - const char *pfx = "", *sfx = ""; + const char *pfx, *sfx = ""; /* handle as a speaker if it's a fixed line-out */ if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) name = "Speaker"; - /* check the location */ - switch (attr) { - case INPUT_PIN_ATTR_DOCK: - pfx = "Dock "; - break; - case INPUT_PIN_ATTR_FRONT: - pfx = "Front "; - break; - } + pfx = check_output_pfx(codec, nid); + if (cfg) { /* try to give a unique suffix if needed */ sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs, @@ -528,8 +553,8 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, indexp); if (!sfx) { /* don't add channel suffix for Headphone controls */ - int idx = find_idx_in_nid_list(nid, cfg->hp_pins, - cfg->hp_outs); + int idx = get_hp_label_index(codec, nid, cfg->hp_pins, + cfg->hp_outs); if (idx >= 0) *indexp = idx; sfx = ""; -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ALSA: aloop - add locking to timer access
by Omair Mohammed Abdullah 08 Oct '12

08 Oct '12

When the loopback timer handler is running, calling del_timer() (for STOP trigger) will not wait for the handler to complete before deactivating the timer. The timer gets rescheduled in the handler as usual. Then a subsequent START trigger will try to start the timer using add_timer() with a timer pending leading to a kernel panic. Serialize the calls to add_timer() and del_timer() using a spin lock to avoid this. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/drivers/aloop.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 5a34355..0fe6d64 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -120,6 +120,7 @@ struct loopback_pcm { unsigned int last_drift; unsigned long last_jiffies; struct timer_list timer; + spinlock_t timer_lock; }; static struct platform_device *devices[SNDRV_CARDS]; @@ -170,6 +171,7 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) unsigned long tick; unsigned int rate_shift = get_rate_shift(dpcm); + spin_lock(&dpcm->timer_lock); if (rate_shift != dpcm->pcm_rate_shift) { dpcm->pcm_rate_shift = rate_shift; dpcm->period_size_frac = frac_pos(dpcm, dpcm->pcm_period_size); @@ -182,12 +184,15 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) tick = (tick + dpcm->pcm_bps - 1) / dpcm->pcm_bps; dpcm->timer.expires = jiffies + tick; add_timer(&dpcm->timer); + spin_unlock(&dpcm->timer_lock); } static inline void loopback_timer_stop(struct loopback_pcm *dpcm) { + spin_lock(&dpcm->timer_lock); del_timer(&dpcm->timer); dpcm->timer.expires = 0; + spin_unlock(&dpcm->timer_lock); } #define CABLE_VALID_PLAYBACK (1 << SNDRV_PCM_STREAM_PLAYBACK) @@ -667,6 +672,7 @@ static int loopback_open(struct snd_pcm_substream *substream) dpcm->substream = substream; setup_timer(&dpcm->timer, loopback_timer_function, (unsigned long)dpcm); + spin_lock_init(&dpcm->timer_lock); cable = loopback->cables[substream->number][dev]; if (!cable) { -- 1.7.4.1

3 3

[alsa-devel] [PATCH] hda-emu: Add check for unnecessary indices of kcontrols to testsuite
by David Henningsson 08 Oct '12

08 Oct '12

--- tester/runner.py | 29 +++++++++++++++++++++++------ 1 file changed, 23 insertions(+), 6 deletions(-) diff --git a/tester/runner.py b/tester/runner.py index 30118ce..1e7ecf9 100644 --- a/tester/runner.py +++ b/tester/runner.py @@ -25,7 +25,9 @@ class ControlInfo(): self.runner = runner carr = list_info.split(":") self.numid = carr[1] - self.name = carr[2] + self.name = carr[2][1:] + self.index, _ = tuple(carr[3].split(" ")) + self.index = int(self.index) def add_info(self, get_info): minmax_regex = re.compile("MIN/MAX: (\d+)/(\d+),\s+VAL:(( \\[\d+\\])+)") @@ -222,13 +224,28 @@ class HdaEmuRunner(): if (int(a) != int(b)): self.add_error("Tried to set " + c.name + " to " + str([int(x) for x in values]) + ", but got " + str(c.values) + " instead", "Error") - def run_kcontrol_test(self): - minmax_regex = re.compile("MIN/MAX: (\d+)/(\d+),\s+VAL:(( \\[\d+\\])+)") - val_regex = re.compile(" \\[(\d+)\\]") + def check_kcontrol_indices(self, cinfo): + d = {} + for c in cinfo: + if not c.name in d: + d[c.name] = [] + d[c.name].append(c.index) + + for k, v in d.iteritems(): + j = 0 + for i in sorted(v): + if i > j: + self.add_error("Unneccesary index on control '" + k + "' (" + str(i) + ")", "Error") +# Commented out - gives false positives for 'Playback Channel Map' +# elif i < j: +# self.add_error("Duplicate control '" + k + "' (" + str(i) + ")", "Error") + j = i+1 + def run_kcontrol_test(self): controls = self.run_command("list") - for control in controls: - c = ControlInfo(self, control) + cinfo = [ControlInfo(self, control) for control in controls] + self.check_kcontrol_indices(cinfo) + for c in cinfo: c.add_info(self.run_command("get " + c.numid)) if "minval" in c.__dict__: minval = c.minval -- 1.7.9.5

2 1

[alsa-devel] [PATCH] hda-emu: Improve directory/path handling for testsuite
by David Henningsson 08 Oct '12

08 Oct '12

There were some shortcomings when running the test scripts from a random directory. Also added functionality to test an entire directory tree at the same time. --- tester/runner.py | 9 ++++++++- tester/summary.py | 28 ++++++++++++++++++++++++---- 2 files changed, 32 insertions(+), 5 deletions(-) With this improvement we can run it over the entire alsa-info database. The initial result is 619 of 5911 (10%) failing, mostly due to invalid verbs sent (no input/output-amp, invalid amp index, setting *_EN without caps, etc). diff --git a/tester/runner.py b/tester/runner.py index b585ad3..30118ce 100644 --- a/tester/runner.py +++ b/tester/runner.py @@ -44,10 +44,13 @@ class ControlInfo(): class HdaEmuRunner(): def __init__(self): - self.child_args = "../hda-emu -M -F" + import os.path + hda_emu_path = os.path.join(os.path.dirname(os.path.realpath(__file__)),"../hda-emu") + self.child_args = '"' + hda_emu_path + '" -M -F' self.alsa_info = "/proc/asound/card0/codec#0" self.errors = 0 self.warnings = 0 + self.fatals = 0 self.print_errors = False self.comm_log_enabled = False self.last_command = None @@ -81,6 +84,8 @@ class HdaEmuRunner(): self.warnings += 1 else: self.errors += 1 + if severity == "Fatal": + self.fatals += 1 if self.print_errors: if not self.last_command_printed: if self.last_command: @@ -89,6 +94,8 @@ class HdaEmuRunner(): print "Encountered during initial parsing:" self.last_command_printed = True print " ", message + if self.fatals > 0: + raise Exception(message) def check_stdout(self): s = os.read(self.child.stdout.fileno(), 65536) diff --git a/tester/summary.py b/tester/summary.py index 6e94c9b..0f9fbf3 100755 --- a/tester/summary.py +++ b/tester/summary.py @@ -18,6 +18,11 @@ # You should have received a copy of the GNU General Public License along # with this program. If not, see <http://www.gnu.org/licenses/>. +def defaultpath(s): + import os.path + q = os.path.dirname(os.path.realpath(__file__)) + return os.path.join(q, s) + def main(): import os import os.path @@ -26,18 +31,28 @@ def main(): import argparse parser = argparse.ArgumentParser(description='Hda-emu automated test wrapper.') parser.add_argument('--verbose', '-v', action='count') + parser.add_argument('--directory', '-d', action='store', default=defaultpath("../codecs/canonical/")) + parser.add_argument('--recursive', '-r', action="store_true", default=False) + parser_dict = parser.parse_args() verbose = parser_dict.verbose + directory = parser_dict.directory - os.chdir(os.path.dirname(os.path.realpath(__file__))) - directory = "../codecs/canonical/" - files = os.listdir(directory) + if parser_dict.recursive: + files = [os.path.join(root, filename) for root, _, filenames in os.walk(directory) for filename in filenames] + else: + files = os.listdir(directory) + files.sort() successes = 0 fails = 0 warnings = 0 errors = 0 + index = 0 for f in files: + index += 1 + if verbose > 2: + print '[{0}/{1}]: Testing {2}'.format(index, len(files), f) try: r = runner.HdaEmuRunner() r.set_alsa_info_file(os.path.join(directory, f)) @@ -51,7 +66,12 @@ def main(): print '{0} errors, {1} warnings. ({2})'.format(r.errors, r.warnings, f) else: successes += 1 - except: + except KeyboardInterrupt: + import sys + sys.exit(1) + except Exception as e: + if verbose > 0: + print 'Fatal error for {0}: {1}'.format(f, e) errors += 1 fails += 1 -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - fix indices on boost volume on Conexant
by David Henningsson 08 Oct '12

08 Oct '12

After the recent patch "ALSA: hda - use both input paths on Conexant auto parser" suddenly we can have more than one "Mic Boost", this happened on Acer Aspire One 722. Therefore we must add the possibility to put an index on this "Mic Boost" just as we do for the other "Mic Boost" earlier in the same function. BugLink: https://bugs.launchpad.net/bugs/1059523 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c03d3b8..c11ad8a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4273,7 +4273,7 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, if (get_wcaps(codec, mux) & AC_WCAP_OUT_AMP) { spec->imux_info[idx].boost = mux; - return cx_auto_add_volume(codec, label, " Boost", 0, + return cx_auto_add_volume(codec, label, " Boost", cidx, mux, HDA_OUTPUT); } return 0; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix hang caused by race during suspend.
by Dylan Reid 08 Oct '12

08 Oct '12

There was a race condition when the system suspends while hda_power_work is running in the work queue. If system suspend (snd_hda_suspend) happens after the work queue releases power_lock but before it calls hda_call_codec_suspend, codec_suspend runs with power_on=0, causing the codec to power up for register reads, and hanging when it calls cancel_delayed_work_sync from the running work queue. The call chain from the work queue will look like this: hda_power_work <<- power_on = 1, unlock, then power_on cleard by suspend hda_call_codec_suspend hda_set_power_state snd_hda_codec_read codec_exec_verb snd_hda_power_up snd_hda_power_save __snd_hda_power_up cancel_delayed_work_sync <<-- cancelling executing wq Fix this by waiting for the work queue to finish before starting suspend if suspend is not happening on the work queue. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_codec.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 960800b..c0ab72c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3612,7 +3612,7 @@ static inline void hda_exec_init_verbs(struct hda_codec *codec) {} * call suspend and power-down; used both from PM and power-save * this function returns the power state in the end */ -static unsigned int hda_call_codec_suspend(struct hda_codec *codec) +static unsigned int hda_call_codec_suspend(struct hda_codec *codec, bool in_wq) { unsigned int state; @@ -3620,7 +3620,9 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); state = hda_set_power_state(codec, AC_PWRST_D3); - cancel_delayed_work(&codec->power_work); + /* Cancel delayed work if we aren't currently running from it. */ + if (!in_wq) + cancel_delayed_work_sync(&codec->power_work); spin_lock(&codec->power_lock); snd_hda_update_power_acct(codec); trace_hda_power_down(codec); @@ -4478,7 +4480,7 @@ static void hda_power_work(struct work_struct *work) } spin_unlock(&codec->power_lock); - state = hda_call_codec_suspend(codec); + state = hda_call_codec_suspend(codec, true); codec->pm_down_notified = 0; if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { codec->pm_down_notified = 1; @@ -5127,7 +5129,7 @@ int snd_hda_suspend(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) - hda_call_codec_suspend(codec); + hda_call_codec_suspend(codec, false); } return 0; } -- 1.7.12.146.g16d26b1

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix detection of ALC271X codec
by Herton Ronaldo Krzesinski 08 Oct '12

08 Oct '12

In commit af741c1 ("ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup"), alc_auto_parse_customize_define was moved after detection of ALC271X. The problem is that detection of ALC271X relies on spec->cdefine.platform_type, and it's set on alc_auto_parse_customize_define. Move the alc_auto_parse_customize_define and its required fixup setup before the block doing the ALC271X and other codec setup. BugLink: https://bugs.launchpad.net/bugs/1006690 Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) It looks safe to move the functions before the block "if (codec->vendor_id == 0x10ec0269) {", as nothing on the fixups seems to depend on what's done there. But if wanted we could just move the detection/codec rename of ALC271X below too. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c4c4b01..3b187e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6273,6 +6273,12 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; + alc_pick_fixup(codec, alc269_fixup_models, + alc269_fixup_tbl, alc269_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + + alc_auto_parse_customize_define(codec); + if (codec->vendor_id == 0x10ec0269) { spec->codec_variant = ALC269_TYPE_ALC269VA; switch (alc_get_coef0(codec) & 0x00f0) { @@ -6300,12 +6306,6 @@ static int patch_alc269(struct hda_codec *codec) alc269_fill_coef(codec); } - alc_pick_fixup(codec, alc269_fixup_models, - alc269_fixup_tbl, alc269_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - - alc_auto_parse_customize_define(codec); - /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); if (err < 0) -- 1.7.10.4

3 3

[alsa-devel] [PATCH 1/1] ALSA: CA0132 - Ensure SCP packet is little-endian format
by Ian Minett 08 Oct '12

08 Oct '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi Takashi, Great - good to hear the previous DSP loader patches are now living in the hda-ca0132-dsp branch. This patch is based on that branch, and addresses the SCP packet endianness. >> The firmware does need to be reloaded when the chip loses power, so it >> doesn't survive S3. >OK, we need suspend/resume hooks, too. Do we need to do anything specific to handle this? The firmware is already restored upon S3 by the current caching code. Thanks, - Ian --- The DSP firmware expects the SCP data to be in little-endian format regardless of the host architecture. Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2fd3121..4d7cda4 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -28,6 +28,7 @@ #include <linux/mutex.h> #include <linux/module.h> #include <linux/firmware.h> +#include <asm/unaligned.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -701,9 +702,13 @@ static int dspio_send(struct hda_codec *codec, unsigned int reg, unsigned int res; int retry = 50; + /* Handling for Big Endian machine */ + unsigned int val; + put_unaligned_le32(data, &val); + /* send bits of data specified by reg to dsp */ do { - res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data); + res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, val); if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY)) return res; } while (--retry); -- 1.7.4.1

2 1

[alsa-devel] [PATCH 0/10] ALSA: Add DSP firmware loader (v3)
by Ian Minett 08 Oct '12

08 Oct '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi, thanks for the response. I've resubmitted the patch series with updates to the ones that needed altering based on the feedback. The MODULE_FIRMWARE line has been shifted to the firmware loading patch, DSP download is now guarded by the config switch and code changes that were bundled in with the previous comments patch have moved to a new patch. Also, hda_stream_format has been removed. Regarding the SCP packet endianness, the DSP image requires data to be sent as little-endian, so we may need to swizzle data being sent to the DSP before the transfer. Is there a recommended or standard way of detecting and handling the case of running on big-endian architectures? >Does the firmware on chip survive after S3? The firmware does need to be reloaded when the chip loses power, so it doesn't survive S3. Thanks very much, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- 1: - memalloc.h - pcm.h - pcm_memory.c - sgbuf.c Include Takashi's patch: Make snd_sgbuf_get_{ptr|addr}() available for non-SG cases. Passing struct snd_dma_buffer pointer instead, so that they work no matter whether real SG buffer is used or not. 2: - hda_intel.c - hda_codec.h Include Takashi's code: Pass DMA buffer pointers in calls to setup_bdle(). Add DSP loader callback routines to controller. Add new DSP loader switch to Kconfig to enable DSP firmware loading. 3: - patch_ca0132.c - ca0132_regs.h Add DSP register definitions header file 4 (updated): - patch_ca0132.c Add DSP firmware enums and defs to CA0132 codec MODULE_FIRMWARE not declared in this patch, moved into patch #5 5 (updated): - patch_ca0132.c Add calls to new DSP loader system to transfer firmware binary to the hardware. Add chip read/write routines, DSP I/O, SCP packet format helper functions and transfer DMA management. Add MODULE_FIRMWARE line for DSP firmware binary Protect DSP download using CONFIG_SND_HDA_DSP_LOADER switch 6: - patch_ca0132.c Add DSP firmware caching to CA0132 codec 7 (updated): - patch_ca0132.c Add comments and descriptions to functions. Previous code changes moved to patch #9 8: - hda_codec.h Change return value for load_dsp_prepare to -ENOSYS in case where DSP loader routines are not available. 9 (new): - patch_ca0132.c Merge chipio write address functions and update timeout mechanism in dsp_write_wait(). 10 (new): - patch_ca0132.c Remove unnecessary struct hda_stream_format and references to it.

3 15

[alsa-devel] [PATCH] ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
by Felix Kaechele 08 Oct '12

08 Oct '12

The Lenovo IdeaPad U310 has an internal mic where the right channel is phase inverted. Signed-off-by: Felix Kaechele <felix(a)fetzig.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e22a8f..784017e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4462,6 +4462,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), {} }; -- 1.7.11.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - make a generic unsol event handler
by David Henningsson 08 Oct '12

08 Oct '12

Moving towards less duplication of code between codecs - this patch takes some of the common code of unsol event handling and makes it generic. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 32 ++++++++++++++++++++++++++++++-- sound/pci/hda/hda_jack.h | 9 +++++++++ 2 files changed, 39 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index c9333c9..5c690cb 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -192,8 +192,9 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect); /** * snd_hda_jack_detect_enable - enable the jack-detection */ -int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, - unsigned char action) +int snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, + unsigned char action, + hda_jack_callback cb) { struct hda_jack_tbl *jack = snd_hda_jack_tbl_new(codec, nid); if (!jack) @@ -203,10 +204,19 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, jack->jack_detect = 1; if (action) jack->action = action; + if (cb) + jack->callback = cb; return snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | jack->tag); } +EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable_callback); + +int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, + unsigned char action) +{ + return snd_hda_jack_detect_enable_callback(codec, nid, action, NULL); +} EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); /** @@ -411,3 +421,21 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctls); + +void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct hda_jack_tbl *event; + int tag = (res >> AC_UNSOL_RES_TAG_SHIFT) & 0x7f; + + event = snd_hda_jack_tbl_get_from_tag(codec, tag); + if (!event) + return; + event->jack_dirty = 1; + + if (event->callback) + event->callback(codec, event); + + snd_hda_jack_report_sync(codec); +} +EXPORT_SYMBOL_HDA(snd_hda_jack_unsol_event); + diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index a9803da..af8dd47 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -13,12 +13,16 @@ #define __SOUND_HDA_JACK_H struct auto_pin_cfg; +struct hda_jack_tbl; + +typedef void (*hda_jack_callback) (struct hda_codec *, struct hda_jack_tbl *); struct hda_jack_tbl { hda_nid_t nid; unsigned char action; /* event action (0 = none) */ unsigned char tag; /* unsol event tag */ unsigned int private_data; /* arbitrary data */ + hda_jack_callback callback; /* jack-detection stuff */ unsigned int pin_sense; /* cached pin-sense value */ unsigned int jack_detect:1; /* capable of jack-detection? */ @@ -61,6 +65,10 @@ void snd_hda_jack_set_dirty_all(struct hda_codec *codec); int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, unsigned char action); +int snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, + unsigned char action, + hda_jack_callback cb); + u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); @@ -74,5 +82,6 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, void snd_hda_jack_report_sync(struct hda_codec *codec); +void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res); #endif /* __SOUND_HDA_JACK_H */ -- 1.7.9.5

2 2

Re: [alsa-devel] Internal Microphone in Lenovo P580 not working with alsa 1.0.25
by Raymond Yau 08 Oct '12

