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[alsa-devel] snd-fm801: autodetect SF64-PCR (tuner-only) card
by Ondrej Zary 03 Dec '09

03 Dec '09

When primary AC97 is not found, don't fail with tons of AC97 errors. Assume that the card is SF64-PCR (tuner-only). This makes the SF64-PCR radio card work "out of the box". Also fixes a bug that can cause an oops here: if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) { when tea575x_tuner == 16, it passes this check and causes problems a couple lines below: chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1]; Tested with SF64-PCR, but I don't have any of those sound or sound+radio cards to test if I didn't break anything. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-source-2.6.31-orig/sound/pci/fm801.c 2009-09-10 00:13:59.000000000 +0200 +++ linux-source-2.6.31/sound/pci/fm801.c 2009-11-26 21:19:59.000000000 +0100 @@ -55,7 +55,6 @@ static int enable[SNDRV_CARDS] = SNDRV_D * 1 = MediaForte 256-PCS * 2 = MediaForte 256-PCPR * 3 = MediaForte 64-PCR - * 16 = setup tuner only (this is additional bit), i.e. SF-64-PCR FM card * High 16-bits are video (radio) device number + 1 */ static int tea575x_tuner[SNDRV_CARDS]; @@ -67,7 +66,7 @@ MODULE_PARM_DESC(id, "ID string for the module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); -MODULE_PARM_DESC(tea575x_tuner, "Enable TEA575x tuner."); +MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner type (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR)."); /* * Direct registers @@ -160,7 +159,8 @@ struct fm801 { unsigned int multichannel: 1, /* multichannel support */ secondary: 1; /* secondary codec */ unsigned char secondary_addr; /* address of the secondary codec */ - unsigned int tea575x_tuner; /* tuner flags */ + unsigned int tea575x_tuner; /* tuner type */ + bool tuner_only; unsigned short ply_ctrl; /* playback control */ unsigned short cap_ctrl; /* capture control */ @@ -1287,20 +1287,20 @@ static int snd_fm801_chip_init(struct fm { unsigned short cmdw; - if (chip->tea575x_tuner & 0x0010) - goto __ac97_ok; - /* codec cold reset + AC'97 warm reset */ outw((1<<5) | (1<<6), FM801_REG(chip, CODEC_CTRL)); inw(FM801_REG(chip, CODEC_CTRL)); /* flush posting data */ udelay(100); outw(0, FM801_REG(chip, CODEC_CTRL)); - if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) { - snd_printk(KERN_ERR "Primary AC'97 codec not found\n"); - if (! resume) - return -EIO; - } + if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) + if (!resume) { + snd_printk(KERN_ERR "Primary AC'97 codec not found, " + "assume SF64-PCR (tuner-only)\n"); + chip->tuner_only = 1; + chip->tea575x_tuner = 3; + goto __ac97_ok; + } if (chip->multichannel) { if (chip->secondary_addr) { @@ -1414,21 +1414,27 @@ static int __devinit snd_fm801_create(st return err; } chip->port = pci_resource_start(pci, 0); - if ((tea575x_tuner & 0x0010) == 0) { - if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED, - "FM801", chip)) { - snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq); - snd_fm801_free(chip); - return -EBUSY; - } - chip->irq = pci->irq; - pci_set_master(pci); + if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED, + "FM801", chip)) { + snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq); + snd_fm801_free(chip); + return -EBUSY; } + chip->irq = pci->irq; + pci_set_master(pci); if (pci->revision >= 0xb1) /* FM801-AU */ chip->multichannel = 1; snd_fm801_chip_init(chip, 0); + /* init might set tuner type */ + tea575x_tuner = chip->tea575x_tuner; + + if (chip->tuner_only) { + pci_clear_master(pci); + free_irq(chip->irq, chip); + chip->irq = -1; + } if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_fm801_free(chip); @@ -1438,7 +1444,7 @@ static int __devinit snd_fm801_create(st snd_card_set_dev(card, &pci->dev); #ifdef TEA575X_RADIO - if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) { + if ((tea575x_tuner & 0x000f) > 0 && (tea575x_tuner & 0x000f) < 4) { chip->tea.dev_nr = tea575x_tuner >> 16; chip->tea.card = card; chip->tea.freq_fixup = 10700; @@ -1483,7 +1489,7 @@ static int __devinit snd_card_fm801_prob sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->port, chip->irq); - if (tea575x_tuner[dev] & 0x0010) + if (chip->tuner_only) goto __fm801_tuner_only; if ((err = snd_fm801_pcm(chip, 0, NULL)) < 0) { -- Ondrej Zary