08 Oct '12

2012-10-5 下午3:39 於 "mefis" <mefistof(a)gmx.com> 寫道： > > On 10/05/2012 02:09 AM, Raymond Yau wrote: >> >> 2012-10-5 下午2:20 於 "mefis" <mefistof(a)gmx.com> 寫道： >>> >>> Hi, >>> >>> Mic not working in Lenovo P580 with alsa 1.0.25 >>> >>> Alsa Information script output: >>> >>> http://www.alsa-project.org/db/?f=001523ed61bc76fb5b2ad0d3c354c043f89e5a33 >>> >>> Hope it helps. >>> >>> >>> >>> !!################################ >>> !!ALSA Information Script v 0.4.61 >>> !!################################ >>> >>> !!Script ran on: Fri Oct 5 05:56:38 UTC 2012 >>> >>> >>> !!Linux Distribution >>> !!------------------ >>> >>> Ubuntu 12.04.1 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu >> >> 12.04.1 LTS" >>> >>> >>> !!DMI Information >>> !!--------------- >>> >>> Manufacturer: LENOVO >>> Product Name: Lenovo IdeaPad P580 >>> Product Version: Lenovo IdeaPad P580 >>> Firmware Version: 5ECN33WW(V2.03) >>> >>> >>> !!Kernel Information >>> !!------------------ >>> >>> Kernel release: 3.2.0-31-generic >>> Operating System: GNU/Linux >>> Architecture: x86_64 >>> Processor: x86_64 >>> SMP Enabled: Yes >>> >>> >>> !!ALSA Version >>> !!------------ >>> >>> Driver version: 1.0.25 >>> Library version: 1.0.16 >>> Utilities version: 1.0.25 >> >> is this an upgrade since alsa-lib is quite old ? > > sorry, this was an old log. The newest is here: http://www.alsa-project.org/db/?f=08ae62e61ccc89d92a6154d4e5907b4f676c164a > >> >> it is strange that unsolicied event enabled at fixed mic >> instead of mic jack ? >> > > !!################################ > !!ALSA Information Script v 0.4.61 > !!################################ > > !!Script ran on: Fri Oct 5 07:34:06 UTC 2012 > > > > !!Linux Distribution > !!------------------ > > Ubuntu 12.04.1 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 12.04.1 LTS" > > > !!DMI Information > !!--------------- > > Manufacturer: LENOVO > Product Name: Lenovo IdeaPad P580 > Product Version: Lenovo IdeaPad P580 > Firmware Version: 5ECN33WW(V2.03) > > > !!Kernel Information > !!------------------ > > Kernel release: 3.2.0-31-generic > Operating System: GNU/Linux > Architecture: x86_64 > Processor: x86_64 > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.25 > Library version: 1.0.25 > > Utilities version: 1.0.25 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > > > !!Sound Servers on this system > !!---------------------------- > > Pulseaudio: > Installed - Yes (/usr/bin/pulseaudio) > Running - Yes > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [PCH ]: HDA-Intel - HDA Intel PCH > HDA Intel PCH at 0xe0610000 irq 46 > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation Panther Point High Definition Audio Controller (rev 04) > > > !!Advanced information - PCI Vendor/Device/Subsystem ID's > !!------------------------------------------------------- > > 00:1b.0 0403: 8086:1e20 (rev 04) > Subsystem: 17aa:3977 > > > !!Modprobe options (Sound related) > !!-------------------------------- > > snd-atiixp-modem: index=-2 > snd-intel8x0m: index=-2 > snd-via82xx-modem: index=-2 > snd-usb-audio: index=-2 > snd-usb-caiaq: index=-2 > snd-usb-ua101: index=-2 > snd-usb-us122l: index=-2 > snd-usb-usx2y: index=-2 > snd-cmipci: mpu_port=0x330 fm_port=0x388 > snd-pcsp: index=-2 > snd-usb-audio: index=-2 > snd-hda-intel: model=basic try model=auto http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=blob_plain;f=Do… Conexant 5066 ============= laptopBasic Laptop config (default) hp-laptopHP laptops, e g G60 asusAsus K52JU, Lenovo G560 dell-laptopDell laptops dell-vostroDell Vostro olpc-xo-1_5OLPC XO 1.5 ideapad Lenovo IdeaPad U150 thinkpadLenovo Thinkpad the model "auto" is not listed in the documentation http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=blob_plain;f=so… static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_LAPTOP]= "laptop", [CXT5066_DELL_LAPTOP]= "dell-laptop", [CXT5066_OLPC_XO_1_5]= "olpc-xo-1_5", [CXT5066_DELL_VOSTRO]= "dell-vostro", [CXT5066_IDEAPAD]= "ideapad", [CXT5066_THINKPAD]= "thinkpad", [CXT5066_ASUS]= "asus", [CXT5066_HP_LAPTOP]= "hp-laptop", [CXT5066_AUTO]= "auto", }; > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: Conexant CX20590 > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x14f1506e > Subsystem Id: 0x17aac02d > Revision Id: 0x100003 > Node 0x1a [Pin Complex] wcaps 0x400481: Stereo > Pincap 0x00001324: IN Detect > Vref caps: HIZ 50 80 > Pin Default 0x04a11030: [Jack] Mic at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > > Power states: D0 D1 D2 D3 D3cold EPSS > Power: setting=D0, actual=D0 > Node 0x1b [Pin Complex] wcaps 0x400581: Stereo > Pincap 0x00011334: IN OUT EAPD Detect > Vref caps: HIZ 50 80 > EAPD 0x2: EAPD > Pin Default 0x95a70120: [Fixed] Mic at Int Top > Conn = Analog, Color = Unknown > DefAssociation = 0x2, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: VREF_HIZ > Unsolicited: tag=38, enabled=1 > Power states: D0 D1 D2 D3 D3cold EPSS > Power: setting=D0, actual=D0 > Connection: 2 > 0x10* 0x11 >

1 0

Re: [alsa-devel] sound/soc/codecs/wm2200.c:2082:2: error: unknown field 'num_ranges' specified in initializer
by Mark Brown 06 Oct '12

06 Oct '12

On Sat, Oct 06, 2012 at 09:23:04AM +0800, Fengguang Wu wrote: > All error/warnings: > sound/soc/codecs/wm2200.c:2082:2: error: unknown field 'num_ranges' specified in initializer Not a real error (nor is anything else like this in my trees right now), all the branches will be rebased after the merge window which will pull in the dependencies.

1 0

Re: [alsa-devel] [PATCH 0/2] Convert davinci ASoC to genalloc SRAM
by Matt Porter 05 Oct '12

05 Oct '12

On Thu, Oct 04, 2012 at 06:15:43AM -0400, Matt Porter wrote: > This series converts davinci ASoC to use genalloc and enables > that support on DA850. It applies on top of the uio_pruss/genalloc > series [1] which allows DaVinci to provide a gen_pool via pdata > for driver use. > > I've tested this on the AM180x EVM. Note that prior to this, > the SRAM paths in the driver were completely unused. I've only > enabled ping-pong buffering on the platform I can test as it's > best to allow those with DM644x and similar platforms to set the > playback/capture sram size to something that's known to work > for them. > > [1] http://www.spinics.net/lists/arm-kernel/msg198854.html Tested again on top of v4 of the uio_pruss/genalloc series http://www.spinics.net/lists/arm-kernel/msg199255.html on next-20121005 since there's some conflicts falling out of of arm-soc and asoc/sound header move patches on mainline atm. -Matt

1 0

[alsa-devel] [PATCH 1/7] ASoC: wm2200: Convert to devm APIs
by Mark Brown 05 Oct '12

05 Oct '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 39 ++++++++++++++------------------------- 1 file changed, 14 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index eab64a1..6ca9bcc 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2011,7 +2011,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, wm2200->dev = &i2c->dev; init_completion(&wm2200->fll_lock); - wm2200->regmap = regmap_init_i2c(i2c, &wm2200_regmap); + wm2200->regmap = devm_regmap_init_i2c(i2c, &wm2200_regmap); if (IS_ERR(wm2200->regmap)) { ret = PTR_ERR(wm2200->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2027,8 +2027,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, for (i = 0; i < ARRAY_SIZE(wm2200->core_supplies); i++) wm2200->core_supplies[i].supply = wm2200_core_supply_names[i]; - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); @@ -2044,8 +2045,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, } if (wm2200->pdata.ldo_ena) { - ret = gpio_request_one(wm2200->pdata.ldo_ena, - GPIOF_OUT_INIT_HIGH, "WM2200 LDOENA"); + ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.ldo_ena, + GPIOF_OUT_INIT_HIGH, + "WM2200 LDOENA"); if (ret < 0) { dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", wm2200->pdata.ldo_ena, ret); @@ -2055,8 +2057,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, } if (wm2200->pdata.reset) { - ret = gpio_request_one(wm2200->pdata.reset, - GPIOF_OUT_INIT_HIGH, "WM2200 /RESET"); + ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.reset, + GPIOF_OUT_INIT_HIGH, + "WM2200 /RESET"); if (ret < 0) { dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", wm2200->pdata.reset, ret); @@ -2166,23 +2169,16 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, err_pm_runtime: pm_runtime_disable(&i2c->dev); err_reset: - if (wm2200->pdata.reset) { + if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); - gpio_free(wm2200->pdata.reset); - } err_ldo: - if (wm2200->pdata.ldo_ena) { + if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); - gpio_free(wm2200->pdata.ldo_ena); - } err_enable: regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), wm2200->core_supplies); err_core: - regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); err_regmap: - regmap_exit(wm2200->regmap); err: return ret; } @@ -2194,17 +2190,10 @@ static __devexit int wm2200_i2c_remove(struct i2c_client *i2c) snd_soc_unregister_codec(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm2200); - if (wm2200->pdata.reset) { + if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); - gpio_free(wm2200->pdata.reset); - } - if (wm2200->pdata.ldo_ena) { + if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); - gpio_free(wm2200->pdata.ldo_ena); - } - regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); - regmap_exit(wm2200->regmap); return 0; } -- 1.7.10.4

1 6

[alsa-devel] [PATCH] ASoC: have unique stream_names in codecs
by Peter Meerwald 05 Oct '12

05 Oct '12

From: Peter Meerwald <p.meerwald(a)bct-electronic.com> for example, there are two 'Capture' stream_names in twl4030, resulting in twl4030-codec twl4030-codec: ASoC: Failed to create Capture debugfs file with debug enabled (fails to create two sysfs entries named 'Capture' under /sys/kernel/debug/asoc/omap3beagle/twl4030-codec/dapm this patch tries to avoid duplicate stream_names Signed-off-by: Peter Meerwald <p.meerwald(a)bct-electronic.com> --- sound/soc/codecs/stac9766.c | 4 ++-- sound/soc/codecs/twl4030.c | 4 ++-- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/uda1380.c | 4 ++-- sound/soc/codecs/wm8753.c | 4 ++-- 5 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 33c0f3d..3379119 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -300,14 +300,14 @@ static struct snd_soc_dai_driver stac9766_dai[] = { /* stream cababilities */ .playback = { - .stream_name = "stac9766 analog", + .stream_name = "stac9766 playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SND_SOC_STD_AC97_FMTS, }, .capture = { - .stream_name = "stac9766 analog", + .stream_name = "stac9766 capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 391fcfc..55582db 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2172,7 +2172,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .formats = TWL4030_FORMATS, .sig_bits = 24,}, .capture = { - .stream_name = "Capture", + .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 4, .rates = TWL4030_RATES, @@ -2189,7 +2189,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .capture = { - .stream_name = "Capture", + .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index c084c54..6f3a9cf 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1046,7 +1046,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { { .name = "twl6040-ul", .capture = { - .stream_name = "Capture", + .stream_name = "Capture UL", .channels_min = 1, .channels_max = 2, .rates = TWL6040_RATES, diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 2502214..d780fe7 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -671,7 +671,7 @@ static struct snd_soc_dai_driver uda1380_dai[] = { { /* playback only - dual interface */ .name = "uda1380-hifi-playback", .playback = { - .stream_name = "Playback", + .stream_name = "Playback Only", .channels_min = 1, .channels_max = 2, .rates = UDA1380_RATES, @@ -682,7 +682,7 @@ static struct snd_soc_dai_driver uda1380_dai[] = { { /* capture only - dual interface*/ .name = "uda1380-hifi-capture", .capture = { - .stream_name = "Capture", + .stream_name = "Capture Only", .channels_min = 1, .channels_max = 2, .rates = UDA1380_RATES, diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 13bff87..783ac20 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1400,7 +1400,7 @@ static struct snd_soc_dai_driver wm8753_dai[] = { .formats = WM8753_FORMATS }, .capture = { /* dummy for fast DAI switching */ - .stream_name = "Capture", + .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, @@ -1418,7 +1418,7 @@ static struct snd_soc_dai_driver wm8753_dai[] = { .formats = WM8753_FORMATS, }, .capture = { - .stream_name = "Capture", + .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, -- 1.7.9.5

2 1

Re: [alsa-devel] issue of shdma-base library
by Guennadi Liakhovetski 05 Oct '12

05 Oct '12

Mark, could you take this patch or would you like me to resend it with "subject" in the actual mail subject line or does it also work from the mail body? Thanks Guennadi On Thu, 4 Oct 2012, Do Q.Thang wrote: > Hi Guennadi-san > > Thank you for your hard work. > > On 10/03/2012 09:33 PM, Guennadi Liakhovetski wrote: > > Subject: [PATCH] ASoC: fsi: don't reschedule DMA from an atomic context > > > > shdma doesn't support transfer re-scheduling or triggering from callbacks > > or from atomic context. The fsi driver issues DMA transfers from a tasklet > > context, which is a bug. To fix it convert tasklet to a work. > > > > Reported-by: Do Q.Thang <dq-thang(a)jinso.co.jp> > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > > --- > > > > Thang-san, Please, check, whether the attached patch fixes your issues. It > > seems to fix them for me. > > This patch also fix them for me. > > Thanks > --- > Thang > > > Mark, please, wait with committing this patch until the guys confirm, it > > really helps. It then will also have to go to 3.6-stable. > > > > Thanks > > Guennadi > > > > On Tue, 2 Oct 2012, Do Q.Thang wrote: > > > > > Hi Guennadi-san, > > > > > > I'm testing uptream kernel (v3.6-rc3 ~) on the kzm-a9-gt & > > > armadillo-800eva > > > board. > > > I found 2 issues with the sh-dma driver while testing audio playback. > > > > > > issue 1: this log appeared while playback with the aplay, then aplay was > > > stopped. > > > > > > ------- log ----------------------------- > > > # aplay audio_s16le_44100.wav > > > Playing WAVE 'audio_s16le_44100.wav' : Signed 16 bit Little Endian, Rate > > > 44100 Hz, Stereo > > > aplay: pcm_write:1710: write error: Input/output error > > > ----------------------------------------- > > > > > > issue 2: this log appeared while playback with the aplay, then aplay was > > > stopped. > > > ( this issue isn't appear on the kzm-a9-gt board only ) > > > > > > ------- log ----------------------------- > > > # aplay audio_s16le_44100.wav > > > Playing WAVE 'audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, > > > Stereo > > > BUG: scheduling while atomic: kworker/1:1/273/0x00000103 > > > aplay: pcm_write:1710: write error: Input/output error > > > ----------------------------------------- > > > > > > I think these issues are introduced by "dmaengine:add an shdma-base > > > library" > > > commit. > > > > > > ---------- commit ----------------------- > > > commit 9a7b8e002e331d0599127f16613c32f425a14f2c > > > Author: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > > > Date: Wed May 9 17:09:13 2012 +0200 > > > > > > dmaengine: add an shdma-base library > > > > > > This patch extracts code from shdma.c, that does not directly deal > > > with > > > hardware implementation details and can be re-used with diverse DMA > > > controller variants, found on SH-based SoCs. > > > > > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > > > Cc: Sascha Hauer <s.hauer(a)pengutronix.de> > > > Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> > > > ------------------------------------------ > > > > > > About issue 1: I found that > > > The fsi.c uses fsi_dma_do_tasklet() to setup dma-transfer.This is a > > > function > > > of a tasklet. > > > When one dma-transfer is finished shdma_chan_ld_cleanup() is called.In > > > this > > > function > > > __ld_cleanup() is called to cleanup the schan->ld_queue. In behavior of > > > __ld_cleanup() > > > if descriptor is the last chunk its callback is called, then its node is > > > cleanup in the > > > next call of __ld_cleanup().In callback of the last descriptor, > > > fsi_dma_do_tasklet() is > > > scheduled then is called after the shdma_chan_ld_cleanup() is finished. > > > But when this issue appear, shdma_chan_ld_cleanup() is called while > > > shdma_chan_ld_cleanup() > > > is not finished.At that time, schan->ld_queue is not empty, so in > > > shdma_tx_submit() > > > schan->pm_state is set to SHDMA_PM_PENDING, and then dma-transfer isn't > > > started in > > > shdma_issue_pending(). > > > > > > About issue 2: > > > I found that this issue is appear when pm_runtime_barrier() is running. > > > This issue is not appear when disable CONFIG_PM_RUNTIME config. > > > > > > What do you think about these issues? > > > > > > Best regards > > > --- > > > Thang > > diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c > > index 0540408..1bb0d58c 100644 > > --- a/sound/soc/sh/fsi.c > > +++ b/sound/soc/sh/fsi.c > > @@ -20,6 +20,7 @@ > > #include <linux/sh_dma.h> > > #include <linux/slab.h> > > #include <linux/module.h> > > +#include <linux/workqueue.h> > > #include <sound/soc.h> > > #include <sound/sh_fsi.h> > > @@ -223,7 +224,7 @@ struct fsi_stream { > > */ > > struct dma_chan *chan; > > struct sh_dmae_slave slave; /* see fsi_handler_init() */ > > - struct tasklet_struct tasklet; > > + struct work_struct work; > > dma_addr_t dma; > > }; > > @@ -1085,9 +1086,9 @@ static void fsi_dma_complete(void *data) > > snd_pcm_period_elapsed(io->substream); > > } > > -static void fsi_dma_do_tasklet(unsigned long data) > > +static void fsi_dma_do_work(struct work_struct *work) > > { > > - struct fsi_stream *io = (struct fsi_stream *)data; > > + struct fsi_stream *io = container_of(work, struct fsi_stream, work); > > struct fsi_priv *fsi = fsi_stream_to_priv(io); > > struct snd_soc_dai *dai; > > struct dma_async_tx_descriptor *desc; > > @@ -1129,7 +1130,7 @@ static void fsi_dma_do_tasklet(unsigned long data) > > * FIXME > > * > > * In DMAEngine case, codec and FSI cannot be started simultaneously > > - * since FSI is using tasklet. > > + * since FSI is using the scheduler work queue. > > * Therefore, in capture case, probably FSI FIFO will have got > > * overflow error in this point. > > * in that case, DMA cannot start transfer until error was cleared. > > @@ -1153,7 +1154,7 @@ static bool fsi_dma_filter(struct dma_chan *chan, void > > *param) > > static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) > > { > > - tasklet_schedule(&io->tasklet); > > + schedule_work(&io->work); > > return 0; > > } > > @@ -1195,14 +1196,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, > > struct fsi_stream *io, struct dev > > return fsi_stream_probe(fsi, dev); > > } > > - tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); > > + INIT_WORK(&io->work, fsi_dma_do_work); > > return 0; > > } > > static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) > > { > > - tasklet_kill(&io->tasklet); > > + cancel_work_sync(&io->work); > > fsi_stream_stop(fsi, io); > > > > > --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

2 1

[alsa-devel] Internal Microphone in Lenovo P580 not working with alsa 1.0.25
by mefis 05 Oct '12

05 Oct '12

Hi, Mic not working in Lenovo P580 with alsa 1.0.25 Alsa Information script output: http://www.alsa-project.org/db/?f=001523ed61bc76fb5b2ad0d3c354c043f89e5a33 Hope it helps. !!################################ !!ALSA Information Script v 0.4.61 !!################################ !!Script ran on: Fri Oct 5 05:56:38 UTC 2012 !!Linux Distribution !!------------------ Ubuntu 12.04.1 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 12.04.1 LTS" !!DMI Information !!--------------- Manufacturer: LENOVO Product Name: Lenovo IdeaPad P580 Product Version: Lenovo IdeaPad P580 Firmware Version: 5ECN33WW(V2.03) !!Kernel Information !!------------------ Kernel release: 3.2.0-31-generic Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.25 Library version: 1.0.16 Utilities version: 1.0.25 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xe0610000 irq 46 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation Panther Point High Definition Audio Controller (rev 04) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:1e20 (rev 04) Subsystem: 17aa:3977 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-caiaq: index=-2 snd-usb-ua101: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-usb-audio: index=-2 snd-hda-intel: model=basic snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-caiaq: index=-2 snd-usb-ua101: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-usb-audio: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : N bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : 1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : Y id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 model : basic,(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Conexant CX20590 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f1506e Subsystem Id: 0x17aac02d Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CONEXANT Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x39 0x39] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3c 0x3c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x49 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Device: name="CONEXANT Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x50 0x50] [0x80 0x80] [0x50 0x50] [0x80 0x80] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17 0x18 0x23* 0x24 Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17 0x18 0x23 0x24* Node 0x16 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x17 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x02 0x02] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a 0x1b* 0x1d 0x1e Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a* 0x1b 0x1d 0x1e Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x04211040: [Jack] HP Out at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=37, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1a [Pin Complex] wcaps 0x400481: Stereo Pincap 0x00001324: IN Detect Vref caps: HIZ 50 80 Pin Default 0x04a11030: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=01, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00011334: IN OUT EAPD Detect Vref caps: HIZ 50 80 EAPD 0x2: EAPD Pin Default 0x95a70120: [Fixed] Mic at Int Top Conn = Analog, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=38, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1d [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1e [Pin Complex] wcaps 0x400481: Stereo Pincap 0x00000024: IN Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1f [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000010: OUT Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x20 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x49 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x22 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x21 Node 0x23 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x2f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00000020: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x24 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11 Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Intel PantherPoint HDMI Address: 3 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862806 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x02 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x80] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x58560020: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x03 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x80] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x58560030: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---T 1 root audio 116, 7 Oct 5 00:44 /dev/snd/controlC0 crw-rw---T 1 root audio 116, 6 Oct 5 00:44 /dev/snd/hwC0D0 crw-rw---T 1 root audio 116, 5 Oct 5 00:44 /dev/snd/hwC0D3 crw-rw---T 1 root audio 116, 4 Oct 5 00:48 /dev/snd/pcmC0D0c crw-rw---T 1 root audio 116, 3 Oct 5 00:48 /dev/snd/pcmC0D0p crw-rw---T 1 root audio 116, 2 Oct 5 00:44 /dev/snd/pcmC0D3p crw-rw---T 1 root audio 116, 1 Oct 5 00:44 /dev/snd/seq crw-rw---T 1 root audio 116, 33 Oct 5 00:44 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Oct 5 00:44 . drwxr-xr-x 3 root root 220 Oct 5 00:44 .. lrwxrwxrwx 1 root root 12 Oct 5 00:44 pci-0000:00:1b.0 -> ../controlC0 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [PCH] Card hw:0 'PCH'/'HDA Intel PCH at 0xe0610000 irq 46' Mixer name : 'Intel PantherPoint HDMI' Components : 'HDA:14f1506e,17aac02d,00100003 HDA:80862806,80860101,00100000' Controls : 15 Simple ctrls : 7 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 57 [77%] [-17.00dB] [on] Front Right: Playback 57 [77%] [-17.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 135 [53%] [-24.00dB] Front Right: Playback 135 [53%] [-24.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Beep',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 7 Mono: Playback 3 [43%] [-16.00dB] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 80 Front Left: Capture 80 [100%] [6.00dB] [on] Front Right: Capture 80 [100%] [6.00dB] [on] Simple mixer control 'Analog Mic Boost',0 Capabilities: cenum Items: '0dB' '10dB' '20dB' '30dB' '40dB' Item0: '20dB' Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] !!Alsactl output !!-------------- --startcollapse-- state.PCH { control.1 { iface MIXER name 'Master Playback Volume' value.0 57 value.1 57 comment { access 'read write' type INTEGER count 2 range '0 - 74' dbmin -7400 dbmax 0 dbvalue.0 -1700 dbvalue.1 -1700 } } control.2 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.3 { iface MIXER name 'Analog Mic Boost Capture Enum' value '20dB' comment { access 'read write' type ENUMERATED count 1 item.0 '0dB' item.1 '10dB' item.2 '20dB' item.3 '30dB' item.4 '40dB' } } control.4 { iface MIXER name 'Capture Volume' value.0 80 value.1 80 comment { access 'read write' type INTEGER count 2 range '0 - 80' dbmin -7400 dbmax 600 dbvalue.0 600 dbvalue.1 600 } } control.5 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.6 { iface MIXER name 'Beep Playback Volume' value 3 comment { access 'read write' type INTEGER count 1 range '0 - 7' dbmin -2800 dbmax 0 dbvalue.0 -1600 } } control.7 { iface MIXER name 'Beep Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.8 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.9 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.11 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.12 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.13 { iface PCM device 3 name ELD value '' comment { access read type BYTES count 0 } } control.14 { iface MIXER name 'PCM Playback Volume' value.0 135 value.1 135 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 -2400 dbvalue.1 -2400 } } control.15 { iface MIXER name 'Digital Capture Volume' value.0 60 value.1 60 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module rfcomm bnep bluetooth parport_pc ppdev binfmt_misc snd_hda_codec_hdmi snd_hda_codec_conexant joydev snd_hda_intel snd_hda_codec snd_hwdep snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_timer snd_seq_device snd alx compat arc4 soundcore snd_page_alloc iwlwifi mac80211 uvcvideo ideapad_laptop videodev i915 drm_kms_helper v4l2_compat_ioctl32 mei drm sparse_keymap cfg80211 i2c_algo_bit video psmouse mac_hid serio_raw lp parport usbhid hid !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x19 0x04211040 0x1a 0x04a11030 0x1b 0x95a70120 0x1c 0x400001f0 0x1d 0x400001f0 0x1e 0x400001f0 0x1f 0x92170110 0x20 0x400001f0 0x22 0x400001f0 0x23 0x400001f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D3/init_pin_configs: 0x05 0x18560010 0x06 0x58560020 0x07 0x58560030 /sys/class/sound/hwC0D3/driver_pin_configs: /sys/class/sound/hwC0D3/user_pin_configs: /sys/class/sound/hwC0D3/init_verbs: !!ALSA/HDA dmesg !!-------------- [ 18.438062] mei 0000:00:16.0: irq 45 for MSI/MSI-X [ 18.583957] snd_hda_intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 [ 18.584008] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X [ 18.584036] snd_hda_intel 0000:00:1b.0: setting latency timer to 64 [ 18.650583] psmouse serio1: synaptics: Touchpad model: 1, fw: 8.0, id: 0x1e2b1, caps: 0xd00123/0x840300/0x123c00 [ 18.724639] input: SynPS/2 Synaptics TouchPad as /devices/platform/i8042/serio1/input/input9 [ 19.124515] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input10 [ 19.128227] HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 [ 19.128283] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 [ 19.888422] EXT4-fs (sda5): re-mounted. Opts: errors=remount-ro