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[alsa-devel] [resend] tea575x-tuner: fix mute
by Ondrej Zary 03 Dec '09

03 Dec '09

Fix mute state reporting in tea575x-tuner. This fixes mute function in kradio on SF64-PCR radio card. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-source-2.6.31-orig/sound/i2c/other/tea575x-tuner.c 2009-09-10 00:13:59.000000000 +0200 +++ linux-source-2.6.31/sound/i2c/other/tea575x-tuner.c 2009-11-26 21:30:28.000000000 +0100 @@ -225,7 +225,7 @@ static int vidioc_s_ctrl(struct file *fi case V4L2_CID_AUDIO_MUTE: if (tea->ops->mute) { tea->ops->mute(tea, ctrl->value); - tea->mute = 1; + tea->mute = ctrl->value; return 0; } } -- Ondrej Zary

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[alsa-devel] No CD sound from Conexant CX20561 (Hermosa)
by Shawn Starr 03 Dec '09

03 Dec '09

Hello, I'm having additional issues with this sound card. There is no CD audio listed in alsa at all, when playing a CD, there's no sound. I thought there was a separate channel for CD audio? Thanks, Shawn.

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[alsa-devel] io-plugin does not call start callback function
by Stefan Schoenleitner 03 Dec '09

03 Dec '09

Hi, I'm writing an io plugin that should transfer all PCM samples to a UNIX socket. For this reason in my plugin I open the socket and then fill the plugin struct like this: socketdump->io.mmap_rw = 0; socketdump->io.poll_fd = socketdump->sock_fd; socketdump->io.poll_events = POLLOUT; Thus for my understanding (the alsa doc is really bad), every time the socket file descriptor is available for writing the alsa plugin would generally speaking issue a data transfer. However, when playing audio over that plugin for some reason the start callback function is never called. Instead I get output like this: socketdump_hw_constraint:417 26944 _snd_pcm_socketdump_open:585 26944 SND_PCM_PLUGIN_DEFINE_FUNC(): done, err=0 Playing WAVE 'test_8khz_16LE_mono.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono snd_pcm_socketdump_prepare:309 26944 Preparing with io->period_size=164 io->buffer_size=3936 snd_pcm_socketdump_pointer:229 26944 hw_ptr: 0 () snd_pcm_socketdump_write:268 26944 snd_pcm_socketdump_write(): areas->step=16 areas->first=0 areas->addr=0x22f1f60 offset=0, size=164 io->nonblock=0 ...... Do you have any idea why the start callback function is not called at all ? cheers, stefan

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[alsa-devel] [PATCH] ASoC: DaVinci: Update suspend/resume support for McASP driver
by Chaithrika U S 03 Dec '09

03 Dec '09

Add clock enable and disable calls to resume and suspend respectively. Also add a member to the audio device data structure which tracks the clock status. Tested on DA850/OMAP-L138 EVM. For the purpose of testing, the patches[1] which add suspend-to-RAM support to DA850/OMAP-L138 SoC were applied. [1] http://linux.davincidsp.com/pipermail/davinci-linux-open-source/ 2009-November/016958.html Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-mcasp.c | 18 ++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 1 + sound/soc/davinci/davinci-pcm.c | 2 +- 3 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0a302e1..0d263f1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -767,14 +767,27 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, int ret = 0; switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + if (!dev->clk_active) { + clk_enable(dev->clk); + dev->clk_active = 1; + } + + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_mcasp_start(dev, substream->stream); break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: + davinci_mcasp_stop(dev, substream->stream); + if (dev->clk_active) { + clk_disable(dev->clk); + dev->clk_active = 0; + } + + break; + + case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: davinci_mcasp_stop(dev, substream->stream); break; @@ -866,6 +879,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } clk_enable(dev->clk); + dev->clk_active = 1; dev->base = (void __iomem *)IO_ADDRESS(mem->start); dev->op_mode = pdata->op_mode; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 582c924..e755b51 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -44,6 +44,7 @@ struct davinci_audio_dev { int sample_rate; struct clk *clk; unsigned int codec_fmt; + u8 clk_active; /* McASP specific data */ int tdm_slots; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index ad4d7f4..80c7fdf 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -49,7 +49,7 @@ static void print_buf_info(int slot, char *name) static struct snd_pcm_hardware pcm_hardware_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE), + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats = (SNDRV_PCM_FMTBIT_S16_LE), .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | -- 1.5.6

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[alsa-devel] io plugin: playback and capture over a socket
by Stefan Schoenleitner 03 Dec '09

03 Dec '09

Hi, I'm having difficulties writing a simple alsa plugin that plays and captures from a socket. The background is that there should be a "server" application that processes PCM data. In order to "connect" ALSA to that application I would like to use a socket. I looked at the bluez bluetooth audio plugin which seems to do a similar task. Basically the idea is that the alsa plugin transfer callback function just sends and receives PCM samples in periods to/from the socket (depending on whether the plugin is used for playback or capture). To control the timing and to increment the hw_ptr a separate thread in the alsa plugin is used. Basically for each period the thread will increment the hw_ptr and signal to the alsa plugin that the next period can be written. This signaling is done by using a pipe. As the plugin code is poll-based I guess it needs to poll both the socket and the pipe (which is used to communicate with the "timing" thread). For this reason the alsa plugin would only write to the socket if the socket is ready for writing _and_ the time for writing the next period has come (which will be signaled by a write to the pipe by the timing thread). Would such a implementation work so far ? At the moment I'm having some real difficulties as for some reason the start callback function that would start the thread is not called at all. My code is pretty much the same as in the bluez bluetooth audio plugin. cheers, stefan

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[alsa-devel] [PATCH 1/2] ASoC: sh_fsi: avoid using global variable
by Kuninori Morimoto 03 Dec '09