2 1

[alsa-devel] [PATCH] ASoC: SAMSUNG: spdif: use clk_prepare_enable and clk_disable_unprepare
by Thomas Abraham 05 Oct '12

05 Oct '12

Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare calls as required by common clock framework. Signed-off-by: Thomas Abraham <thomas.abraham(a)linaro.org> --- sound/soc/samsung/spdif.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index bc24c7a..5f3b06d 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -397,7 +397,7 @@ static __devinit int spdif_probe(struct platform_device *pdev) ret = -ENOENT; goto err0; } - clk_enable(spdif->pclk); + clk_prepare_enable(spdif->pclk); spdif->sclk = clk_get(&pdev->dev, "sclk_spdif"); if (IS_ERR(spdif->sclk)) { @@ -405,7 +405,7 @@ static __devinit int spdif_probe(struct platform_device *pdev) ret = -ENOENT; goto err1; } - clk_enable(spdif->sclk); + clk_prepare_enable(spdif->sclk); /* Request S/PDIF Register's memory region */ if (!request_mem_region(mem_res->start, @@ -444,10 +444,10 @@ err4: err3: release_mem_region(mem_res->start, resource_size(mem_res)); err2: - clk_disable(spdif->sclk); + clk_disable_unprepare(spdif->sclk); clk_put(spdif->sclk); err1: - clk_disable(spdif->pclk); + clk_disable_unprepare(spdif->pclk); clk_put(spdif->pclk); err0: return ret; @@ -466,9 +466,9 @@ static __devexit int spdif_remove(struct platform_device *pdev) if (mem_res) release_mem_region(mem_res->start, resource_size(mem_res)); - clk_disable(spdif->sclk); + clk_disable_unprepare(spdif->sclk); clk_put(spdif->sclk); - clk_disable(spdif->pclk); + clk_disable_unprepare(spdif->pclk); clk_put(spdif->pclk); return 0; -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: SAMSUNG: pcm: use clk_prepare_enable and clk_disable_unprepare
by Thomas Abraham 05 Oct '12

05 Oct '12

Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare calls as required by common clock framework. Signed-off-by: Thomas Abraham <thomas.abraham(a)linaro.org> --- sound/soc/samsung/pcm.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index c860819..45f4a75 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -543,7 +543,7 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) ret = PTR_ERR(pcm->cclk); goto err1; } - clk_enable(pcm->cclk); + clk_prepare_enable(pcm->cclk); /* record our pcm structure for later use in the callbacks */ dev_set_drvdata(&pdev->dev, pcm); @@ -568,7 +568,7 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) ret = -ENOENT; goto err4; } - clk_enable(pcm->pclk); + clk_prepare_enable(pcm->pclk); s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start + S3C_PCM_RXFIFO; @@ -592,14 +592,14 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) return 0; err5: - clk_disable(pcm->pclk); + clk_disable_unprepare(pcm->pclk); clk_put(pcm->pclk); err4: iounmap(pcm->regs); err3: release_mem_region(mem_res->start, resource_size(mem_res)); err2: - clk_disable(pcm->cclk); + clk_disable_unprepare(pcm->cclk); clk_put(pcm->cclk); err1: return ret; @@ -619,8 +619,8 @@ static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem_res->start, resource_size(mem_res)); - clk_disable(pcm->cclk); - clk_disable(pcm->pclk); + clk_disable_unprepare(pcm->cclk); + clk_disable_unprepare(pcm->pclk); clk_put(pcm->pclk); clk_put(pcm->cclk); -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: SAMSUNG: ac97: use clk_prepare_enable and clk_disable_unprepare
by Thomas Abraham 05 Oct '12

05 Oct '12

Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare calls as required by common clock framework. Signed-off-by: Thomas Abraham <thomas.abraham(a)linaro.org> --- sound/soc/samsung/ac97.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 14fbcd3..386bab1 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -442,7 +442,7 @@ static __devinit int s3c_ac97_probe(struct platform_device *pdev) ret = -ENODEV; goto err2; } - clk_enable(s3c_ac97.ac97_clk); + clk_prepare_enable(s3c_ac97.ac97_clk); if (ac97_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); @@ -468,7 +468,7 @@ err5: free_irq(irq_res->start, NULL); err4: err3: - clk_disable(s3c_ac97.ac97_clk); + clk_disable_unprepare(s3c_ac97.ac97_clk); clk_put(s3c_ac97.ac97_clk); err2: iounmap(s3c_ac97.regs); @@ -488,7 +488,7 @@ static __devexit int s3c_ac97_remove(struct platform_device *pdev) if (irq_res) free_irq(irq_res->start, NULL); - clk_disable(s3c_ac97.ac97_clk); + clk_disable_unprepare(s3c_ac97.ac97_clk); clk_put(s3c_ac97.ac97_clk); iounmap(s3c_ac97.regs); -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: SAMSUNG: i2s: use clk_prepare_enable and clk_disable_unprepare
by Thomas Abraham 05 Oct '12

05 Oct '12

Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare calls as required by common clock framework. Signed-off-by: Thomas Abraham <thomas.abraham(a)linaro.org> --- sound/soc/samsung/i2s.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 40b00a1..547b919 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -423,7 +423,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (i2s->op_clk) { if ((clk_id && !(mod & MOD_IMS_SYSMUX)) || (!clk_id && (mod & MOD_IMS_SYSMUX))) { - clk_disable(i2s->op_clk); + clk_disable_unprepare(i2s->op_clk); clk_put(i2s->op_clk); } else { i2s->rclk_srcrate = @@ -434,7 +434,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, i2s->op_clk = clk_get(&i2s->pdev->dev, i2s->src_clk[clk_id]); - clk_enable(i2s->op_clk); + clk_prepare_enable(i2s->op_clk); i2s->rclk_srcrate = clk_get_rate(i2s->op_clk); /* Over-ride the other's */ @@ -880,7 +880,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) iounmap(i2s->addr); return -ENOENT; } - clk_enable(i2s->clk); + clk_prepare_enable(i2s->clk); if (other) { other->addr = i2s->addr; @@ -922,7 +922,7 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) if (i2s->quirks & QUIRK_NEED_RSTCLR) writel(0, i2s->addr + I2SCON); - clk_disable(i2s->clk); + clk_disable_unprepare(i2s->clk); clk_put(i2s->clk); iounmap(i2s->addr); -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: fsi: use devm_request_irq()
by Kuninori Morimoto 04 Oct '12

04 Oct '12

This patch uses devm_request_irq() instead of request_irq() and removed free_irq() from driver Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5328ae5..05b8b55 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1701,7 +1701,7 @@ static int fsi_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); dev_set_drvdata(&pdev->dev, master); - ret = request_irq(irq, &fsi_interrupt, 0, + ret = devm_request_irq(&pdev->dev, irq, &fsi_interrupt, 0, id_entry->name, master); if (ret) { dev_err(&pdev->dev, "irq request err\n"); @@ -1711,7 +1711,7 @@ static int fsi_probe(struct platform_device *pdev) ret = snd_soc_register_platform(&pdev->dev, &fsi_soc_platform); if (ret < 0) { dev_err(&pdev->dev, "cannot snd soc register\n"); - goto exit_free_irq; + goto exit_fsib; } ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai, @@ -1725,8 +1725,6 @@ static int fsi_probe(struct platform_device *pdev) exit_snd_soc: snd_soc_unregister_platform(&pdev->dev); -exit_free_irq: - free_irq(irq, master); exit_fsib: pm_runtime_disable(&pdev->dev); fsi_stream_remove(&master->fsib); @@ -1742,7 +1740,6 @@ static int fsi_remove(struct platform_device *pdev) master = dev_get_drvdata(&pdev->dev); - free_irq(master->irq, master); pm_runtime_disable(&pdev->dev); snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai)); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: fsi: fixup channels_min/max
by Kuninori Morimoto 04 Oct '12

04 Oct '12

FSI can support 1 - 8 channel input/output, but current driver is supporting 2 channel format only. This patch fixes channel mismatch Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 0540408..2b1062b 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1497,7 +1497,7 @@ static struct snd_pcm_hardware fsi_pcm_hardware = { .rates = FSI_RATES, .rate_min = 8000, .rate_max = 192000, - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 64 * 1024, .period_bytes_min = 32, @@ -1585,14 +1585,14 @@ static struct snd_soc_dai_driver fsi_soc_dai[] = { .playback = { .rates = FSI_RATES, .formats = FSI_FMTS, - .channels_min = 1, - .channels_max = 8, + .channels_min = 2, + .channels_max = 2, }, .capture = { .rates = FSI_RATES, .formats = FSI_FMTS, - .channels_min = 1, - .channels_max = 8, + .channels_min = 2, + .channels_max = 2, }, .ops = &fsi_dai_ops, }, @@ -1601,14 +1601,14 @@ static struct snd_soc_dai_driver fsi_soc_dai[] = { .playback = { .rates = FSI_RATES, .formats = FSI_FMTS, - .channels_min = 1, - .channels_max = 8, + .channels_min = 2, + .channels_max = 2, }, .capture = { .rates = FSI_RATES, .formats = FSI_FMTS, - .channels_min = 1, - .channels_max = 8, + .channels_min = 2, + .channels_max = 2, }, .ops = &fsi_dai_ops, }, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: ams-delta: fix card initalization failure
by Janusz Krzysztofik 04 Oct '12

04 Oct '12

Since commit 0998d0631001288a5974afc0b2a5f568bcdecb4d, 'device-core: Ensure drvdata = NULL when no driver is bound', the Amstrad Delta sound card no longer initializes correctly due to drvdata reset to NULL by an upper layer before the codec device, required for successful card setup, is registered. Fix this by moving the codec registration bits up, before the card is probed for. Created and tested against linux-3.6-rc3 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- sound/soc/omap/ams-delta.c | 18 ++++++++++-------- 1 files changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 7d4fa8e..270de9c 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -590,20 +590,22 @@ static int __init ams_delta_module_init(void) if (!ams_delta_audio_platform_device) return -ENOMEM; - platform_set_drvdata(ams_delta_audio_platform_device, - &ams_delta_audio_card); - - ret = platform_device_add(ams_delta_audio_platform_device); - if (ret) - goto err; - /* * Codec platform device could be registered from elsewhere (board?), * but I do it here as it makes sense only if used with the card. + * Moreover, it must be registered before the card is probed for, + * or the card setup fails due to drvdata reset by upper layers. */ cx20442_platform_device = platform_device_register_simple("cx20442-codec", -1, NULL, 0); - return 0; + + platform_set_drvdata(ams_delta_audio_platform_device, + &ams_delta_audio_card); + + ret = platform_device_add(ams_delta_audio_platform_device); + if (!ret) + return ret; + err: platform_device_put(ams_delta_audio_platform_device); return ret; -- 1.7.3.4

4 14

[alsa-devel] [PATCH] ASoC: omap-abe-twl6040: Fix typo of Vibrator
by Peter Ujfalusi 04 Oct '12

04 Oct '12

It is not Vinrator but Vibrator. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-abe-twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index be525df..750d595 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -220,7 +220,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); + twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator"); twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); -- 1.7.12

2 1

[alsa-devel] [PATCH] ASoC: twl6040: Fix Stream DAPM mapping
by Peter Ujfalusi 04 Oct '12

04 Oct '12

Fixes commit: 805238b ASoC: twl6040: Convert to use DAI DAPM widgets where the connection between the stream widgets and the ADC widgets was reversed and because of this on capture the DAPM is not powering up the codec. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Liam, in the mentioned commit I made a mistake and the DAPM route got reversed for capture. I did the same change for twl4030 and twl6040 at the same time and only tested twl4030 - since it was easier for me. Obviously I did it wrong for twl6040 :( Can you queue this as a fix for 3.7? Thank you, Peter sound/soc/codecs/twl6040.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index e8f97af..00b85cc 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -820,10 +820,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"VIBRA DAC", NULL, "Vibra Playback"}, /* ADC -> Stream mapping */ - {"ADC Left", NULL, "Legacy Capture"}, - {"ADC Left", NULL, "Capture"}, - {"ADC Right", NULL, "Legacy Capture"}, - {"ADC Right", NULL, "Capture"}, + {"Legacy Capture" , NULL, "ADC Left"}, + {"Capture", NULL, "ADC Left"}, + {"Legacy Capture", NULL, "ADC Right"}, + {"Capture" , NULL, "ADC Right"}, /* Capture path */ {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, -- 1.7.12

2 1

[alsa-devel] [PATCH v2] ALSA: ASoC: cs4271: add support for AMUTEB=BMUTEC feature
by Daniel Mack 04 Oct '12

04 Oct '12

The CS4271 has a feature to sync its analog mute flags, so one mute circuitry can be used for both channels. Give users access to this feature with a new DT property and a flag in the platform data. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Alexander Sverdlin <subaparts(a)yandex.ru> --- Documentation/devicetree/bindings/sound/cs4271.txt | 2 ++ include/sound/cs4271.h | 1 + sound/soc/codecs/cs4271.c | 21 ++++++++++++++++++--- 3 files changed, 21 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index c81b5fd..a850fb9 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -18,6 +18,8 @@ Optional properties: - reset-gpio: a GPIO spec to define which pin is connected to the chip's !RESET pin + - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag + is enabled. Examples: diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index 50a059e..6d9e15e 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -19,6 +19,7 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ + int amutec_eq_bmutec:1; /* flag to enable AMUTEC=BMUTEC */ }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f994af3..2ac5fe6 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -474,15 +474,25 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; + int amutec_eq_bmutec = 0; #ifdef CONFIG_OF - if (of_match_device(cs4271_dt_ids, codec->dev)) + if (of_match_device(cs4271_dt_ids, codec->dev)) { gpio_nreset = of_get_named_gpio(codec->dev->of_node, "reset-gpio", 0); + + if (!of_get_property(codec->dev->of_node, + "cirrus,amutec-eq-bmutec", NULL)) + amutec_eq_bmutec = 1; + } #endif - if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; + if (cs4271plat) { + if (gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; + + amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; + } if (gpio_nreset >= 0) if (gpio_request(gpio_nreset, "CS4271 Reset")) @@ -528,6 +538,11 @@ static int cs4271_probe(struct snd_soc_codec *codec) /* Power-up sequence requires 85 uS */ udelay(85); + if (amutec_eq_bmutec) + snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_MUTECAEQUB, + CS4271_MODE2_MUTECAEQUB); + return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); } -- 1.7.11.4

2 1

[alsa-devel] snd_hwdep_open() as non-root user
by Rafal Powierski 04 Oct '12

04 Oct '12

Developers, I would like to access pin verb hda interface using snd_hwdep_open() function. It works fine if I run as root. I hope there is a way to configure my kubuntu so I can do it as non-root user (e.g. debugging) so far, I have been trying changing access flags for devices in /dev/snd/ directory to crwxrwxrwx no plus character. It did not change my privileges for snd_hwdep_open() - still returns –13. I assume that the device-file responsible is /dev/snd/hwC1D0, the same as parameter given to “hda-verb” program. I am developing a new hardware, using HDA codec. Please help

2 2

[alsa-devel] [PATCH] ALSA: ASoC: cs4271: add DT property for AMUTEB=BMUTEC feature
by Daniel Mack 04 Oct '12

04 Oct '12

The CS4271 has a feature to sync its analog mute flags, so one mute circuitry can be used for both channels. Give DT uses access to this feature with a new property. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Alexander Sverdlin <subaparts(a)yandex.ru> --- Documentation/devicetree/bindings/sound/cs4271.txt | 2 ++ sound/soc/codecs/cs4271.c | 13 ++++++++++++- 2 files changed, 14 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index c81b5fd..a850fb9 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -18,6 +18,8 @@ Optional properties: - reset-gpio: a GPIO spec to define which pin is connected to the chip's !RESET pin + - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag + is enabled. Examples: diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f994af3..56ec8dd2 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -474,11 +474,17 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; + int amutec_eq_bmutec = 0; #ifdef CONFIG_OF - if (of_match_device(cs4271_dt_ids, codec->dev)) + if (of_match_device(cs4271_dt_ids, codec->dev)) { gpio_nreset = of_get_named_gpio(codec->dev->of_node, "reset-gpio", 0); + + if (!of_get_property(codec->dev->of_node, + "cirrus,amutec-eq-bmutec", NULL)) + amutec_eq_bmutec = 1; + } #endif if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) @@ -528,6 +534,11 @@ static int cs4271_probe(struct snd_soc_codec *codec) /* Power-up sequence requires 85 uS */ udelay(85); + if (amutec_eq_bmutec) + snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_MUTECAEQUB, + CS4271_MODE2_MUTECAEQUB); + return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); } -- 1.7.11.4

2 2

Re: [alsa-devel] [PATCH 2/3] ASoC: Davinci: pcm: add support for sram-support-less platforms
by Daniel Mack 04 Oct '12

04 Oct '12

On 04.10.2012 12:28, Matt Porter wrote: > On Thu, Oct 04, 2012 at 11:57:30AM +0200, Daniel Mack wrote: >> On 04.10.2012 11:38, Porter, Matt wrote: >>> >>> On Oct 4, 2012, at 5:21 AM, Daniel Mack wrote: >>> >>>> >>>> >>>> On 02.10.2012 18:50, Daniel Mack wrote: >>>>> On 02.10.2012 18:41, Matt Porter wrote: >>>>>> On Tue, Oct 02, 2012 at 03:42:47PM +0200, Daniel Mack wrote: >>>>>>> On 02.10.2012 13:06, Sekhar Nori wrote: >>>>>>>> On 10/2/2012 4:03 PM, Daniel Mack wrote: >>>>>>>>> On 02.10.2012 11:37, Mark Brown wrote: >>>>>>>>>> On Tue, Oct 02, 2012 at 10:48:53AM +0300, Peter Ujfalusi wrote: >>>>>>>>>> >>>>>>>>>>> I also agree that ifdef is not a good solution. >>>>>>>>>>> It is better to have this information passed as device_data and via DT it can >>>>>>>>>>> be decided based on the compatible property for the device. >>>>>>>>>> >>>>>>>>>> That's not really the problem here - the problem is that the APIs used >>>>>>>>>> to get the SRAM are DaVinci only so it's not possible to build on OMAP >>>>>>>>>> or other platforms. The SRAM code needs to move to a standard API. >>>>>>>>> >>>>>>>>> What about following Matt Porter's idea and ignore the SRAM code >>>>>>>>> entirely and port the entire PCM code to generic dmaengine code first? >>>>>>>>> The EDMA driver needs to learn support for cyclic DMA for that, and I >>>>>>>>> might give that a try in near future. >>>>>>>>> >>>>>>>>> Later on, the SRAM ping-pong code can get added back using genalloc >>>>>>>>> functions, as Sekhar proposed. That needs to be done by someone who has >>>>>>>>> access to a Davinci board though, I only have a AM33xx/OMAP here. >>>>>>>> >>>>>>>> We cannot "get rid" of SRAM code and add it back "later". It is required >>>>>>>> for most DaVinci parts. The SRAM code can be converted to use genalloc >>>>>>>> (conversion should be straightforward and I can help test it) and the >>>>>>>> code that uses SRAM can probably keep using the private EDMA API till >>>>>>>> the dmaengine EDMA driver has cyclic DMA support. Matt has already >>>>>>>> posted patches to move private EDMA APIs to a common location. That >>>>>>>> should keep AM335x build from breaking. Is this something that is feasible? >>>>>>> >>>>>>> Yes - by "later" I just meant in a subsequent patch. But you're probably >>>>>>> right, we can also do that first. >>>>>>> >>>>>>> I'm looking at that right now and the problem seems that we don't have a >>>>>>> sane way to dynamically look up gen_pools independently of the actual >>>>>>> run-time platform. All users of gen_pools seem to know which platform >>>>>>> they run on and access a platform-specific pool. So I don't currently >>>>>>> see how we could implement multi-platform code, gen_pools are fine but >>>>>>> don't solve the problem here. >>>>>>> >>>>>>> Would it be an idea to add a char* member to gen_pools and a function >>>>>>> that can be used to dynamically look it up again? If a buffer with a >>>>>>> certain name exists, we can use it and install that ping-pong buffer, >>>>>>> otherwise we just don't. While that would be easy to do, it's a >>>>>>> tree-wide change. >>>>>>> >>>>>>> Is there a better way that I miss? >>>>>> >>>>>> At the high level there's two platform models we have to handle, the >>>>>> boot from board file !DT case, and then the boot from DT case. Since >>>>>> Davinci is just starting DT conversion, we mostly care about the !DT >>>>>> base in which the struct gen_pool * is passed in pdata to the ASoC >>>>>> driver. It is then selectable on a per-platform basis where the decision >>>>>> should be made. >>>>>> >>>>>> Given a separate discussion with Sekhar, we're only going to have one >>>>>> SRAM pool on any DaVinci part right now...this was only a question on >>>>>> L138 anyway. But regardless, the created gen_pool will be passed via >>>>>> pdata. >>>>> >>>>> I thought about this too, as mmp does it that way. >>>>> >>>>>> Since DT conversion is starting and we need to consider that now, >>>>>> the idea there is to use the DT-based generic sram driver [1] such that >>>>>> when we do boot from DT on Davinci, the genpool is provided via phandle >>>>>> and the pointer extracted with the OF helpers that are part of the >>>>>> series. >>>>> >>>>> A phandle is the cleanest way I think, yes. >>>>> >>>>>> That's pretty much it. I'm reworking the backend support as discussed >>>>>> with Sekhar wrt to my uio_pruss series. I can post a standalone series >>>>>> that just replaces sram_* with genalloc for davinci ASoC. >>>>> >>>>> As you can also test it, it would be easiest if you came up with a patch >>>>> for that, yes. I can have a look at the dma bits laters, once my OMAP >>>>> board finally works with the code as it currently stands. I'm still >>>>> fighting with the mcasp driver right now ... >>>> >>>> I quickly prepared two patches to change that, so that topic is out of >>>> the way. But I did only compile-test them on OMAP - could you check on >>>> your Davinci platform? Note that these apply on top of the patch in >>>> discussion here (which isn't applied to the asoc tree yet). >>> >>> I put a series together yesterday, just ran out of time to post >>> last night after testing. I'm posting that now...it's on top of my >>> uio_pruss/genalloc series and only addresss switching davinci-pcm to >>> genalloc (and actually enabling ping-pon from sram). >> >> Ok, I don't care which version makes it in after all :) > > Ok :) > >>> I'll take a look a your OMAP patches. >> >> The patches I just sent out are only for Davinci and change the SRAM >> functions to genalloc in the mcasp driver. They work just fine on OMAP >> of course, as the code is disabled in there. > > Ok, same thing as I did and just sent out. These are tested on AM180x > and hook up ping-pong buffering so that there's actually one user of > all this ping-pong code. It's been sitting idle for a long time. > > I didn't see your patches come by yet but will take a look to see if > we need to combine. I attached them to my mail for a quick review. I wanted to queue them up to a bigger series I'll send out later ... Daniel