03 Dec '09

Current FSI driver use global variable to access device data. But this style will be broken if SuperH come with multiple FSI blocks in future. To solve this problem, this patch use cpu_dai->private_data. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/sh/fsi.c | 115 ++++++++++++++++++++++++++++------------------------ 1 files changed, 62 insertions(+), 53 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9c49c11..7506ef6 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -92,6 +92,7 @@ struct fsi_priv { void __iomem *base; struct snd_pcm_substream *substream; + struct fsi_master *master; int fifo_max; int chan; @@ -110,8 +111,6 @@ struct fsi_master { struct sh_fsi_platform_info *info; }; -static struct fsi_master *master; - /************************************************************************ @@ -166,7 +165,7 @@ static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); } -static int fsi_master_write(u32 reg, u32 data) +static int fsi_master_write(struct fsi_master *master, u32 reg, u32 data) { if ((reg < MREG_START) || (reg > MREG_END)) @@ -175,7 +174,7 @@ static int fsi_master_write(u32 reg, u32 data) return __fsi_reg_write((u32)(master->base + reg), data); } -static u32 fsi_master_read(u32 reg) +static u32 fsi_master_read(struct fsi_master *master, u32 reg) { if ((reg < MREG_START) || (reg > MREG_END)) @@ -184,7 +183,8 @@ static u32 fsi_master_read(u32 reg) return __fsi_reg_read((u32)(master->base + reg)); } -static int fsi_master_mask_set(u32 reg, u32 mask, u32 data) +static int fsi_master_mask_set(struct fsi_master *master, + u32 reg, u32 mask, u32 data) { if ((reg < MREG_START) || (reg > MREG_END)) @@ -200,43 +200,29 @@ static int fsi_master_mask_set(u32 reg, u32 mask, u32 data) ************************************************************************/ -static struct fsi_priv *fsi_get(struct snd_pcm_substream *substream) +static struct fsi_master *fsi_get_master(struct fsi_priv *fsi) { - struct snd_soc_pcm_runtime *rtd; - struct fsi_priv *fsi = NULL; - - if (!substream || !master) - return NULL; - - rtd = substream->private_data; - switch (rtd->dai->cpu_dai->id) { - case 0: - fsi = &master->fsia; - break; - case 1: - fsi = &master->fsib; - break; - } - - return fsi; + return fsi->master; } static int fsi_is_port_a(struct fsi_priv *fsi) { - /* return - * 1 : port a - * 0 : port b - */ + return fsi->master->base == fsi->base; +} - if (fsi == &master->fsia) - return 1; +static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *machine = rtd->dai; + struct snd_soc_dai *dai = machine->cpu_dai; - return 0; + return dai->private_data; } static u32 fsi_get_info_flags(struct fsi_priv *fsi) { int is_porta = fsi_is_port_a(fsi); + struct fsi_master *master = fsi_get_master(fsi); return is_porta ? master->info->porta_flags : master->info->portb_flags; @@ -314,27 +300,30 @@ static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play) static void fsi_irq_enable(struct fsi_priv *fsi, int is_play) { u32 data = fsi_port_ab_io_bit(fsi, is_play); + struct fsi_master *master = fsi_get_master(fsi); - fsi_master_mask_set(IMSK, data, data); - fsi_master_mask_set(IEMSK, data, data); + fsi_master_mask_set(master, IMSK, data, data); + fsi_master_mask_set(master, IEMSK, data, data); } static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) { u32 data = fsi_port_ab_io_bit(fsi, is_play); + struct fsi_master *master = fsi_get_master(fsi); - fsi_master_mask_set(IMSK, data, 0); - fsi_master_mask_set(IEMSK, data, 0); + fsi_master_mask_set(master, IMSK, data, 0); + fsi_master_mask_set(master, IEMSK, data, 0); } static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable) { u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4); + struct fsi_master *master = fsi_get_master(fsi); if (enable) - fsi_master_mask_set(CLK_RST, val, val); + fsi_master_mask_set(master, CLK_RST, val, val); else - fsi_master_mask_set(CLK_RST, val, 0); + fsi_master_mask_set(master, CLK_RST, val, 0); } static void fsi_irq_init(struct fsi_priv *fsi, int is_play) @@ -355,23 +344,23 @@ static void fsi_irq_init(struct fsi_priv *fsi, int is_play) fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR); /* clear interrupt factor */ - fsi_master_mask_set(INT_ST, data, 0); + fsi_master_mask_set(fsi_get_master(fsi), INT_ST, data, 0); } -static void fsi_soft_all_reset(void) +static void fsi_soft_all_reset(struct fsi_master *master) { - u32 status = fsi_master_read(SOFT_RST); + u32 status = fsi_master_read(master, SOFT_RST); /* port AB reset */ status &= 0x000000ff; - fsi_master_write(SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status); mdelay(10); /* soft reset */ status &= 0x000000f0; - fsi_master_write(SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status); status |= 0x00000001; - fsi_master_write(SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status); mdelay(10); } @@ -517,12 +506,13 @@ static int fsi_data_pop(struct fsi_priv *fsi) static irqreturn_t fsi_interrupt(int irq, void *data) { - u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; - u32 int_st = fsi_master_read(INT_ST); + struct fsi_master *master = data; + u32 status = fsi_master_read(master, SOFT_RST) & ~0x00000010; + u32 int_st = fsi_master_read(master, INT_ST); /* clear irq status */ - fsi_master_write(SOFT_RST, status); - fsi_master_write(SOFT_RST, status | 0x00000010); + fsi_master_write(master, SOFT_RST, status); + fsi_master_write(master, SOFT_RST, status | 0x00000010); if (int_st & INT_A_OUT) fsi_data_push(&master->fsia); @@ -533,7 +523,7 @@ static irqreturn_t fsi_interrupt(int irq, void *data) if (int_st & INT_B_IN) fsi_data_pop(&master->fsib); - fsi_master_write(INT_ST, 0x0000000); + fsi_master_write(master, INT_ST, 0x0000000); return IRQ_HANDLED; } @@ -548,7 +538,7 @@ static irqreturn_t fsi_interrupt(int irq, void *data) static int fsi_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); const char *msg; u32 flags = fsi_get_info_flags(fsi); u32 fmt; @@ -667,7 +657,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, static void fsi_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; fsi_irq_disable(fsi, is_play); @@ -679,7 +669,7 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); struct snd_pcm_runtime *runtime = substream->runtime; int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int ret = 0; @@ -760,7 +750,7 @@ static int fsi_hw_free(struct snd_pcm_substream *substream) static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct fsi_priv *fsi = fsi_get(substream); + struct fsi_priv *fsi = fsi_get_priv(substream); long location; location = (fsi->byte_offset - 1); @@ -870,10 +860,16 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform); ************************************************************************/ static int fsi_probe(struct platform_device *pdev) { + struct fsi_master *master; struct resource *res; unsigned int irq; int ret; + if (0 != pdev->id) { + dev_err(&pdev->dev, "current fsi support id 0 only now\n"); + return -ENODEV; + } + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); irq = platform_get_irq(pdev, 0); if (!res || !irq) { @@ -899,15 +895,19 @@ static int fsi_probe(struct platform_device *pdev) master->irq = irq; master->info = pdev->dev.platform_data; master->fsia.base = master->base; + master->fsia.master = master; master->fsib.base = master->base + 0x40; + master->fsib.master = master; pm_runtime_enable(&pdev->dev); pm_runtime_resume(&pdev->dev); fsi_soc_dai[0].dev = &pdev->dev; + fsi_soc_dai[0].private_data = &master->fsia; fsi_soc_dai[1].dev = &pdev->dev; + fsi_soc_dai[1].private_data = &master->fsib; - fsi_soft_all_reset(); + fsi_soft_all_reset(master); ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); if (ret) { @@ -937,6 +937,10 @@ exit: static int fsi_remove(struct platform_device *pdev) { + struct fsi_master *master; + + master = fsi_get_master(fsi_soc_dai[0].private_data); + snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); snd_soc_unregister_platform(&fsi_soc_platform); @@ -946,7 +950,12 @@ static int fsi_remove(struct platform_device *pdev) iounmap(master->base); kfree(master); - master = NULL; + + fsi_soc_dai[0].dev = NULL; + fsi_soc_dai[0].private_data = NULL; + fsi_soc_dai[1].dev = NULL; + fsi_soc_dai[1].private_data = NULL; + return 0; } -- 1.6.3.3