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[alsa-devel] HDA-Intel: despite `power_save=1` device hwC0D0 uses 100 % according to Powertop 2.x
by Paul Menzel 04 Oct '12

04 Oct '12

Dear ALSA folks, running Debian Wheezy with Linux 3.2.23 on a T60 with 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xee240000 irq 46 29 [ThinkPadEC ]: ThinkPad EC - ThinkPad Console Audio Control ThinkPad Console Audio Control at EC reg 0x30, fw 79HT50WW-1.07 (attached `alsa-info.sh` output from before running the following command) despite enabling power saving $ echo 1 | sudo tee /sys/module/snd_hda_intel/parameters/power_save 1 and touching the device or playing some music, I am able to hear a pop/click before playing another piece of music, so some power saving mode seems to be enabled, Powertop 2.1 built from Git still shows the device using 100 %. Others seem to have similar problem with other devices though [1]. Thanks, Paul [1] http://mancoosi.org/~abate/audio-codec-hwc0d0-conexant-and-powertop

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Re: [alsa-devel] [PATCH 2/3] ASoC: Davinci: pcm: add support for sram-support-less platforms
by Daniel Mack 04 Oct '12

04 Oct '12

On 04.10.2012 11:38, Porter, Matt wrote: > > On Oct 4, 2012, at 5:21 AM, Daniel Mack wrote: > >> >> >> On 02.10.2012 18:50, Daniel Mack wrote: >>> On 02.10.2012 18:41, Matt Porter wrote: >>>> On Tue, Oct 02, 2012 at 03:42:47PM +0200, Daniel Mack wrote: >>>>> On 02.10.2012 13:06, Sekhar Nori wrote: >>>>>> On 10/2/2012 4:03 PM, Daniel Mack wrote: >>>>>>> On 02.10.2012 11:37, Mark Brown wrote: >>>>>>>> On Tue, Oct 02, 2012 at 10:48:53AM +0300, Peter Ujfalusi wrote: >>>>>>>> >>>>>>>>> I also agree that ifdef is not a good solution. >>>>>>>>> It is better to have this information passed as device_data and via DT it can >>>>>>>>> be decided based on the compatible property for the device. >>>>>>>> >>>>>>>> That's not really the problem here - the problem is that the APIs used >>>>>>>> to get the SRAM are DaVinci only so it's not possible to build on OMAP >>>>>>>> or other platforms. The SRAM code needs to move to a standard API. >>>>>>> >>>>>>> What about following Matt Porter's idea and ignore the SRAM code >>>>>>> entirely and port the entire PCM code to generic dmaengine code first? >>>>>>> The EDMA driver needs to learn support for cyclic DMA for that, and I >>>>>>> might give that a try in near future. >>>>>>> >>>>>>> Later on, the SRAM ping-pong code can get added back using genalloc >>>>>>> functions, as Sekhar proposed. That needs to be done by someone who has >>>>>>> access to a Davinci board though, I only have a AM33xx/OMAP here. >>>>>> >>>>>> We cannot "get rid" of SRAM code and add it back "later". It is required >>>>>> for most DaVinci parts. The SRAM code can be converted to use genalloc >>>>>> (conversion should be straightforward and I can help test it) and the >>>>>> code that uses SRAM can probably keep using the private EDMA API till >>>>>> the dmaengine EDMA driver has cyclic DMA support. Matt has already >>>>>> posted patches to move private EDMA APIs to a common location. That >>>>>> should keep AM335x build from breaking. Is this something that is feasible? >>>>> >>>>> Yes - by "later" I just meant in a subsequent patch. But you're probably >>>>> right, we can also do that first. >>>>> >>>>> I'm looking at that right now and the problem seems that we don't have a >>>>> sane way to dynamically look up gen_pools independently of the actual >>>>> run-time platform. All users of gen_pools seem to know which platform >>>>> they run on and access a platform-specific pool. So I don't currently >>>>> see how we could implement multi-platform code, gen_pools are fine but >>>>> don't solve the problem here. >>>>> >>>>> Would it be an idea to add a char* member to gen_pools and a function >>>>> that can be used to dynamically look it up again? If a buffer with a >>>>> certain name exists, we can use it and install that ping-pong buffer, >>>>> otherwise we just don't. While that would be easy to do, it's a >>>>> tree-wide change. >>>>> >>>>> Is there a better way that I miss? >>>> >>>> At the high level there's two platform models we have to handle, the >>>> boot from board file !DT case, and then the boot from DT case. Since >>>> Davinci is just starting DT conversion, we mostly care about the !DT >>>> base in which the struct gen_pool * is passed in pdata to the ASoC >>>> driver. It is then selectable on a per-platform basis where the decision >>>> should be made. >>>> >>>> Given a separate discussion with Sekhar, we're only going to have one >>>> SRAM pool on any DaVinci part right now...this was only a question on >>>> L138 anyway. But regardless, the created gen_pool will be passed via >>>> pdata. >>> >>> I thought about this too, as mmp does it that way. >>> >>>> Since DT conversion is starting and we need to consider that now, >>>> the idea there is to use the DT-based generic sram driver [1] such that >>>> when we do boot from DT on Davinci, the genpool is provided via phandle >>>> and the pointer extracted with the OF helpers that are part of the >>>> series. >>> >>> A phandle is the cleanest way I think, yes. >>> >>>> That's pretty much it. I'm reworking the backend support as discussed >>>> with Sekhar wrt to my uio_pruss series. I can post a standalone series >>>> that just replaces sram_* with genalloc for davinci ASoC. >>> >>> As you can also test it, it would be easiest if you came up with a patch >>> for that, yes. I can have a look at the dma bits laters, once my OMAP >>> board finally works with the code as it currently stands. I'm still >>> fighting with the mcasp driver right now ... >> >> I quickly prepared two patches to change that, so that topic is out of >> the way. But I did only compile-test them on OMAP - could you check on >> your Davinci platform? Note that these apply on top of the patch in >> discussion here (which isn't applied to the asoc tree yet). > > I put a series together yesterday, just ran out of time to post > last night after testing. I'm posting that now...it's on top of my > uio_pruss/genalloc series and only addresss switching davinci-pcm to > genalloc (and actually enabling ping-pon from sram). Ok, I don't care which version makes it in after all :) > I'll take a look a your OMAP patches. The patches I just sent out are only for Davinci and change the SRAM functions to genalloc in the mcasp driver. They work just fine on OMAP of course, as the code is disabled in there. Daniel

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Re: [alsa-devel] [PATCH 2/3] ASoC: Davinci: pcm: add support for sram-support-less platforms
by Daniel Mack 04 Oct '12

04 Oct '12

On 02.10.2012 18:41, Matt Porter wrote: > On Tue, Oct 02, 2012 at 03:42:47PM +0200, Daniel Mack wrote: >> On 02.10.2012 13:06, Sekhar Nori wrote: >>> On 10/2/2012 4:03 PM, Daniel Mack wrote: >>>> On 02.10.2012 11:37, Mark Brown wrote: >>>>> On Tue, Oct 02, 2012 at 10:48:53AM +0300, Peter Ujfalusi wrote: >>>>> >>>>>> I also agree that ifdef is not a good solution. >>>>>> It is better to have this information passed as device_data and via DT it can >>>>>> be decided based on the compatible property for the device. >>>>> >>>>> That's not really the problem here - the problem is that the APIs used >>>>> to get the SRAM are DaVinci only so it's not possible to build on OMAP >>>>> or other platforms. The SRAM code needs to move to a standard API. >>>> >>>> What about following Matt Porter's idea and ignore the SRAM code >>>> entirely and port the entire PCM code to generic dmaengine code first? >>>> The EDMA driver needs to learn support for cyclic DMA for that, and I >>>> might give that a try in near future. >>>> >>>> Later on, the SRAM ping-pong code can get added back using genalloc >>>> functions, as Sekhar proposed. That needs to be done by someone who has >>>> access to a Davinci board though, I only have a AM33xx/OMAP here. >>> >>> We cannot "get rid" of SRAM code and add it back "later". It is required >>> for most DaVinci parts. The SRAM code can be converted to use genalloc >>> (conversion should be straightforward and I can help test it) and the >>> code that uses SRAM can probably keep using the private EDMA API till >>> the dmaengine EDMA driver has cyclic DMA support. Matt has already >>> posted patches to move private EDMA APIs to a common location. That >>> should keep AM335x build from breaking. Is this something that is feasible? >> >> Yes - by "later" I just meant in a subsequent patch. But you're probably >> right, we can also do that first. >> >> I'm looking at that right now and the problem seems that we don't have a >> sane way to dynamically look up gen_pools independently of the actual >> run-time platform. All users of gen_pools seem to know which platform >> they run on and access a platform-specific pool. So I don't currently >> see how we could implement multi-platform code, gen_pools are fine but >> don't solve the problem here. >> >> Would it be an idea to add a char* member to gen_pools and a function >> that can be used to dynamically look it up again? If a buffer with a >> certain name exists, we can use it and install that ping-pong buffer, >> otherwise we just don't. While that would be easy to do, it's a >> tree-wide change. >> >> Is there a better way that I miss? > > At the high level there's two platform models we have to handle, the > boot from board file !DT case, and then the boot from DT case. Since > Davinci is just starting DT conversion, we mostly care about the !DT > base in which the struct gen_pool * is passed in pdata to the ASoC > driver. It is then selectable on a per-platform basis where the decision > should be made. > > Given a separate discussion with Sekhar, we're only going to have one > SRAM pool on any DaVinci part right now...this was only a question on > L138 anyway. But regardless, the created gen_pool will be passed via > pdata. I thought about this too, as mmp does it that way. > Since DT conversion is starting and we need to consider that now, > the idea there is to use the DT-based generic sram driver [1] such that > when we do boot from DT on Davinci, the genpool is provided via phandle > and the pointer extracted with the OF helpers that are part of the > series. A phandle is the cleanest way I think, yes. > That's pretty much it. I'm reworking the backend support as discussed > with Sekhar wrt to my uio_pruss series. I can post a standalone series > that just replaces sram_* with genalloc for davinci ASoC. As you can also test it, it would be easiest if you came up with a patch for that, yes. I can have a look at the dma bits laters, once my OMAP board finally works with the code as it currently stands. I'm still fighting with the mcasp driver right now ... Daniel

1 1

[alsa-devel] [PATCH 0/2] ASoC: omap-mcpdm updates for 3.7
by Peter Ujfalusi 04 Oct '12

04 Oct '12

Hello Mark, Tony, The mcpdm driver no longer needs to include any plat/*.h file, clearing the way for the single kernel image. Regards, Peter --- Peter Ujfalusi (1): ASoC: omap-mcpdm: Remove OMAP revision check Tony Lindgren (1): ASoC: Fix wrong include for McPDM sound/soc/omap/omap-mcpdm.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) -- 1.7.12

1 3

[alsa-devel] ASoC: Fix wrong include for McPDM
by Tony Lindgren 04 Oct '12

04 Oct '12

McPDM needs platt/cpu.h for omap_rev and not omap_hwmod.h. Drivers must not include omap_hwmod.h at, it will be private to mach-omap2 soon. Fix the problem before other drivers will also start including omap_hwmod.h. Note that also plat/cpu.h will be going away, so the omap_rev check needs to be replaced with mcpdm-watchdog flag from platform_data or DT. Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- Can you guys please queue this as fix for -rc1? This removes and unnecessary dependency for making omap_hwmod.h local to mach-omap2. Then I'm hoping Peter will remove the plat/cpu.h dependency for v3.8 merge window ;) --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -40,7 +40,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <plat/omap_hwmod.h> +#include <plat/cpu.h> #include "omap-mcpdm.h" #include "omap-pcm.h"

2 1

[alsa-devel] [PATCH] ASoC: wm2200: Fix non-inverted OUT2 mute control
by Mark Brown 02 Oct '12

02 Oct '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 664067c..6ca9bcc 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1028,7 +1028,7 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_VOL_SHIFT, 0x9f, 0, digital_tlv), SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, - WM2200_SPK1R_MUTE_SHIFT, 1, 0), + WM2200_SPK1R_MUTE_SHIFT, 1, 1), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); -- 1.7.10.4

1 0

Re: [alsa-devel] [PATCH 2/3] ASoC: Davinci: pcm: add support for sram-support-less platforms
by Daniel Mack 02 Oct '12

02 Oct '12

On 02.10.2012 13:06, Sekhar Nori wrote: > On 10/2/2012 4:03 PM, Daniel Mack wrote: >> On 02.10.2012 11:37, Mark Brown wrote: >>> On Tue, Oct 02, 2012 at 10:48:53AM +0300, Peter Ujfalusi wrote: >>> >>>> I also agree that ifdef is not a good solution. >>>> It is better to have this information passed as device_data and via DT it can >>>> be decided based on the compatible property for the device. >>> >>> That's not really the problem here - the problem is that the APIs used >>> to get the SRAM are DaVinci only so it's not possible to build on OMAP >>> or other platforms. The SRAM code needs to move to a standard API. >> >> What about following Matt Porter's idea and ignore the SRAM code >> entirely and port the entire PCM code to generic dmaengine code first? >> The EDMA driver needs to learn support for cyclic DMA for that, and I >> might give that a try in near future. >> >> Later on, the SRAM ping-pong code can get added back using genalloc >> functions, as Sekhar proposed. That needs to be done by someone who has >> access to a Davinci board though, I only have a AM33xx/OMAP here. > > We cannot "get rid" of SRAM code and add it back "later". It is required > for most DaVinci parts. The SRAM code can be converted to use genalloc > (conversion should be straightforward and I can help test it) and the > code that uses SRAM can probably keep using the private EDMA API till > the dmaengine EDMA driver has cyclic DMA support. Matt has already > posted patches to move private EDMA APIs to a common location. That > should keep AM335x build from breaking. Is this something that is feasible? Yes - by "later" I just meant in a subsequent patch. But you're probably right, we can also do that first. I'm looking at that right now and the problem seems that we don't have a sane way to dynamically look up gen_pools independently of the actual run-time platform. All users of gen_pools seem to know which platform they run on and access a platform-specific pool. So I don't currently see how we could implement multi-platform code, gen_pools are fine but don't solve the problem here. Would it be an idea to add a char* member to gen_pools and a function that can be used to dynamically look it up again? If a buffer with a certain name exists, we can use it and install that ping-pong buffer, otherwise we just don't. While that would be easy to do, it's a tree-wide change. Is there a better way that I miss? Daniel

1 0

[alsa-devel] [PATCH] ASoC: wm2200: Convert to devm APIs
by Mark Brown 01 Oct '12

01 Oct '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 39 ++++++++++++++------------------------- 1 file changed, 14 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index efa93dbb..02ea49e 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2011,7 +2011,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, wm2200->dev = &i2c->dev; init_completion(&wm2200->fll_lock); - wm2200->regmap = regmap_init_i2c(i2c, &wm2200_regmap); + wm2200->regmap = devm_regmap_init_i2c(i2c, &wm2200_regmap); if (IS_ERR(wm2200->regmap)) { ret = PTR_ERR(wm2200->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2027,8 +2027,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, for (i = 0; i < ARRAY_SIZE(wm2200->core_supplies); i++) wm2200->core_supplies[i].supply = wm2200_core_supply_names[i]; - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); @@ -2044,8 +2045,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, } if (wm2200->pdata.ldo_ena) { - ret = gpio_request_one(wm2200->pdata.ldo_ena, - GPIOF_OUT_INIT_HIGH, "WM2200 LDOENA"); + ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.ldo_ena, + GPIOF_OUT_INIT_HIGH, + "WM2200 LDOENA"); if (ret < 0) { dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", wm2200->pdata.ldo_ena, ret); @@ -2055,8 +2057,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, } if (wm2200->pdata.reset) { - ret = gpio_request_one(wm2200->pdata.reset, - GPIOF_OUT_INIT_HIGH, "WM2200 /RESET"); + ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.reset, + GPIOF_OUT_INIT_HIGH, + "WM2200 /RESET"); if (ret < 0) { dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", wm2200->pdata.reset, ret); @@ -2165,23 +2168,16 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, err_pm_runtime: pm_runtime_disable(&i2c->dev); err_reset: - if (wm2200->pdata.reset) { + if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); - gpio_free(wm2200->pdata.reset); - } err_ldo: - if (wm2200->pdata.ldo_ena) { + if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); - gpio_free(wm2200->pdata.ldo_ena); - } err_enable: regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), wm2200->core_supplies); err_core: - regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); err_regmap: - regmap_exit(wm2200->regmap); err: return ret; } @@ -2193,17 +2189,10 @@ static __devexit int wm2200_i2c_remove(struct i2c_client *i2c) snd_soc_unregister_codec(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm2200); - if (wm2200->pdata.reset) { + if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); - gpio_free(wm2200->pdata.reset); - } - if (wm2200->pdata.ldo_ena) { + if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); - gpio_free(wm2200->pdata.ldo_ena); - } - regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); - regmap_exit(wm2200->regmap); return 0; } -- 1.7.10.4

1 0

Re: [alsa-devel] 0.2ms latency
by Pavel Roskin 01 Oct '12

01 Oct '12

Quoting Uwaysi Bin Kareem <uwaysi.bin.kareem(a)paradoxuncreated.com>: > Hi. I am aiming for 0.2ms latency with a HDA onboard soundcard. > > I recently did 0.3ms on a firewire card, with realtime streams. (which > is the lowest possible for firewire protocol.) > This does not work with the alsa HDA device. (some clicking) > Is this a weakness software-wise, that can be improved? > I also do notice that the driver (probably) seems to allow lower > latencies at 96k. Hello! This is the first time I got a message from this list. I subscribed to it about a month ago to report a kernel bug. I guess this mailing list is defunct and the discussion is happening elsewhere. -- Regards, Pavel Roskin -- 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

3 2

Re: [alsa-devel] [PATCH] ASoC: Revert 'ASoC: imx-ssi: Remove mono support'
by Mark Brown 01 Oct '12

01 Oct '12

On Mon, Oct 01, 2012 at 11:33:45AM +0200, javier Martin wrote: > Hi Mark, Fabio, > this patch should be applied as a fix to stable 3.6 too. Otherwise, > audio for Visstrim M10 will continue broken in 3.6. You need to tell the stable team about this, then.

1 0

[alsa-devel] ASoC updates for 3.7
by Mark Brown 01 Oct '12

01 Oct '12

The following changes since commit ddfb43f3881edb47aa0083651ad31983cdc42c33: Merge tag 'v3.6-rc6' into for-3.7 (2012-09-22 11:26:27 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.7 for you to fetch changes up to 9911f7f7562a25381eff93fdc660a4a3b4c0f6e0: ASoC: codecs: Add DA9055 codec driver (2012-09-28 14:21:54 +0100) ---------------------------------------------------------------- ASoC: Additional updates for v3.7 A couple more updates for 3.7, enhancements to the ux500 and wm2000 drivers, a new driver for DA9055 and the support for regulator bypass mode. With the exception of the DA9055 this has all had a chance to soak in -next (the driver was added on Friday so should be in -next today). ---------------------------------------------------------------- Ashish Chavan (1): ASoC: codecs: Add DA9055 codec driver Charles Keepax (1): ASoC: wm5110: Adding missing volume update bits Daniel Mack (1): ALSA: ASoC: add DT bindings for CS4271 Fabio Estevam (2): ASoC: cs4270: Remove mono support ASoC: eukrea-tlv320: Convert it to platform driver Lee Jones (7): ASoC: Ux500: Move MSP pinctrl setup into the MSP driver ASoC: Ux500: Enable MOP500 driver for Device Tree ASoC: Ux500: Enable ux500 MSP driver for Device Tree ASoC: codecs: Enable AB8500 CODEC for Device Tree ASoC: Ux500: Minor coding layout changes Documentation: Define the MOP500 Audio Machine Driver Device Tree bindings Documentation: Define the MSP Driver Device Tree bindings Mark Brown (22): regulator: core: Support bypass mode regulator: core: Provide regmap get/set bypass operations regulator: arizona-ldo: Support get/set bypass regulator: arizona-micsupp: Support get/set bypass regulator: wm831x-ldo: Add bypass support extcon: arizona: Use bypass mode for MICVDD ASoC: wm2000: Convert to devm_regmap_init_i2c() ASoC: wm2000: Add regulator support Merge remote-tracking branch 'asoc/topic/ux500' into for-3.7 Merge tag 'bypass' of git://git.kernel.org/.../broonie/regulator into for-3.7 ASoC: dapm: Allow regulators to bypass as well as disable when idle ASoC: wm5102: Enable bypass mode for MICVDD ASoC: wm5110: Enable bypass mode for MICVDD ASoC: wm0010: Allow slow GPIO for reset ASoC: wm0010: Don't check if reset GPIO is defined when removing ASoC: wm0010: Initialise chip state before we register the interrupt ASoC: arizona: Add more DSP options for mixer input muxes ASoC: arizona: Add more clock rates ASoC: wm5110: Rename EPOUT to HPOUT3 ASoC: wm5110: Add AEC loopback support ASoC: wm5110: Add OUT3R support ASoC: wm_hubs: Ensure volume updates are handled during class W startup Documentation/ABI/testing/sysfs-class-regulator | 21 + Documentation/devicetree/bindings/sound/cs4271.txt | 36 + .../devicetree/bindings/sound/ux500-mop500.txt | 39 + .../devicetree/bindings/sound/ux500-msp.txt | 43 + arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c | 1 + arch/arm/mach-ux500/board-mop500-msp.c | 79 +- arch/arm/mach-ux500/include/mach/msp.h | 2 - drivers/extcon/extcon-arizona.c | 5 + drivers/regulator/arizona-ldo1.c | 4 + drivers/regulator/arizona-micsupp.c | 5 + drivers/regulator/core.c | 126 ++ drivers/regulator/wm831x-ldo.c | 8 + include/linux/mfd/abx500/ab8500-codec.h | 6 +- include/linux/regulator/consumer.h | 8 + include/linux/regulator/driver.h | 14 + include/linux/regulator/machine.h | 2 + include/sound/da9055.h | 33 + include/sound/soc-dapm.h | 3 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ab8500-codec.c | 81 ++ sound/soc/codecs/arizona.c | 42 + sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/cs4270.c | 4 +- sound/soc/codecs/cs4271.c | 24 +- sound/soc/codecs/da9055.c | 1510 ++++++++++++++++++++ sound/soc/codecs/wm0010.c | 18 +- sound/soc/codecs/wm2000.c | 60 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 79 +- sound/soc/codecs/wm_hubs.c | 5 + sound/soc/fsl/eukrea-tlv320.c | 37 +- sound/soc/soc-dapm.c | 23 +- sound/soc/ux500/mop500.c | 47 +- sound/soc/ux500/ux500_msp_dai.c | 6 + sound/soc/ux500/ux500_msp_i2s.c | 89 +- sound/soc/ux500/ux500_msp_i2s.h | 8 +- 40 files changed, 2307 insertions(+), 174 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs4271.txt create mode 100644 Documentation/devicetree/bindings/sound/ux500-mop500.txt create mode 100644 Documentation/devicetree/bindings/sound/ux500-msp.txt create mode 100644 include/sound/da9055.h create mode 100644 sound/soc/codecs/da9055.c

1 0

Re: [alsa-devel] [PATCH] ASoC: fixup simple-card probe order
by Mark Brown 01 Oct '12

01 Oct '12

On Mon, Oct 01, 2012 at 09:44:37AM +0900, Do Q.Thang wrote: > Hi all, > > KZM-A9-GT board and Armadillo-800eva board use the asoc-simple-card. > but probe of this sound card needs the fsi-dai's registered before. > So this message appear in the bootlog. > > " asoc-simple-card asoc-simple-card.0: CPU DAI fsia-dai not registered > platform asoc-simple-card.0: Driver asoc-simple-card requests probe > deferral " > > The asoc-simple-card was probed while fsi-dai probe hasn't run yet. > So asoc-simple-card probe deferral is requested. > I think the simple-card should be probed after the fsi-dai probe. > I think this patch can fix this issue. > > What do you think about this? The above is normal and expected behaviour, there is no issue here.

1 0

[alsa-devel] [PATCH 1/2] ALSA: core: add hooks for audio timestamps
by Pierre-Louis Bossart 30 Sep '12

30 Sep '12

ALSA did not provide any direct means to infer the audio time for A/V sync and system/audio time correlations (eg. PulseAudio). Applications had to track the number of samples read/written and add/subtract the number of samples queued in the ring buffer. This accounting led to small errors, typically several samples, due to the two-step process. Computing the audio time in the kernel is more direct, as all the information is available in the same routines. Also add new .audio_wallclock routine to enable fine-grain synchronization between monotonic system time and audio hardware time. Using the wallclock, if supported in hardware, allows for a much better sub-microsecond precision and a common drift tracking for all devices sharing the same wall clock (master clock). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/asound.h | 7 +++++-- include/sound/pcm.h | 2 ++ sound/core/pcm_compat.c | 13 +++++++++++-- sound/core/pcm_lib.c | 29 +++++++++++++++++++++++++++-- sound/core/pcm_native.c | 2 ++ 5 files changed, 47 insertions(+), 6 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 0876a1e..2d1ba63 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -152,7 +152,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -274,6 +274,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ typedef int __bitwise snd_pcm_state_t; @@ -422,7 +423,8 @@ struct snd_pcm_status { snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ snd_pcm_state_t suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ + unsigned char reserved[60-sizeof(struct timespec)]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -431,6 +433,7 @@ struct snd_pcm_mmap_status { snd_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ snd_pcm_state_t suspended_state; /* RO: suspended stream state */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ }; struct snd_pcm_mmap_control { diff --git a/include/sound/pcm.h b/include/sound/pcm.h index cdca2ab..35c2bbf 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -71,6 +71,8 @@ struct snd_pcm_ops { int (*prepare)(struct snd_pcm_substream *substream); int (*trigger)(struct snd_pcm_substream *substream, int cmd); snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); + int (*wall_clock)(struct snd_pcm_substream *substream, + struct timespec *audio_ts); int (*copy)(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *buf, snd_pcm_uframes_t count); diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 91cdf94..ab4c953 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -190,7 +190,8 @@ struct snd_pcm_status32 { u32 avail_max; u32 overrange; s32 suspended_state; - unsigned char reserved[60]; + struct timespec audio_tstamp; + unsigned char reserved[60-sizeof(struct timespec)]; } __attribute__((packed)); @@ -215,7 +216,9 @@ static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, put_user(status.avail, &src->avail) || put_user(status.avail_max, &src->avail_max) || put_user(status.overrange, &src->overrange) || - put_user(status.suspended_state, &src->suspended_state)) + put_user(status.suspended_state, &src->suspended_state) || + put_user(status.audio_tstamp.tv_sec, &src->audio_tstamp.tv_sec) || + put_user(status.audio_tstamp.tv_nsec, &src->audio_tstamp.tv_nsec)) return -EFAULT; return err; @@ -364,6 +367,7 @@ struct snd_pcm_mmap_status32 { u32 hw_ptr; struct compat_timespec tstamp; s32 suspended_state; + struct compat_timespec audio_tstamp; } __attribute__((packed)); struct snd_pcm_mmap_control32 { @@ -426,12 +430,17 @@ static int snd_pcm_ioctl_sync_ptr_compat(struct snd_pcm_substream *substream, sstatus.hw_ptr = status->hw_ptr % boundary; sstatus.tstamp = status->tstamp; sstatus.suspended_state = status->suspended_state; + sstatus.audio_tstamp = status->audio_tstamp; snd_pcm_stream_unlock_irq(substream); if (put_user(sstatus.state, &src->s.status.state) || put_user(sstatus.hw_ptr, &src->s.status.hw_ptr) || put_user(sstatus.tstamp.tv_sec, &src->s.status.tstamp.tv_sec) || put_user(sstatus.tstamp.tv_nsec, &src->s.status.tstamp.tv_nsec) || put_user(sstatus.suspended_state, &src->s.status.suspended_state) || + put_user(sstatus.audio_tstamp.tv_sec, + &src->s.status.audio_tstamp.tv_sec) || + put_user(sstatus.audio_tstamp.tv_nsec, + &src->s.status.audio_tstamp.tv_nsec) || put_user(scontrol.appl_ptr, &src->c.control.appl_ptr) || put_user(scontrol.avail_min, &src->c.control.avail_min)) return -EFAULT; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7ae6719..ee0c931 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -315,6 +315,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, unsigned long jdelta; unsigned long curr_jiffies; struct timespec curr_tstamp; + struct timespec audio_tstamp; old_hw_ptr = runtime->status->hw_ptr; @@ -326,9 +327,14 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, */ pos = substream->ops->pointer(substream); curr_jiffies = jiffies; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); + if ((runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK) && + (substream->ops->wall_clock)) + substream->ops->wall_clock(substream, &audio_tstamp); + } + if (pos == SNDRV_PCM_POS_XRUN) { xrun(substream); return -EPIPE; @@ -506,8 +512,27 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = curr_jiffies; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { runtime->status->tstamp = curr_tstamp; + if (!(runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK)) { + /* + * no wall clock available, provide audio timestamp + * derived from pointer position+delay + */ + u64 audio_frames, audio_nsecs; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + audio_frames = runtime->status->hw_ptr + - runtime->delay; + else + audio_frames = runtime->status->hw_ptr + + runtime->delay; + audio_nsecs = audio_frames * 1000000000LL / + runtime->rate; + audio_tstamp = ns_to_timespec(audio_nsecs); + } + runtime->status->audio_tstamp = audio_tstamp; + } return snd_pcm_update_state(substream, runtime); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 53b5ada..c5206f4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -594,6 +594,8 @@ int snd_pcm_status(struct snd_pcm_substream *substream, snd_pcm_update_hw_ptr(substream); if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { status->tstamp = runtime->status->tstamp; + status->audio_tstamp = + runtime->status->audio_tstamp; goto _tstamp_end; } } -- 1.7.9.5

3 8

[alsa-devel] snd_had_intel turning off internal speakers by going to power state D3
by Jeff Robins 29 Sep '12

29 Sep '12

Hello, I have a Toshiba Satellite C655D with Mageia 2 and I have a problem where the internal speakers eventually stop producing sound, but the headphones still work. After doing a bit of research and running alsa-info (attached) I figured out that one of the parts of the sound card had gone into a D3 power state. Removing and then reloading the snd_hda_intel module fixed the problem, but then the sound would turn off again. I was able to find a post in alsa-users where someone else came up with a better solution using hda-verb to reset the power state: http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg28769.html I also found a thread in the gentoo forums where other people are having the same problem. One of them has kernel 3.5.0: http://forums.gentoo.org/viewtopic-t-932038.html?sid=2bee6543c63f2342057c71… I'm open to compiling my own alsa and helping out in any way I can. Please let me know what I can do. Mageia bug report for completeness: https://bugs.mageia.org/show_bug.cgi?id=7292 Thank you, Jeff

2 3

[alsa-devel] Problem with AC97 and WM9714 on S3C6410
by Andrey Gusakov 29 Sep '12

29 Sep '12

Hi. I'm working with mini6410 devkit/demoboard. Itry to move forward from old kernel to last samsung kernel from git. I run into interesting problem with ALSA and WM9714 codec. On kernel boot I see some activity on AC97 bus, but then BCLK and SYNC stops. As i figure out clock stops after write to WM9714 register 0x00. As I understand it's a reset comand. But it brokes all communication. No blayback or control regiter read/write posible after. And there is no reset or warm_reset commands after it. If i filter out all writes to register 0x00 in samsung AC97 driver, everything seems to work corect. But it's a dirty fix. I use dump_stack() to collect some aditional info. That what i got: Try to reset codec! (write at 0x00) [<c0014978>] (unwind_backtrace+0x0/0xf4) from [<c035caf0>] (ac97_write+0x4c/0x64) [<c035caf0>] (ac97_write+0x4c/0x64) from [<c034d00c>] (snd_soc_write+0xc/0x10) [<c034d00c>] (snd_soc_write+0xc/0x10) from [<c035355c>] (soc_widget_update_bits_locked+0x114/0x190) [<c035355c>] (soc_widget_update_bits_locked+0x114/0x190) from [<c0353830>] (dapm_seq_run_coalesced+0x114/0x16c) [<c0353830>] (dapm_seq_run_coalesced+0x114/0x16c) from [<c0353b88>] (dapm_seq_run+0x300/0x4a0) [<c0353b88>] (dapm_seq_run+0x300/0x4a0) from [<c0354be0>] (dapm_power_widgets+0x404/0x600) [<c0354be0>] (dapm_power_widgets+0x404/0x600) from [<c03577ec>] (snd_soc_dapm_new_widgets+0x2ac/0x6ac) [<c03577ec>] (snd_soc_dapm_new_widgets+0x2ac/0x6ac) from [<c034f940>] (soc_post_component_init+0x44/0x27c) [<c034f940>] (soc_post_component_init+0x44/0x27c) from [<c03513f4>] (snd_soc_register_card+0xc44/0x1a00) [<c03513f4>] (snd_soc_register_card+0xc44/0x1a00) from [<c03521e8>] (soc_probe+0x38/0x64) [<c03521e8>] (soc_probe+0x38/0x64) from [<c025f9d8>] (platform_drv_probe+0x18/0x1c) [<c025f9d8>] (platform_drv_probe+0x18/0x1c) from [<c025e518>] (driver_probe_device+0x78/0x1fc) [<c025e518>] (driver_probe_device+0x78/0x1fc) from [<c025cd9c>] (bus_for_each_drv+0x60/0x8c) [<c025cd9c>] (bus_for_each_drv+0x60/0x8c) from [<c025e810>] (device_attach+0x80/0x98) [<c025e810>] (device_attach+0x80/0x98) from [<c025d8bc>] (bus_probe_device+0x84/0xac) [<c025d8bc>] (bus_probe_device+0x84/0xac) from [<c025c6ec>] (device_add+0x5b0/0x67c) [<c025c6ec>] (device_add+0x5b0/0x67c) from [<c025ffa8>] (platform_device_add+0xf4/0x20c) [<c025ffa8>] (platform_device_add+0xf4/0x20c) from [<c060b3a4>] (mini6410_init+0x60/0xa0) [<c060b3a4>] (mini6410_init+0x60/0xa0) from [<c00085d0>] (do_one_initcall+0x34/0x178) [<c00085d0>] (do_one_initcall+0x34/0x178) from [<c05f3310>] (kernel_init+0xe0/0x1ac) [<c05f3310>] (kernel_init+0xe0/0x1ac) from [<c000f848>] (kernel_thread_exit+0x0/0x8) I'm begginer in kernel programing and have no enouth experience to figure what's happening. Can any one help with this problem? Thank you.

1 0

[alsa-devel] imx28 pcm issue
by George Stefan 29 Sep '12

29 Sep '12

Hi! I have an issues with a modem connected to the SAIF port of an imx28. I have the following configuration running on the modem: -modem is slave(but still have to fix the clock at 256 khz) -"short frame" sync. PCM data is always formatted as 16-bit uncompressed two’s complement. Also, all PCM data and frame synchronization signals are written to the PCM bus on the rising clock edge and read on the falling edge. - 16 bits liniar sample, MSB first. On the imx side i have the following configuration priv->sysclk = 512 * rate; snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSP_SYS_MCLK, 32); dai_format = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM; snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, priv->sysclk, SND_SOC_CLOCK_OUT); The modem has the 2 options for testing, either outputing a tone or an audio loop. My problem is that i cannot receive anything else but noise. arecord -f S16_LE -c2 -r8000 test.wav hexdump -C test.wav 00000000 52 49 46 46 24 c4 09 00 57 41 56 45 66 6d 74 20 |RIFF$...WAVEfmt | 00000010 10 00 00 00 01 00 02 00 40 1f 00 00 00 7d 00 00 |........@....}..| 00000020 04 00 10 00 64 61 74 61 00 c4 09 00 00 00 00 00 |....data........| 00000030 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 |................| * 00040f10 00 00 00 00 f0 03 00 00 00 00 00 00 00 00 00 00 |................| 00040f20 00 00 0f 00 ff ff ff ff ff ff f0 ff fc 00 00 00 |................| 00040f30 00 00 00 00 00 00 00 00 00 00 ff 01 ff ff ff ff |................| 00040f40 00 fc 03 00 00 e0 00 00 00 00 00 00 00 00 00 00 |................| 00040f50 00 00 ff 1f ff ff ff ff 00 ff ff 1f 00 f8 00 00 |................| 00040f60 00 00 00 00 00 00 00 00 03 00 ff ff ff ff ff ff |................| 00040f70 81 ff ff ff 00 00 00 00 00 00 00 00 00 00 00 00 |................| 00040f80 3f 00 ff ff ff ff 80 ff 00 00 00 00 00 00 00 00 |?...............| When i select the audio loop mode, play a wav sample, and record it at the same time, i hear noise every time i play that wav and silence between them(so something is moving out there) Do you have any suggestions on why this is happening? Regards.

1 0

[alsa-devel] imx28 pcm issues
by George Stefan 29 Sep '12

29 Sep '12

Hi! I have an issues with a modem connected to the SAIF port of an imx28. I have the following configuration running on the modem: -modem is slave(but still have to fix the clock at 256 khz)

1 0

[alsa-devel] Delay calculations for USB audio?
by David Henningsson 28 Sep '12

28 Sep '12

Hi, While researching some crackling audio problem, I noticed that I got these messages in my dmesg: delay: estimated 0, actual 384 delay: estimated 384, actual 768 They come at every start of playback. I want to understand this stuff, so I tried to dig a little. It looks like the endpoint is started in the prepare stage, so between prepare and trigger, silence data will be sent. First question: Isn't this messing up the delay calculations? If you fill some "urb buffer" [1] with silence, depending on the urb buffer size, delay calculation should take this into account for proper lipsync. I mean, the USB audio device would first have to play all the silence data before it could start the audio data, right? (And how big is this buffer anyway?) Second question: This also means that some of the silence packets will go through retire_playback_urb, right? Isn't this also messing with the delay calculation? (Oh, and btw, if you have an idea of the crackling/distorted USB audio, let me know. I'm using a standard Logitech USB Headset, using speaker-test -c 2 -D plughw:Headset -t sine -r 48000 I can make the problem go away using any of these options: * Booting a 3.2 kernel - I've tried 3.5, 3.6-rc7 and 3.2 * Choosing the 44100 sample rate instead of the default of 48000 * Selecting the outer USB port instead of the inner one. Yes, there are two USB ports next to each other, and this only happens on one of them. (!)) -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic [1] maybe there's a better name for it?

4 5

[alsa-devel] [GIT PULL] MOP500 Audio patches due for ASoC
by Lee Jones 28 Sep '12

28 Sep '12

The following changes since commit 5698bd757d55b1bb87edd1a9744ab09c142abfc2: Linux 3.6-rc6 (2012-09-16 14:58:51 -0700) are available in the git repository at: git://git.linaro.org/people/ljones/linux-3.0-ux500.git for-asoc-next for you to fetch changes up to b4cad7af6665743647f28119b689e1552326d4da: Documentation: Define the MSP Driver Device Tree bindings (2012-09-20 09:10:59 +0200) ---------------------------------------------------------------- Lee Jones (7): ASoC: Ux500: Move MSP pinctrl setup into the MSP driver ASoC: Ux500: Enable MOP500 driver for Device Tree ASoC: Ux500: Enable ux500 MSP driver for Device Tree ASoC: codecs: Enable AB8500 CODEC for Device Tree ASoC: Ux500: Minor coding layout changes Documentation: Define the MOP500 Audio Machine Driver Device Tree bindings Documentation: Define the MSP Driver Device Tree bindings .../devicetree/bindings/sound/ux500-mop500.txt | 39 +++++++++ .../devicetree/bindings/sound/ux500-msp.txt | 43 ++++++++++ arch/arm/mach-ux500/board-mop500-msp.c | 79 +---------------- arch/arm/mach-ux500/include/mach/msp.h | 2 - include/linux/mfd/abx500/ab8500-codec.h | 6 +- sound/soc/codecs/ab8500-codec.c | 81 ++++++++++++++++++ sound/soc/ux500/mop500.c | 47 +++++++++-- sound/soc/ux500/ux500_msp_dai.c | 6 ++ sound/soc/ux500/ux500_msp_i2s.c | 89 +++++++++++++++----- sound/soc/ux500/ux500_msp_i2s.h | 8 +- 10 files changed, 292 insertions(+), 108 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ux500-mop500.txt create mode 100644 Documentation/devicetree/bindings/sound/ux500-msp.txt

2 12

[alsa-devel] [PATCH v3] ASoC: codecs: Add DA9055 codec driver
by Ashish Chavan 28 Sep '12

28 Sep '12

This patch adds support for Dialog semiconductor's DA9055 audio codec. This has been tested on DA9055 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- Changes since v2: - Added platform data support for Mic bias voltage and source - Used DAPM for DAC source selection - Other minor improvements --- Changes since v1: - Made extensive use of DAPM throughout the driver - Removed LDO, DMIC and other non used controls - Corrected names for many controls - Other minor improvements --- include/sound/da9055.h | 33 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da9055.c | 1510 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1549 insertions(+), 0 deletions(-) diff --git a/include/sound/da9055.h b/include/sound/da9055.h new file mode 100644 index 0000000..cf1241b --- /dev/null +++ b/include/sound/da9055.h @@ -0,0 +1,33 @@ +/* + * DA9055 ALSA Soc codec driver + * + * Copyright (c) 2012 Dialog Semiconductor + * + * Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C + * Written by David Chen <david.chen(a)diasemi.com> and + * Ashish Chavan <ashish.chavan(a)kpitcummins.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __SOUND_DA9055_H__ +#define __SOUND_DA9055_H__ + +enum da9055_micbias_voltage { + DA9055_MICBIAS_1_6V = 0, + DA9055_MICBIAS_1_8V = 1, + DA9055_MICBIAS_2_1V = 2, + DA9055_MICBIAS_2_2V = 3, +}; + +struct da9055_platform_data { + /* Selects which of the two MicBias pins acts as the bias source */ + bool micbias_source; + /* Selects the micbias voltage */ + enum da9055_micbias_voltage micbias; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3684255..b92759a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DA732X if I2C + select SND_SOC_DA9055 if I2C select SND_SOC_DFBMCS320 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC @@ -239,6 +240,9 @@ config SND_SOC_DA7210 config SND_SOC_DA732X tristate +config SND_SOC_DA9055 + tristate + config SND_SOC_DFBMCS320 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ca508b2..9bd4d95 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da732x-objs := da732x.o +snd-soc-da9055-objs := da9055.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o @@ -144,6 +145,7 @@ obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o +obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c new file mode 100644 index 0000000..185d8dd --- /dev/null +++ b/sound/soc/codecs/da9055.c @@ -0,0 +1,1510 @@ +/* + * DA9055 ALSA Soc codec driver + * + * Copyright (c) 2012 Dialog Semiconductor + * + * Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C + * Written by David Chen <david.chen(a)diasemi.com> and + * Ashish Chavan <ashish.chavan(a)kpitcummins.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/da9055.h> + +/* DA9055 register space */ + +/* Status Registers */ +#define DA9055_STATUS1 0x02 +#define DA9055_PLL_STATUS 0x03 +#define DA9055_AUX_L_GAIN_STATUS 0x04 +#define DA9055_AUX_R_GAIN_STATUS 0x05 +#define DA9055_MIC_L_GAIN_STATUS 0x06 +#define DA9055_MIC_R_GAIN_STATUS 0x07 +#define DA9055_MIXIN_L_GAIN_STATUS 0x08 +#define DA9055_MIXIN_R_GAIN_STATUS 0x09 +#define DA9055_ADC_L_GAIN_STATUS 0x0A +#define DA9055_ADC_R_GAIN_STATUS 0x0B +#define DA9055_DAC_L_GAIN_STATUS 0x0C +#define DA9055_DAC_R_GAIN_STATUS 0x0D +#define DA9055_HP_L_GAIN_STATUS 0x0E +#define DA9055_HP_R_GAIN_STATUS 0x0F +#define DA9055_LINE_GAIN_STATUS 0x10 + +/* System Initialisation Registers */ +#define DA9055_CIF_CTRL 0x20 +#define DA9055_DIG_ROUTING_AIF 0X21 +#define DA9055_SR 0x22 +#define DA9055_REFERENCES 0x23 +#define DA9055_PLL_FRAC_TOP 0x24 +#define DA9055_PLL_FRAC_BOT 0x25 +#define DA9055_PLL_INTEGER 0x26 +#define DA9055_PLL_CTRL 0x27 +#define DA9055_AIF_CLK_MODE 0x28 +#define DA9055_AIF_CTRL 0x29 +#define DA9055_DIG_ROUTING_DAC 0x2A +#define DA9055_ALC_CTRL1 0x2B + +/* Input - Gain, Select and Filter Registers */ +#define DA9055_AUX_L_GAIN 0x30 +#define DA9055_AUX_R_GAIN 0x31 +#define DA9055_MIXIN_L_SELECT 0x32 +#define DA9055_MIXIN_R_SELECT 0x33 +#define DA9055_MIXIN_L_GAIN 0x34 +#define DA9055_MIXIN_R_GAIN 0x35 +#define DA9055_ADC_L_GAIN 0x36 +#define DA9055_ADC_R_GAIN 0x37 +#define DA9055_ADC_FILTERS1 0x38 +#define DA9055_MIC_L_GAIN 0x39 +#define DA9055_MIC_R_GAIN 0x3A + +/* Output - Gain, Select and Filter Registers */ +#define DA9055_DAC_FILTERS5 0x40 +#define DA9055_DAC_FILTERS2 0x41 +#define DA9055_DAC_FILTERS3 0x42 +#define DA9055_DAC_FILTERS4 0x43 +#define DA9055_DAC_FILTERS1 0x44 +#define DA9055_DAC_L_GAIN 0x45 +#define DA9055_DAC_R_GAIN 0x46 +#define DA9055_CP_CTRL 0x47 +#define DA9055_HP_L_GAIN 0x48 +#define DA9055_HP_R_GAIN 0x49 +#define DA9055_LINE_GAIN 0x4A +#define DA9055_MIXOUT_L_SELECT 0x4B +#define DA9055_MIXOUT_R_SELECT 0x4C + +/* System Controller Registers */ +#define DA9055_SYSTEM_MODES_INPUT 0x50 +#define DA9055_SYSTEM_MODES_OUTPUT 0x51 + +/* Control Registers */ +#define DA9055_AUX_L_CTRL 0x60 +#define DA9055_AUX_R_CTRL 0x61 +#define DA9055_MIC_BIAS_CTRL 0x62 +#define DA9055_MIC_L_CTRL 0x63 +#define DA9055_MIC_R_CTRL 0x64 +#define DA9055_MIXIN_L_CTRL 0x65 +#define DA9055_MIXIN_R_CTRL 0x66 +#define DA9055_ADC_L_CTRL 0x67 +#define DA9055_ADC_R_CTRL 0x68 +#define DA9055_DAC_L_CTRL 0x69 +#define DA9055_DAC_R_CTRL 0x6A +#define DA9055_HP_L_CTRL 0x6B +#define DA9055_HP_R_CTRL 0x6C +#define DA9055_LINE_CTRL 0x6D +#define DA9055_MIXOUT_L_CTRL 0x6E +#define DA9055_MIXOUT_R_CTRL 0x6F + +/* Configuration Registers */ +#define DA9055_LDO_CTRL 0x90 +#define DA9055_IO_CTRL 0x91 +#define DA9055_GAIN_RAMP_CTRL 0x92 +#define DA9055_MIC_CONFIG 0x93 +#define DA9055_PC_COUNT 0x94 +#define DA9055_CP_VOL_THRESHOLD1 0x95 +#define DA9055_CP_DELAY 0x96 +#define DA9055_CP_DETECTOR 0x97 +#define DA9055_AIF_OFFSET 0x98 +#define DA9055_DIG_CTRL 0x99 +#define DA9055_ALC_CTRL2 0x9A +#define DA9055_ALC_CTRL3 0x9B +#define DA9055_ALC_NOISE 0x9C +#define DA9055_ALC_TARGET_MIN 0x9D +#define DA9055_ALC_TARGET_MAX 0x9E +#define DA9055_ALC_GAIN_LIMITS 0x9F +#define DA9055_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA9055_ALC_ANTICLIP_CTRL 0xA1 +#define DA9055_ALC_ANTICLIP_LEVEL 0xA2 +#define DA9055_ALC_OFFSET_OP2M_L 0xA6 +#define DA9055_ALC_OFFSET_OP2U_L 0xA7 +#define DA9055_ALC_OFFSET_OP2M_R 0xAB +#define DA9055_ALC_OFFSET_OP2U_R 0xAC +#define DA9055_ALC_CIC_OP_LVL_CTRL 0xAD +#define DA9055_ALC_CIC_OP_LVL_DATA 0xAE +#define DA9055_DAC_NG_SETUP_TIME 0xAF +#define DA9055_DAC_NG_OFF_THRESHOLD 0xB0 +#define DA9055_DAC_NG_ON_THRESHOLD 0xB1 +#define DA9055_DAC_NG_CTRL 0xB2 + +/* SR bit fields */ +#define DA9055_SR_8000 (0x1 << 0) +#define DA9055_SR_11025 (0x2 << 0) +#define DA9055_SR_12000 (0x3 << 0) +#define DA9055_SR_16000 (0x5 << 0) +#define DA9055_SR_22050 (0x6 << 0) +#define DA9055_SR_24000 (0x7 << 0) +#define DA9055_SR_32000 (0x9 << 0) +#define DA9055_SR_44100 (0xA << 0) +#define DA9055_SR_48000 (0xB << 0) +#define DA9055_SR_88200 (0xE << 0) +#define DA9055_SR_96000 (0xF << 0) + +/* REFERENCES bit fields */ +#define DA9055_BIAS_EN (1 << 3) +#define DA9055_VMID_EN (1 << 7) + +/* PLL_CTRL bit fields */ +#define DA9055_PLL_INDIV_10_20_MHZ (1 << 2) +#define DA9055_PLL_SRM_EN (1 << 6) +#define DA9055_PLL_EN (1 << 7) + +/* AIF_CLK_MODE bit fields */ +#define DA9055_AIF_BCLKS_PER_WCLK_32 (0 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_64 (1 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_128 (2 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_256 (3 << 0) +#define DA9055_AIF_CLK_EN_SLAVE_MODE (0 << 7) +#define DA9055_AIF_CLK_EN_MASTER_MODE (1 << 7) + +/* AIF_CTRL bit fields */ +#define DA9055_AIF_FORMAT_I2S_MODE (0 << 0) +#define DA9055_AIF_FORMAT_LEFT_J (1 << 0) +#define DA9055_AIF_FORMAT_RIGHT_J (2 << 0) +#define DA9055_AIF_WORD_S16_LE (0 << 2) +#define DA9055_AIF_WORD_S20_3LE (1 << 2) +#define DA9055_AIF_WORD_S24_LE (2 << 2) +#define DA9055_AIF_WORD_S32_LE (3 << 2) + +/* MIXIN_L_CTRL bit fields */ +#define DA9055_MIXIN_L_MIX_EN (1 << 3) + +/* MIXIN_R_CTRL bit fields */ +#define DA9055_MIXIN_R_MIX_EN (1 << 3) + +/* ADC_L_CTRL bit fields */ +#define DA9055_ADC_L_EN (1 << 7) + +/* ADC_R_CTRL bit fields */ +#define DA9055_ADC_R_EN (1 << 7) + +/* DAC_L_CTRL bit fields */ +#define DA9055_DAC_L_MUTE_EN (1 << 6) + +/* DAC_R_CTRL bit fields */ +#define DA9055_DAC_R_MUTE_EN (1 << 6) + +/* HP_L_CTRL bit fields */ +#define DA9055_HP_L_AMP_OE (1 << 3) + +/* HP_R_CTRL bit fields */ +#define DA9055_HP_R_AMP_OE (1 << 3) + +/* LINE_CTRL bit fields */ +#define DA9055_LINE_AMP_OE (1 << 3) + +/* MIXOUT_L_CTRL bit fields */ +#define DA9055_MIXOUT_L_MIX_EN (1 << 3) + +/* MIXOUT_R_CTRL bit fields */ +#define DA9055_MIXOUT_R_MIX_EN (1 << 3) + +/* MIC bias select bit fields */ +#define DA9055_MICBIAS2_EN (1 << 6) + +/* ALC_CIC_OP_LEVEL_CTRL bit fields */ +#define DA9055_ALC_DATA_MIDDLE (2 << 0) +#define DA9055_ALC_DATA_TOP (3 << 0) +#define DA9055_ALC_CIC_OP_CHANNEL_LEFT (0 << 7) +#define DA9055_ALC_CIC_OP_CHANNEL_RIGHT (1 << 7) + +#define DA9055_AIF_BCLK_MASK (3 << 0) +#define DA9055_AIF_CLK_MODE_MASK (1 << 7) +#define DA9055_AIF_FORMAT_MASK (3 << 0) +#define DA9055_AIF_WORD_LENGTH_MASK (3 << 2) +#define DA9055_GAIN_RAMPING_EN (1 << 5) +#define DA9055_MICBIAS_LEVEL_MASK (3 << 4) + +#define DA9055_ALC_OFFSET_15_8 0x00FF00 +#define DA9055_ALC_OFFSET_17_16 0x030000 +#define DA9055_ALC_AVG_ITERATIONS 5 + +struct pll_div { + int fref; + int fout; + u8 frac_top; + u8 frac_bot; + u8 integer; + u8 mode; /* 0 = slave, 1 = master */ +}; + +/* PLL divisor table */ +static const struct pll_div da9055_pll_div[] = { + /* for MASTER mode, fs = 44.1Khz and its harmonics */ + {11289600, 2822400, 0x00, 0x00, 0x20, 1}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x03, 0x61, 0x1E, 1}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x0C, 0xCC, 0x1D, 1}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x19, 0x45, 0x1B, 1}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x18, 0x56, 0x1A, 1}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x02, 0xD0, 0x19, 1}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x1A, 0x1C, 0x12, 1}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x0B, 0x6D, 0x12, 1}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x07, 0xDD, 0x12, 1}, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz and its harmonics */ + {11289600, 3072000, 0x1A, 0x8E, 0x22, 1}, /* MCLK=11.2896Mhz */ + {12000000, 3072000, 0x18, 0x93, 0x20, 1}, /* MCLK=12Mhz */ + {12288000, 3072000, 0x00, 0x00, 0x20, 1}, /* MCLK=12.288Mhz */ + {13000000, 3072000, 0x07, 0xEA, 0x1E, 1}, /* MCLK=13Mhz */ + {13500000, 3072000, 0x04, 0x11, 0x1D, 1}, /* MCLK=13.5Mhz */ + {14400000, 3072000, 0x09, 0xD0, 0x1B, 1}, /* MCLK=14.4Mhz */ + {19200000, 3072000, 0x0F, 0x5C, 0x14, 1}, /* MCLK=19.2Mhz */ + {19680000, 3072000, 0x1F, 0x60, 0x13, 1}, /* MCLK=19.68Mhz */ + {19800000, 3072000, 0x1B, 0x80, 0x13, 1}, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + {11289600, 2822400, 0x0D, 0x47, 0x21, 0}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x0D, 0xFA, 0x1F, 0}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x16, 0x66, 0x1E, 0}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x00, 0x98, 0x1D, 0}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x1E, 0x33, 0x1B, 0}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x06, 0x50, 0x1A, 0}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x14, 0xBC, 0x13, 0}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x05, 0x66, 0x13, 0}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x01, 0xAE, 0x13, 0}, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA9055_CLKSRC_MCLK +}; + +/* Gain and Volume */ + +static const unsigned int aux_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(-5400, 0, 0), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int digital_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -78dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7800, 75, 0) +}; + +static const unsigned int alc_analog_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* 0dB to 36dB */ + 0x01, 0x07, TLV_DB_SCALE_ITEM(0, 600, 0) +}; + +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(lineout_vol_tlv, -4800, 100, 0); +static const DECLARE_TLV_DB_SCALE(alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_gain_tlv, 0, 600, 0); + +/* ADC and DAC high pass filter cutoff value */ +static const char * const da9055_hpf_cutoff_txt[] = { + "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" +}; + +static const struct soc_enum da9055_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +static const struct soc_enum da9055_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +/* ADC and DAC voice mode (8kHz) high pass cutoff value */ +static const char * const da9055_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da9055_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +static const struct soc_enum da9055_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +/* Gain ramping rate value */ +static const char * const da9055_gain_ramping_txt[] = { + "nominal rate", "nominal rate * 4", "nominal rate * 8", + "nominal rate / 8" +}; + +static const struct soc_enum da9055_gain_ramping_rate = + SOC_ENUM_SINGLE(DA9055_GAIN_RAMP_CTRL, 0, 4, da9055_gain_ramping_txt); + +/* DAC noise gate setup time value */ +static const char * const da9055_dac_ng_setup_time_txt[] = { + "256 samples", "512 samples", "1024 samples", "2048 samples" +}; + +static const struct soc_enum da9055_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 0, 4, + da9055_dac_ng_setup_time_txt); + +/* DAC noise gate rampup rate value */ +static const char * const da9055_dac_ng_rampup_txt[] = { + "0.02 ms/dB", "0.16 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 2, 2, + da9055_dac_ng_rampup_txt); + +/* DAC noise gate rampdown rate value */ +static const char * const da9055_dac_ng_rampdown_txt[] = { + "0.64 ms/dB", "20.48 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 3, 2, + da9055_dac_ng_rampdown_txt); + +/* DAC soft mute rate value */ +static const char * const da9055_dac_soft_mute_rate_txt[] = { + "1", "2", "4", "8", "16", "32", "64" +}; + +static const struct soc_enum da9055_dac_soft_mute_rate = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS5, 4, 7, + da9055_dac_soft_mute_rate_txt); + +/* DAC routing select */ +static const char * const da9055_dac_src_txt[] = { + "ADC output left", "ADC output right", "AIF input left", + "AIF input right" +}; + +static const struct soc_enum da9055_dac_l_src = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 0, 4, da9055_dac_src_txt); + +static const struct soc_enum da9055_dac_r_src = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 4, 4, da9055_dac_src_txt); + +/* MIC PGA Left source select */ +static const char * const da9055_mic_l_src_txt[] = { + "MIC1_P_N", "MIC1_P", "MIC1_N", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_l_src = + SOC_ENUM_SINGLE(DA9055_MIXIN_L_SELECT, 4, 4, da9055_mic_l_src_txt); + +/* MIC PGA Right source select */ +static const char * const da9055_mic_r_src_txt[] = { + "MIC2_R_L", "MIC2_R", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_r_src = + SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 4, 3, da9055_mic_r_src_txt); + +/* ALC Input Signal Tracking rate select */ +static const char * const da9055_signal_tracking_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da9055_integ_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 4, 4, + da9055_signal_tracking_rate_txt); + +static const struct soc_enum da9055_integ_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 6, 4, + da9055_signal_tracking_rate_txt); + +/* ALC Attack Rate select */ +static const char * const da9055_attack_rate_txt[] = { + "44/fs", "88/fs", "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", + "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 0, 13, da9055_attack_rate_txt); + +/* ALC Release Rate select */ +static const char * const da9055_release_rate_txt[] = { + "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", "5632/fs", + "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 4, 11, da9055_release_rate_txt); + +/* ALC Hold Time select */ +static const char * const da9055_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da9055_hold_time = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 0, 16, da9055_hold_time_txt); + +static int da9055_get_alc_data(struct snd_soc_codec *codec, u8 reg_val) +{ + int mid_data, top_data; + int sum = 0; + u8 iteration; + + for (iteration = 0; iteration < DA9055_ALC_AVG_ITERATIONS; + iteration++) { + /* Select the left or right channel and capture data */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, reg_val); + + /* Select middle 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_MIDDLE); + mid_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + /* Select top 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_TOP); + top_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + sum += ((mid_data << 8) | (top_data << 16)); + } + + return sum / DA9055_ALC_AVG_ITERATIONS; +} + +static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 reg_val, adc_left, adc_right; + int avg_left_data, avg_right_data, offset_l, offset_r; + + if (ucontrol->value.integer.value[0]) { + /* + * While enabling ALC (or ALC sync mode), calibration of the DC + * offsets must be done first + */ + + /* Save current values from ADC control registers */ + adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); + adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); + + /* Enable ADC Left and Right */ + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_ADC_L_EN, DA9055_ADC_L_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_ADC_R_EN, DA9055_ADC_R_EN); + + /* Calculate average for Left and Right data */ + /* Left Data */ + avg_left_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_LEFT); + /* Right Data */ + avg_right_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_RIGHT); + + /* Calculate DC offset */ + offset_l = -avg_left_data; + offset_r = -avg_right_data; + + reg_val = (offset_l & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_L, reg_val); + reg_val = (offset_l & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_L, reg_val); + + reg_val = (offset_r & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_R, reg_val); + reg_val = (offset_r & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_R, reg_val); + + /* Restore original values of ADC control registers */ + snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); + snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +static const struct snd_kcontrol_new da9055_snd_controls[] = { + + /* Volume controls */ + SOC_DOUBLE_R_TLV("Mic Volume", + DA9055_MIC_L_GAIN, DA9055_MIC_R_GAIN, + 0, 0x7, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux Volume", + DA9055_AUX_L_GAIN, DA9055_AUX_R_GAIN, + 0, 0x3f, 0, aux_vol_tlv), + SOC_DOUBLE_R_TLV("Mixin PGA Volume", + DA9055_MIXIN_L_GAIN, DA9055_MIXIN_R_GAIN, + 0, 0xf, 0, mixin_gain_tlv), + SOC_DOUBLE_R_TLV("ADC Volume", + DA9055_ADC_L_GAIN, DA9055_ADC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + + SOC_DOUBLE_R_TLV("DAC Volume", + DA9055_DAC_L_GAIN, DA9055_DAC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + SOC_DOUBLE_R_TLV("Headphone Volume", + DA9055_HP_L_GAIN, DA9055_HP_R_GAIN, + 0, 0x3f, 0, hp_vol_tlv), + SOC_SINGLE_TLV("Lineout Volume", DA9055_LINE_GAIN, 0, 0x3f, 0, + lineout_vol_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA9055_DAC_FILTERS4, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA9055_DAC_FILTERS2, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA9055_DAC_FILTERS2, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA9055_DAC_FILTERS3, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA9055_DAC_FILTERS3, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA9055_DAC_FILTERS4, 0, 0xf, 0, + eq_gain_tlv), + + /* High Pass Filter and Voice Mode controls */ + SOC_SINGLE("ADC HPF Switch", DA9055_ADC_FILTERS1, 7, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da9055_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA9055_ADC_FILTERS1, 3, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da9055_adc_vf_cutoff), + + SOC_SINGLE("DAC HPF Switch", DA9055_DAC_FILTERS1, 7, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da9055_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA9055_DAC_FILTERS1, 3, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da9055_dac_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Switch", DA9055_MIC_L_CTRL, + DA9055_MIC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Aux Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Mixin PGA Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("ADC Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Headphone Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 6, 1, 0), + SOC_SINGLE("Lineout Switch", DA9055_LINE_CTRL, 6, 1, 0), + SOC_SINGLE("DAC Soft Mute Switch", DA9055_DAC_FILTERS5, 7, 1, 0), + SOC_ENUM("DAC Soft Mute Rate", da9055_dac_soft_mute_rate), + + /* Zero Cross controls */ + SOC_DOUBLE_R("Aux ZC Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("Mixin PGA ZC Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("Headphone ZC Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 4, 1, 0), + SOC_SINGLE("Lineout ZC Switch", DA9055_LINE_CTRL, 4, 1, 0), + + /* Gain Ramping controls */ + SOC_DOUBLE_R("Aux Gain Ramping Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("Mixin Gain Ramping Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("ADC Gain Ramping Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("DAC Gain Ramping Switch", DA9055_DAC_L_CTRL, + DA9055_DAC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("Headphone Gain Ramping Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 5, 1, 0), + SOC_SINGLE("Lineout Gain Ramping Switch", DA9055_LINE_CTRL, 5, 1, 0), + SOC_ENUM("Gain Ramping Rate", da9055_gain_ramping_rate), + + /* DAC Noise Gate controls */ + SOC_SINGLE("DAC NG Switch", DA9055_DAC_NG_CTRL, 7, 1, 0), + SOC_SINGLE("DAC NG ON Threshold", DA9055_DAC_NG_ON_THRESHOLD, + 0, 0x7, 0), + SOC_SINGLE("DAC NG OFF Threshold", DA9055_DAC_NG_OFF_THRESHOLD, + 0, 0x7, 0), + SOC_ENUM("DAC NG Setup Time", da9055_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da9055_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da9055_dac_ng_rampdown_rate), + + /* DAC Invertion control */ + SOC_SINGLE("DAC Left Invert", DA9055_DIG_CTRL, 3, 1, 0), + SOC_SINGLE("DAC Right Invert", DA9055_DIG_CTRL, 7, 1, 0), + + /* DMIC controls */ + SOC_DOUBLE_R("DMIC Switch", DA9055_MIXIN_L_SELECT, + DA9055_MIXIN_R_SELECT, 7, 1, 0), + + /* ALC Controls */ + SOC_DOUBLE_EXT("ALC Switch", DA9055_ALC_CTRL1, 3, 7, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE_EXT("ALC Sync Mode Switch", DA9055_ALC_CTRL1, 1, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE("ALC Offset Switch", DA9055_ALC_CTRL1, 0, 1, 0), + SOC_SINGLE("ALC Anticlip Mode Switch", DA9055_ALC_ANTICLIP_CTRL, + 7, 1, 0), + SOC_SINGLE("ALC Anticlip Level", DA9055_ALC_ANTICLIP_LEVEL, + 0, 0x7f, 0), + SOC_SINGLE_TLV("ALC Min Threshold Volume", DA9055_ALC_TARGET_MIN, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", DA9055_ALC_TARGET_MAX, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Noise Threshold Volume", DA9055_ALC_NOISE, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Gain Volume", DA9055_ALC_GAIN_LIMITS, + 4, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation Volume", DA9055_ALC_GAIN_LIMITS, + 0, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Min Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 0, 0x7, 0, alc_analog_gain_tlv), + SOC_SINGLE_TLV("ALC Max Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 4, 0x7, 0, alc_analog_gain_tlv), + SOC_ENUM("ALC Attack Rate", da9055_attack_rate), + SOC_ENUM("ALC Release Rate", da9055_release_rate), + SOC_ENUM("ALC Hold Time", da9055_hold_time), + /* + * Rate at which input signal envelope is tracked as the signal gets + * larger + */ + SOC_ENUM("ALC Integ Attack Rate", da9055_integ_attack_rate), + /* + * Rate at which input signal envelope is tracked as the signal gets + * smaller + */ + SOC_ENUM("ALC Integ Release Rate", da9055_integ_release_rate), +}; + +/* DAPM Controls */ + +/* Mic PGA Left Source */ +static const struct snd_kcontrol_new da9055_mic_l_mux_controls = +SOC_DAPM_ENUM("Route", da9055_mic_l_src); + +/* Mic PGA Right Source */ +static const struct snd_kcontrol_new da9055_mic_r_mux_controls = +SOC_DAPM_ENUM("Route", da9055_mic_r_src); + +/* In Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixinl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXIN_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_L_SELECT, 2, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixinr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXIN_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXIN_R_SELECT, 3, 1, 0), +}; + +/* DAC Left Source */ +static const struct snd_kcontrol_new da9055_dac_l_mux_controls = +SOC_DAPM_ENUM("Route", da9055_dac_l_src); + +/* DAC Right Source */ +static const struct snd_kcontrol_new da9055_dac_r_mux_controls = +SOC_DAPM_ENUM("Route", da9055_dac_r_src); + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixoutl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXOUT_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_L_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", DA9055_MIXOUT_L_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_L_SELECT, + 6, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixoutr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXOUT_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", DA9055_MIXOUT_R_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_R_SELECT, + 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da9055_dapm_widgets[] = { + /* Input Side */ + + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + /* MUXs for Mic PGA source selection */ + SND_SOC_DAPM_MUX("Mic Left Source", SND_SOC_NOPM, 0, 0, + &da9055_mic_l_mux_controls), + SND_SOC_DAPM_MUX("Mic Right Source", SND_SOC_NOPM, 0, 0, + &da9055_mic_r_mux_controls), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA9055_MIC_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA9055_MIC_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Left", DA9055_AUX_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Right", DA9055_AUX_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Left", DA9055_MIXIN_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Right", DA9055_MIXIN_R_CTRL, 7, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", DA9055_MIC_BIAS_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF", DA9055_AIF_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Charge Pump", DA9055_CP_CTRL, 7, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinl_controls[0], + ARRAY_SIZE(da9055_dapm_mixinl_controls)), + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinr_controls[0], + ARRAY_SIZE(da9055_dapm_mixinr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA9055_ADC_L_CTRL, 7, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA9055_ADC_R_CTRL, 7, 0), + + /* Output Side */ + + /* MUXs for DAC source selection */ + SND_SOC_DAPM_MUX("DAC Left Source", SND_SOC_NOPM, 0, 0, + &da9055_dac_l_mux_controls), + SND_SOC_DAPM_MUX("DAC Right Source", SND_SOC_NOPM, 0, 0, + &da9055_dac_r_mux_controls), + + /* AIF input */ + SND_SOC_DAPM_AIF_IN("AIFIN Left", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFIN Right", "Playback", 0, SND_SOC_NOPM, 0, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA9055_DAC_L_CTRL, 7, 0), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA9055_DAC_R_CTRL, 7, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutl_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutl_controls)), + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutr_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutr_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("MIXOUT Left", DA9055_MIXOUT_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXOUT Right", DA9055_MIXOUT_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout", DA9055_LINE_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA9055_HP_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA9055_HP_R_CTRL, 7, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LINE"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da9055_audio_map[] = { + /* Dest Connecting Widget source */ + + /* Input path */ + {"Mic Left Source", "MIC1_P_N", "MIC1"}, + {"Mic Left Source", "MIC1_P", "MIC1"}, + {"Mic Left Source", "MIC1_N", "MIC1"}, + {"Mic Left Source", "MIC2_L", "MIC2"}, + + {"Mic Right Source", "MIC2_R_L", "MIC2"}, + {"Mic Right Source", "MIC2_R", "MIC2"}, + {"Mic Right Source", "MIC2_L", "MIC2"}, + + {"Mic Left", NULL, "Mic Left Source"}, + {"Mic Right", NULL, "Mic Right Source"}, + + {"Aux Left", NULL, "AUXL"}, + {"Aux Right", NULL, "AUXR"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux Left Switch", "Aux Left"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux Right Switch", "Aux Right"}, + {"In Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + + {"MIXIN Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "MIXIN Left"}, + + {"MIXIN Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "MIXIN Right"}, + + {"ADC Left", NULL, "AIF"}, + {"ADC Right", NULL, "AIF"}, + + /* Output path */ + {"AIFIN Left", NULL, "AIF"}, + {"AIFIN Right", NULL, "AIF"}, + + {"DAC Left Source", "ADC output left", "ADC Left"}, + {"DAC Left Source", "ADC output right", "ADC Right"}, + {"DAC Left Source", "AIF input left", "AIFIN Left"}, + {"DAC Left Source", "AIF input right", "AIFIN Right"}, + + {"DAC Right Source", "ADC output left", "ADC Left"}, + {"DAC Right Source", "ADC output right", "ADC Right"}, + {"DAC Right Source", "AIF input left", "AIFIN Left"}, + {"DAC Right Source", "AIF input right", "AIFIN Right"}, + + {"DAC Left", NULL, "DAC Left Source"}, + {"DAC Right", NULL, "DAC Right Source"}, + + {"Out Mixer Left", "Aux Left Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Left", "Aux Left Invert Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux Right Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Right", "Aux Right Invert Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"MIXOUT Left", NULL, "Out Mixer Left"}, + {"Headphone Left", NULL, "MIXOUT Left"}, + {"Headphone Left", NULL, "Charge Pump"}, + {"HPL", NULL, "Headphone Left"}, + + {"MIXOUT Right", NULL, "Out Mixer Right"}, + {"Headphone Right", NULL, "MIXOUT Right"}, + {"Headphone Right", NULL, "Charge Pump"}, + {"HPR", NULL, "Headphone Right"}, + + {"MIXOUT Right", NULL, "Out Mixer Right"}, + {"Lineout", NULL, "MIXOUT Right"}, + {"LINE", NULL, "Lineout"}, +}; + +/* Codec private data */ +struct da9055_priv { + struct regmap *regmap; + unsigned int mclk_rate; + int master; + struct da9055_platform_data *pdata; +}; + +static struct reg_default da9055_reg_defaults[] = { + { 0x21, 0x10 }, + { 0x22, 0x0A }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x0C }, + { 0x28, 0x01 }, + { 0x29, 0x08 }, + { 0x2A, 0x32 }, + { 0x2B, 0x00 }, + { 0x30, 0x35 }, + { 0x31, 0x35 }, + { 0x32, 0x00 }, + { 0x33, 0x00 }, + { 0x34, 0x03 }, + { 0x35, 0x03 }, + { 0x36, 0x6F }, + { 0x37, 0x6F }, + { 0x38, 0x80 }, + { 0x39, 0x01 }, + { 0x3A, 0x01 }, + { 0x40, 0x00 }, + { 0x41, 0x88 }, + { 0x42, 0x88 }, + { 0x43, 0x08 }, + { 0x44, 0x80 }, + { 0x45, 0x6F }, + { 0x46, 0x6F }, + { 0x47, 0x61 }, + { 0x48, 0x35 }, + { 0x49, 0x35 }, + { 0x4A, 0x35 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x60, 0x44 }, + { 0x61, 0x44 }, + { 0x62, 0x00 }, + { 0x63, 0x40 }, + { 0x64, 0x40 }, + { 0x65, 0x40 }, + { 0x66, 0x40 }, + { 0x67, 0x40 }, + { 0x68, 0x40 }, + { 0x69, 0x48 }, + { 0x6A, 0x40 }, + { 0x6B, 0x41 }, + { 0x6C, 0x40 }, + { 0x6D, 0x40 }, + { 0x6E, 0x10 }, + { 0x6F, 0x10 }, + { 0x90, 0x80 }, + { 0x92, 0x02 }, + { 0x93, 0x00 }, + { 0x99, 0x00 }, + { 0x9A, 0x00 }, + { 0x9B, 0x00 }, + { 0x9C, 0x3F }, + { 0x9D, 0x00 }, + { 0x9E, 0x3F }, + { 0x9F, 0xFF }, + { 0xA0, 0x71 }, + { 0xA1, 0x00 }, + { 0xA2, 0x00 }, + { 0xA6, 0x00 }, + { 0xA7, 0x00 }, + { 0xAB, 0x00 }, + { 0xAC, 0x00 }, + { 0xAD, 0x00 }, + { 0xAF, 0x08 }, + { 0xB0, 0x00 }, + { 0xB1, 0x00 }, + { 0xB2, 0x00 }, +}; + +static bool da9055_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case DA9055_STATUS1: + case DA9055_PLL_STATUS: + case DA9055_AUX_L_GAIN_STATUS: + case DA9055_AUX_R_GAIN_STATUS: + case DA9055_MIC_L_GAIN_STATUS: + case DA9055_MIC_R_GAIN_STATUS: + case DA9055_MIXIN_L_GAIN_STATUS: + case DA9055_MIXIN_R_GAIN_STATUS: + case DA9055_ADC_L_GAIN_STATUS: + case DA9055_ADC_R_GAIN_STATUS: + case DA9055_DAC_L_GAIN_STATUS: + case DA9055_DAC_R_GAIN_STATUS: + case DA9055_HP_L_GAIN_STATUS: + case DA9055_HP_R_GAIN_STATUS: + case DA9055_LINE_GAIN_STATUS: + case DA9055_ALC_CIC_OP_LVL_DATA: + return 1; + default: + return 0; + } +} + +/* Set DAI word length */ +static int da9055_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_ctrl, fs; + u32 sysclk; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + aif_ctrl = DA9055_AIF_WORD_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aif_ctrl = DA9055_AIF_WORD_S20_3LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aif_ctrl = DA9055_AIF_WORD_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif_ctrl = DA9055_AIF_WORD_S32_LE; + break; + default: + return -EINVAL; + } + + /* Set AIF format */ + snd_soc_update_bits(codec, DA9055_AIF_CTRL, DA9055_AIF_WORD_LENGTH_MASK, + aif_ctrl); + + switch (params_rate(params)) { + case 8000: + fs = DA9055_SR_8000; + sysclk = 3072000; + break; + case 11025: + fs = DA9055_SR_11025; + sysclk = 2822400; + break; + case 12000: + fs = DA9055_SR_12000; + sysclk = 3072000; + break; + case 16000: + fs = DA9055_SR_16000; + sysclk = 3072000; + break; + case 22050: + fs = DA9055_SR_22050; + sysclk = 2822400; + break; + case 32000: + fs = DA9055_SR_32000; + sysclk = 3072000; + break; + case 44100: + fs = DA9055_SR_44100; + sysclk = 2822400; + break; + case 48000: + fs = DA9055_SR_48000; + sysclk = 3072000; + break; + case 88200: + fs = DA9055_SR_88200; + sysclk = 2822400; + break; + case 96000: + fs = DA9055_SR_96000; + sysclk = 3072000; + break; + default: + return -EINVAL; + } + + if (da9055->mclk_rate) { + /* PLL Mode, Write actual FS */ + snd_soc_write(codec, DA9055_SR, fs); + } else { + /* + * Non-PLL Mode + * When PLL is bypassed, chip assumes constant MCLK of + * 12.288MHz and uses sample rate value to divide this MCLK + * to derive its sys clk. As sys clk has to be 256 * Fs, we + * need to write constant sample rate i.e. 48KHz. + */ + snd_soc_write(codec, DA9055_SR, DA9055_SR_48000); + } + + if (da9055->mclk_rate && (da9055->mclk_rate != sysclk)) { + /* PLL Mode */ + if (!da9055->master) { + /* PLL slave mode, enable PLL and also SRM */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN | DA9055_PLL_SRM_EN, + DA9055_PLL_EN | DA9055_PLL_SRM_EN); + } else { + /* PLL master mode, only enable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN, DA9055_PLL_EN); + } + } else { + /* Non PLL Mode, disable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + } + + return 0; +} + +/* Set DAI mode and Format */ +static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_clk_mode, aif_ctrl, mode; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* DA9055 in I2S Master Mode */ + mode = 1; + aif_clk_mode = DA9055_AIF_CLK_EN_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + /* DA9055 in I2S Slave Mode */ + mode = 0; + aif_clk_mode = DA9055_AIF_CLK_EN_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* Don't allow change of mode if PLL is enabled */ + if ((snd_soc_read(codec, DA9055_PLL_CTRL) & DA9055_PLL_EN) && + (da9055->master != mode)) + return -EINVAL; + + da9055->master = mode; + + /* Only I2S is supported */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif_ctrl = DA9055_AIF_FORMAT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif_ctrl = DA9055_AIF_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + aif_ctrl = DA9055_AIF_FORMAT_RIGHT_J; + break; + default: + return -EINVAL; + } + + /* By default only 32 BCLK per WCLK is supported */ + aif_clk_mode |= DA9055_AIF_BCLKS_PER_WCLK_32; + + snd_soc_update_bits(codec, DA9055_AIF_CLK_MODE, + (DA9055_AIF_CLK_MODE_MASK | DA9055_AIF_BCLK_MASK), + aif_clk_mode); + snd_soc_update_bits(codec, DA9055_AIF_CTRL, DA9055_AIF_FORMAT_MASK, + aif_ctrl); + return 0; +} + +static int da9055_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, DA9055_DAC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, DA9055_DAC_R_MUTE_EN); + } else { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, 0); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, 0); + } + + return 0; +} + +#define DA9055_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int da9055_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA9055_CLKSRC_MCLK: + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da9055->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/* + * da9055_set_dai_pll : Configure the codec PLL + * @param codec_dai : Pointer to codec DAI + * @param pll_id : da9055 has only one pll, so pll_id is always zero + * @param fref : Input MCLK frequency + * @param fout : FsDM value + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 11.2896MHz, 12MHz, 12.288MHz, + * 13MHz, 13.5MHz, 14.4MHz, 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da9055_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + u8 pll_frac_top, pll_frac_bot, pll_integer, cnt; + + /* Disable PLL before setting the divisors */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + + /* In slave mode, there is only one set of divisors */ + if (!da9055->master && (fout != 2822400)) + goto pll_err; + + /* Search pll div array for correct divisors */ + for (cnt = 0; cnt < ARRAY_SIZE(da9055_pll_div); cnt++) { + /* Check fref, mode and fout */ + if ((fref == da9055_pll_div[cnt].fref) && + (da9055->master == da9055_pll_div[cnt].mode) && + (fout == da9055_pll_div[cnt].fout)) { + /* All match, pick up divisors */ + pll_frac_top = da9055_pll_div[cnt].frac_top; + pll_frac_bot = da9055_pll_div[cnt].frac_bot; + pll_integer = da9055_pll_div[cnt].integer; + break; + } + } + if (cnt >= ARRAY_SIZE(da9055_pll_div)) + goto pll_err; + + /* Write PLL dividers */ + snd_soc_write(codec, DA9055_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA9055_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA9055_PLL_INTEGER, pll_integer); + + return 0; +pll_err: + dev_err(codec_dai->dev, "Error in setting up PLL\n"); + return -EINVAL; +} + +/* DAI operations */ +static const struct snd_soc_dai_ops da9055_dai_ops = { + .hw_params = da9055_hw_params, + .set_fmt = da9055_set_dai_fmt, + .set_sysclk = da9055_set_dai_sysclk, + .set_pll = da9055_set_dai_pll, + .digital_mute = da9055_mute, +}; + +static struct snd_soc_dai_driver da9055_dai = { + .name = "da9055-hifi", + /* Playback Capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + /* Capture Capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + .ops = &da9055_dai_ops, + .symmetric_rates = 1, +}; + +static int da9055_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* Enable VMID reference & master bias */ + snd_soc_update_bits(codec, DA9055_REFERENCES, + DA9055_VMID_EN | DA9055_BIAS_EN, + DA9055_VMID_EN | DA9055_BIAS_EN); + } + break; + case SND_SOC_BIAS_OFF: + /* Disable VMID reference & master bias */ + snd_soc_update_bits(codec, DA9055_REFERENCES, + DA9055_VMID_EN | DA9055_BIAS_EN, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int da9055_probe(struct snd_soc_codec *codec) +{ + int ret; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + codec->control_data = da9055->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Enable all Gain Ramps */ + snd_soc_update_bits(codec, DA9055_AUX_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_AUX_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + + /* + * There are two separate control bits for input and output mixers as + * well as headphone and line outs. + * One to enable corresponding amplifier and other to enable its + * output. As amplifier bits are related to power control, they are + * being managed by DAPM while other (non power related) bits are + * enabled here + */ + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_L_MIX_EN, DA9055_MIXIN_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_MIXIN_R_MIX_EN, DA9055_MIXIN_R_MIX_EN); + + snd_soc_update_bits(codec, DA9055_MIXOUT_L_CTRL, + DA9055_MIXOUT_L_MIX_EN, DA9055_MIXOUT_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXOUT_R_CTRL, + DA9055_MIXOUT_R_MIX_EN, DA9055_MIXOUT_R_MIX_EN); + + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_HP_L_AMP_OE, DA9055_HP_L_AMP_OE); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_HP_R_AMP_OE, DA9055_HP_R_AMP_OE); + + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_LINE_AMP_OE, DA9055_LINE_AMP_OE); + + /* Set this as per your system configuration */ + snd_soc_write(codec, DA9055_PLL_CTRL, DA9055_PLL_INDIV_10_20_MHZ); + + /* Set platform data values */ + if (da9055->pdata) { + /* set mic bias source */ + if (da9055->pdata->micbias_source) { + snd_soc_update_bits(codec, DA9055_MIXIN_R_SELECT, + DA9055_MICBIAS2_EN, + DA9055_MICBIAS2_EN); + } else { + snd_soc_update_bits(codec, DA9055_MIXIN_R_SELECT, + DA9055_MICBIAS2_EN, 0); + } + /* set mic bias voltage */ + switch (da9055->pdata->micbias) { + case DA9055_MICBIAS_2_2V: + case DA9055_MICBIAS_2_1V: + case DA9055_MICBIAS_1_8V: + case DA9055_MICBIAS_1_6V: + snd_soc_update_bits(codec, DA9055_MIC_CONFIG, + DA9055_MICBIAS_LEVEL_MASK, + (da9055->pdata->micbias) << 4); + break; + } + } + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_da9055 = { + .probe = da9055_probe, + .set_bias_level = da9055_set_bias_level, + + .controls = da9055_snd_controls, + .num_controls = ARRAY_SIZE(da9055_snd_controls), + + .dapm_widgets = da9055_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da9055_dapm_widgets), + .dapm_routes = da9055_audio_map, + .num_dapm_routes = ARRAY_SIZE(da9055_audio_map), +}; + +static const struct regmap_config da9055_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = da9055_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da9055_reg_defaults), + .volatile_reg = da9055_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit da9055_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da9055_priv *da9055; + struct da9055_platform_data *pdata = dev_get_platdata(&i2c->dev); + int ret; + + da9055 = devm_kzalloc(&i2c->dev, sizeof(struct da9055_priv), + GFP_KERNEL); + if (!da9055) + return -ENOMEM; + + if (pdata) + da9055->pdata = pdata; + + i2c_set_clientdata(i2c, da9055); + + da9055->regmap = devm_regmap_init_i2c(i2c, &da9055_regmap_config); + if (IS_ERR(da9055->regmap)) { + ret = PTR_ERR(da9055->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_da9055, &da9055_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da9055 codec: %d\n", + ret); + } + return ret; +} + +static int __devexit da9055_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id da9055_i2c_id[] = { + { "da9055", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da9055_i2c_driver = { + .driver = { + .name = "da9055", + .owner = THIS_MODULE, + }, + .probe = da9055_i2c_probe, + .remove = __devexit_p(da9055_remove), + .id_table = da9055_i2c_id, +}; + +module_i2c_driver(da9055_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA9055 Codec driver"); +MODULE_AUTHOR("David Chen, Ashish Chavan"); +MODULE_LICENSE("GPL"); -- 1.7.1

2 1

[alsa-devel] [PATCH v2] ASoC: eukrea-tlv320: Convert it to platform driver
by Fabio Estevam 28 Sep '12

28 Sep '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Convert eukrea-tlv320 to platform driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - No changes, only CC'ed ARM kernel list as requested by Mark Brown arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c | 1 + sound/soc/fsl/eukrea-tlv320.c | 37 ++++++++++++------------ 5 files changed, 23 insertions(+), 18 deletions(-) diff --git a/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c b/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c index fd3177f..98aef57 100644 --- a/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c @@ -348,4 +348,5 @@ void __init eukrea_mbimx27_baseboard_init(void) imx27_add_imx_keypad(&eukrea_mbimx27_keymap_data); gpio_led_register_device(-1, &eukrea_mbimx27_gpio_led_info); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c b/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c index dfd2da8..0b84666 100644 --- a/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c @@ -306,4 +306,5 @@ void __init eukrea_mbimxsd25_baseboard_init(void) platform_add_devices(platform_devices, ARRAY_SIZE(platform_devices)); gpio_led_register_device(-1, &eukrea_mbimxsd_led_info); imx_add_gpio_keys(&eukrea_mbimxsd_button_data); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c b/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c index 6e9dd12..c6532a0 100644 --- a/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c @@ -315,4 +315,5 @@ void __init eukrea_mbimxsd35_baseboard_init(void) platform_add_devices(platform_devices, ARRAY_SIZE(platform_devices)); gpio_led_register_device(-1, &eukrea_mbimxsd_led_info); imx_add_gpio_keys(&eukrea_mbimxsd_button_data); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c b/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c index 96a24b7..8b0de30 100644 --- a/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c @@ -228,4 +228,5 @@ void __init eukrea_mbimxsd51_baseboard_init(void) gpio_led_register_device(-1, &eukrea_mbimxsd51_led_info); imx_add_gpio_keys(&eukrea_mbimxsd51_button_data); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index efb9ede..267d5b4 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -93,9 +93,7 @@ static struct snd_soc_card eukrea_tlv320 = { .num_links = 1, }; -static struct platform_device *eukrea_tlv320_snd_device; - -static int __init eukrea_tlv320_init(void) +static int __devinit eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; @@ -136,29 +134,32 @@ static int __init eukrea_tlv320_init(void) return 0; } - eukrea_tlv320_snd_device = platform_device_alloc("soc-audio", -1); - if (!eukrea_tlv320_snd_device) - return -ENOMEM; - - platform_set_drvdata(eukrea_tlv320_snd_device, &eukrea_tlv320); - ret = platform_device_add(eukrea_tlv320_snd_device); - - if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(eukrea_tlv320_snd_device); - } + eukrea_tlv320.dev = &pdev->dev; + ret = snd_soc_register_card(&eukrea_tlv320); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); return ret; } -static void __exit eukrea_tlv320_exit(void) +static int __devexit eukrea_tlv320_remove(struct platform_device *pdev) { - platform_device_unregister(eukrea_tlv320_snd_device); + snd_soc_unregister_card(&eukrea_tlv320); + + return 0; } -module_init(eukrea_tlv320_init); -module_exit(eukrea_tlv320_exit); +static struct platform_driver eukrea_tlv320_driver = { + .driver = { + .name = "eukrea_tlv320", + .owner = THIS_MODULE, + }, + .probe = eukrea_tlv320_probe, + .remove = __devexit_p(eukrea_tlv320_remove),}; + +module_platform_driver(eukrea_tlv320_driver); MODULE_AUTHOR("Eric Bénard <eric(a)eukrea.com>"); MODULE_DESCRIPTION("CPUIMX ALSA SoC driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:eukrea_tlv320"); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: ASoC: add DT bindings for CS4271
by Daniel Mack 28 Sep '12

28 Sep '12

Apart from pure matching, the bindings also support setting the the reset gpio line. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Alexander Sverdlin <subaparts(a)yandex.ru> --- Documentation/devicetree/bindings/sound/cs4271.txt | 36 ++++++++++++++++++++++ sound/soc/codecs/cs4271.c | 24 +++++++++++++-- 2 files changed, 57 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs4271.txt diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt new file mode 100644 index 0000000..c81b5fd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -0,0 +1,36 @@ +Cirrus Logic CS4271 DT bindings + +This driver supports both the I2C and the SPI bus. + +Required properties: + + - compatible: "cirrus,cs4271" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Required properties on I2C: + + - reg: the i2c address + + +Optional properties: + + - reset-gpio: a GPIO spec to define which pin is connected to the chip's + !RESET pin + +Examples: + + codec_i2c: cs4271@10 { + compatible = "cirrus,cs4271"; + reg = <0x10>; + reset-gpio = <&gpio 23 0>; + }; + + codec_spi: cs4271@0 { + compatible = "cirrus,cs4271"; + reg = <0x0>; + reset-gpio = <&gpio 23 0>; + spi-max-frequency = <6000000>; + }; + diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 9eb01d7..f994af3 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -22,12 +22,14 @@ #include <linux/module.h> #include <linux/slab.h> #include <linux/delay.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/tlv.h> #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/tlv.h> #include <sound/cs4271.h> #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -458,6 +460,14 @@ static int cs4271_soc_resume(struct snd_soc_codec *codec) #define cs4271_soc_resume NULL #endif /* CONFIG_PM */ +#ifdef CONFIG_OF +static const struct of_device_id cs4271_dt_ids[] = { + { .compatible = "cirrus,cs4271", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4271_dt_ids); +#endif + static int cs4271_probe(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); @@ -465,6 +475,12 @@ static int cs4271_probe(struct snd_soc_codec *codec) int ret; int gpio_nreset = -EINVAL; +#ifdef CONFIG_OF + if (of_match_device(cs4271_dt_ids, codec->dev)) + gpio_nreset = of_get_named_gpio(codec->dev->of_node, + "reset-gpio", 0); +#endif + if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) gpio_nreset = cs4271plat->gpio_nreset; @@ -569,6 +585,7 @@ static struct spi_driver cs4271_spi_driver = { .driver = { .name = "cs4271", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(cs4271_dt_ids), }, .probe = cs4271_spi_probe, .remove = __devexit_p(cs4271_spi_remove), @@ -608,6 +625,7 @@ static struct i2c_driver cs4271_i2c_driver = { .driver = { .name = "cs4271", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(cs4271_dt_ids), }, .id_table = cs4271_i2c_id, .probe = cs4271_i2c_probe, -- 1.7.11.4

2 1

Re: [alsa-devel] ASoC: dapm: Allow regulators to bypass as well as disable when idle
by Dan Carpenter 28 Sep '12

28 Sep '12

Hi Mark, I had a question about patch c05b84d14b23: "ASoC: dapm: Allow regulators to bypass as well as disable when idle" from Sep 7, 2012. 1022 if (SND_SOC_DAPM_EVENT_ON(event)) { 1023 if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 1024 ret = regulator_allow_bypass(w->regulator, true); 1025 if (ret != 0) 1026 dev_warn(w->dapm->dev, 1027 "Failed to bypass %s: %d\n", 1028 w->name, ret); 1029 } 1030 1031 return regulator_enable(w->regulator); 1032 } else { 1033 if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Right now ->invert is a single bit bitfield. We mostly test "if (w->invert) {...". Are we planning to add some more bits in there? 1034 ret = regulator_allow_bypass(w->regulator, false); 1035 if (ret != 0) 1036 dev_warn(w->dapm->dev, 1037 "Failed to unbypass %s: %d\n", 1038 w->name, ret); 1039 } regards, dan carpenter

2 1

Re: [alsa-devel] [PATCH V2] ASoC: SAMSUNG: Add DT support for i2s
by Stephen Warren 28 Sep '12

28 Sep '12

On 09/27/2012 04:25 AM, Padmavathi Venna wrote: > Add support for device based discovery. > +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > +i2s@03830000 { > + compatible = "samsung,samsung-i2s"; > + reg = <0x03830000 0x100>; > + tx-dma-channel-secondary = <&pdma0 8>; > + tx-dma-channel = <&pdma0 10>; > + rx-dma-channel = <&pdma0 9>; > + supports-6ch; > + supports-rstclr; > + supports-secdai; Those 6 properties are custom to this binding so should have a vendor prefix, e.g. "samsung,supports-6ch". Also, I wonder if this shouldn't wait for the forthcoming standardized DT DMA bindings (or perhaps document custom bindings that are identical to the new standardized bindings)? Otherwise, the DMA stuff will all have to be redefined again. Apparently it's hoped the bindings will be in 3.7, or if not there, in 3.8.

2 1

Re: [alsa-devel] [PATCH V2] ASoC: SAMSUNG: Add DT support for i2s
by Sangbeom Kim 28 Sep '12

28 Sep '12

Hi! On Thu, Sep 27, 2012 at 07:26 PM, Padmavathi Venna wrote: > Subject: [PATCH V2] ASoC: SAMSUNG: Add DT support for i2s > > Add support for device based discovery. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> Please describe the difference between v1 and v2. Thanks, Sangbeom.

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[alsa-devel] [PATCH] pcm: support for audio timestamps
by Pierre-Louis Bossart 28 Sep '12

28 Sep '12

add new snd_pcm_status_get_audio_htstamp() routine to query the audio timestamps provided by the kernel. This change provides applications with better ways to track elapsed time. Before this patch, applications would subtract queued samples (delay) from written samples, resulting in a 1-2 sample error. Also add snd_pcm_hw_params_supports_audio_wallclock_ts() to query what the hardware supports. TODO: check protocol compatibility? Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/pcm.h | 2 ++ include/sound/asound.h | 7 +++++-- src/pcm/pcm.c | 31 +++++++++++++++++++++++++++++++ test/audio_time.c | 27 +++++++++++++++++---------- 4 files changed, 55 insertions(+), 12 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 4997557..8d08471 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -550,6 +550,7 @@ int snd_pcm_hw_params_is_half_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_is_joint_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_sync_start(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *params); +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_get_rate_numden(const snd_pcm_hw_params_t *params, unsigned int *rate_num, unsigned int *rate_den); @@ -858,6 +859,7 @@ void snd_pcm_status_get_trigger_tstamp(const snd_pcm_status_t *obj, snd_timestam void snd_pcm_status_get_trigger_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); void snd_pcm_status_get_tstamp(const snd_pcm_status_t *obj, snd_timestamp_t *ptr); void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); +void snd_pcm_status_get_audio_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); diff --git a/include/sound/asound.h b/include/sound/asound.h index 07c03fa..109302e 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -160,7 +160,7 @@ enum { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) typedef unsigned long sndrv_pcm_uframes_t; typedef long sndrv_pcm_sframes_t; @@ -285,6 +285,7 @@ enum sndrv_pcm_subformat { #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ enum sndrv_pcm_state { SNDRV_PCM_STATE_OPEN = 0, /* stream is open */ @@ -426,7 +427,8 @@ struct sndrv_pcm_status { sndrv_pcm_uframes_t avail_max; /* max frames available on hw since last status */ sndrv_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ int suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ + unsigned char reserved[60-sizeof(struct timespec)]; /* must be filled with zero */ }; struct sndrv_pcm_mmap_status { @@ -435,6 +437,7 @@ struct sndrv_pcm_mmap_status { sndrv_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ int suspended_state; /* RO: suspended stream state */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ }; struct sndrv_pcm_mmap_control { diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 18b43b3..748e8a6 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -3123,6 +3123,26 @@ int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *param } /** + * \brief Check if hardware supports audio wallclock timestamps + * \param params Configuration space + * \retval 0 Hardware doesn't support audio wallclock timestamps + * \retval 1 Hardware supports audio wallclock timestamps + * + * This function should only be called when the configuration space + * contains a single configuration. Call #snd_pcm_hw_params to choose + * a single configuration from the configuration space. + */ +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params) +{ + assert(params); + if (CHECK_SANITY(params->info == ~0U)) { + SNDMSG("invalid PCM info field"); + return 0; /* FIXME: should be a negative error? */ + } + return !!(params->info & SNDRV_PCM_INFO_HAS_WALL_CLOCK); +} + +/** * \brief Get rate exact info from a configuration space * \param params Configuration space * \param rate_num Pointer to returned rate numerator @@ -6213,6 +6233,17 @@ void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *p use_default_symbol_version(__snd_pcm_status_get_htstamp, snd_pcm_status_get_htstamp, ALSA_0.9.0rc8); /** + * \brief Get "now" hi-res audio timestamp from a PCM status container + * \param obj pointer to #snd_pcm_status_t + * \param ptr Pointer to returned timestamp + */ +void snd_pcm_status_get_audio_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +{ + assert(obj && ptr); + *ptr = obj->audio_tstamp; +} + +/** * \brief Get delay from a PCM status container (see #snd_pcm_delay) * \return Delay in frames * diff --git a/test/audio_time.c b/test/audio_time.c index a910783..de8df17 100644 --- a/test/audio_time.c +++ b/test/audio_time.c @@ -33,6 +33,7 @@ long long timediff(snd_htimestamp_t t1, snd_htimestamp_t t2) void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, snd_htimestamp_t *trigger_timestamp, + snd_htimestamp_t *audio_timestamp, snd_pcm_uframes_t *avail, snd_pcm_sframes_t *delay) { int err; @@ -45,6 +46,7 @@ void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, } snd_pcm_status_get_trigger_htstamp(status, trigger_timestamp); snd_pcm_status_get_htstamp(status, timestamp); + snd_pcm_status_get_audio_htstamp(status, audio_timestamp); *avail = snd_pcm_status_get_avail(status); *delay = snd_pcm_status_get_delay(status); } @@ -65,6 +67,7 @@ int main(void) snd_pcm_sframes_t frames; snd_htimestamp_t tstamp_c, tstamp_p; snd_htimestamp_t trigger_tstamp_c, trigger_tstamp_p; + snd_htimestamp_t audio_tstamp_c, audio_tstamp_p; unsigned char buffer_p[PERIOD*4*4]; unsigned char buffer_c[PERIOD*4*4]; @@ -202,26 +205,30 @@ int main(void) frame_count_p += frames; #if defined(TRACK_PLAYBACK) - gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &avail_p, &delay_p); + gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &audio_tstamp_p, &avail_p, &delay_p); curr_count_p = frame_count_p - delay_p; /* written minus queued */ - printf("playback: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", - timediff(tstamp_p,trigger_tstamp_p), - (long long)round(((float)curr_count_p * 1000000000.0 / 48000.0)), - timediff(tstamp_p, trigger_tstamp_p) - (long long)round((double)curr_count_p * 1000000000.0 / 48000.0) + /* uncomment next line to see difference between old sample count and audio timestamps. + printf("curr_count %lli driver count %lli\n", (long long)curr_count_p * 1000000000LL / 48000, timestamp2ns(audio_tstamp_p)); + */ + + printf("playback: systime: %lli nsec, audio time %lli nsec, \tsystime delta %lli\n", + timediff(tstamp_p, trigger_tstamp_p), + timestamp2ns(audio_tstamp_p), + timediff(tstamp_p, trigger_tstamp_p) - timestamp2ns(audio_tstamp_p) ); #endif #if defined(TRACK_CAPTURE) - gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &avail_c, &delay_c); + gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &audio_tstamp_c, &avail_c, &delay_c); curr_count_c = frame_count_c + delay_c; /* read plus queued */ - printf("\t capture: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", - timediff(tstamp_c,trigger_tstamp_c), - (long long)round(((float)curr_count_c * 1000000000.0 / 48000.0)), - timediff(tstamp_c, trigger_tstamp_c) - (long long)round((double)curr_count_c * 1000000000.0 / 48000.0) + printf("\t capture: systime: %lli nsec, audio time %lli nsec, \tsystime delta %lli\n", + timediff(tstamp_c, trigger_tstamp_c), + timestamp2ns(audio_tstamp_c), + timediff(tstamp_c, trigger_tstamp_c) - timestamp2ns(audio_tstamp_c) ); #endif -- 1.7.9.5

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[alsa-devel] setup_dig_out_stream() resets the SPDIF steam causing clicks
by Laurence Darby 28 Sep '12

28 Sep '12

Hi, I'm playing audio via SPDIF, with a Realtek codec, and there's a problem that whenever audio is started, or seeking in every music player I tried, the DAC lost sync briefly, which made it 'click' when it muted and unmuted itself. I recently upgraded the DAC, and this problem only happened with the new DAC, but not the old one, I don't know (or really care) why - the old DAC had much worse problems. Here's what I did to fix this: diff -up linux-3.5.4/sound/pci/hda/hda_codec.c.orig linux-3.5.4/sound/pci/hda/hda_codec.c --- linux-3.5.4/sound/pci/hda/hda_codec.c.orig 2012-09-28 00:13:25.000000000 +0100 +++ linux-3.5.4/sound/pci/hda/hda_codec.c 2012-09-28 00:39:31.000000000 +0100 @@ -4693,9 +4693,11 @@ static void setup_dig_out_stream(struct format); } /* turn on again (if needed) */ - if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE)) + if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE)) { set_dig_out_convert(codec, nid, spdif->ctls & 0xff, -1); + codec->spdif_status_reset=0; + } } static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) That makes SPDIF turn off and on just once when sound is first played (i.e. it clicks once then, then no more). In patch_realtek.c it's got this: /* FIXME: do we need this for all Realtek codec models? */ codec->spdif_status_reset = 1; If I set that to 0, then the left channel doesn't initialize and is silent, so the reset really does need to happen once (for my realtek model anyway, I don't know how to find out what that would be?) So, is the above an acceptable fix? Regards, Laurence

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[alsa-devel] [PATCH] ASoC: wm5110: Adding missing volume update bits.
by Mark Brown 27 Sep '12

27 Sep '12

From: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> The volume update bits were being set on all but one input and one output. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index b99451e..9211e41 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -916,6 +916,8 @@ static unsigned int wm5110_digital_vu[] = { ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_ADC_DIGITAL_VOLUME_3L, ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, @@ -927,6 +929,8 @@ static unsigned int wm5110_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_DAC_DIGITAL_VOLUME_5L, ARIZONA_DAC_DIGITAL_VOLUME_5R, + ARIZONA_DAC_DIGITAL_VOLUME_6L, + ARIZONA_DAC_DIGITAL_VOLUME_6R, }; static struct snd_soc_codec_driver soc_codec_dev_wm5110 = { -- 1.7.10.4

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[alsa-devel] ASoC updates for 3.6
by Mark Brown 27 Sep '12

27 Sep '12

The following changes since commit 5698bd757d55b1bb87edd1a9744ab09c142abfc2: Linux 3.6-rc6 (2012-09-16 14:58:51 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to d0e12f3ff3472cbd8f52d3c0e6ee07a841787c40: ASoC: wm2000: Correct register size (2012-09-26 12:06:20 +0100) Takashi is on vacation now so he asked me to send directly, it's a pretty bad bug with low regression risk. ---------------------------------------------------------------- ASoC: Updates for 3.6 One small and obvious driver-specific fix. ---------------------------------------------------------------- Mark Brown (1): ASoC: wm2000: Correct register size sound/soc/codecs/wm2000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

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[alsa-devel] ASoC on today's linux-next: dma_alloc_writecombine() failing at least on OMAP platforms
by Peter Ujfalusi 27 Sep '12

27 Sep '12

Hello, I tried today's linux-next on OMAP platform and audio is not probing because dma_alloc_writecombine() fails to get the area. My linux-next from two days ago works OK and I don't see anything directly connected to OMAP which can cause this failure. Can someone try linux-next on non OMAP platform to see if dma_alloc_writecombine() works there? Meanwhile I'll try to find the reason why it fails for me. Thanks, Péter

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[alsa-devel] OMAP4 McBSP AC97 link
by Sebastian 27 Sep '12

27 Sep '12

Hi All, i have to support a 2nd Audio Codec with AC97 Interface on McBSP Port2. Processor: OMAP4460 If any one have any idea on how to go about this. Please let me know. BR, Sebastian

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[alsa-devel] sound/pci/hda/patch_ca0132.c: potential null dereference 'dma_engine'
by Fengguang Wu 27 Sep '12

27 Sep '12

Hi Ian, FYI, there are new smatch warnings show up in commit 0b37d4c7982442730061a737c16939e8cc363861 Author: Ian Minett <ian_minett(a)creativelabs.com> Commit: Takashi Iwai <tiwai(a)suse.de> CommitDate: Fri Sep 21 09:54:29 2012 +0200 + sound/pci/hda/patch_ca0132.c:2056 dspxfr_image() error: potential null dereference 'dma_engine'. (kzalloc returns null) + sound/pci/hda/patch_ca0132.c:2056 dspxfr_image() error: we previously assumed 'dma_engine' could be null (see line 1963) --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [PATCH 1/2] ASoC: arizona: Add more clock rates
by Mark Brown 26 Sep '12

26 Sep '12

Some devices support additional clock rates. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5764960..c03b65a 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -270,6 +270,9 @@ static unsigned int arizona_sysclk_48k_rates[] = { 12288000, 22579200, 49152000, + 73728000, + 98304000, + 147456000, }; static unsigned int arizona_sysclk_44k1_rates[] = { @@ -277,6 +280,9 @@ static unsigned int arizona_sysclk_44k1_rates[] = { 11289600, 24576000, 45158400, + 67737600, + 90316800, + 135475200, }; static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: pcm: Report error code when we fail to init platform
by Mark Brown 26 Sep '12

26 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a67b239..40c4edb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2097,7 +2097,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { - dev_err(platform->dev, "pcm constructor failed\n"); + dev_err(platform->dev, "pcm constructor failed: %d\n", + ret); return ret; } } -- 1.7.10.4

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