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 03 Dec '09

03 Dec '09

The following changes since commit a22eaf4ce106404f6c5283da30b4d514ede964c1: Takashi Iwai (1): ASoC: Revert missing reset_err in wm97*.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Anuj Aggarwal (1): ASoC: AIC23: Fixing infinite loop in resume path Daniel Mack (2): ALSA: ARM: add Raumfeld audio support ASoC: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API Kuninori Morimoto (2): sh: ms7724se: Add runtime PM support for FSI ASoC: sh: fsi: Add runtime PM support Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 arch/sh/boards/mach-se/7724/setup.c | 3 + sound/soc/codecs/tlv320aic23.c | 3 +- sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/raumfeld.c | 335 +++++++++++++++++++++++++++++++++++ sound/soc/sh/fsi.c | 39 +++-- 6 files changed, 375 insertions(+), 16 deletions(-) create mode 100644 sound/soc/pxa/raumfeld.c

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[alsa-devel] [PATCH 0/2] ASoC: au1x updates for 2.6.33
by Manuel Lauss 03 Dec '09

03 Dec '09

Hi Mark, The following two patches fix 2 issues in the updated au1x asoc drivers in your/Takashis queue for 2.6.33. If possible, please either fold them into commit 0f83d639d84c99a775c60696dbde77372c2cf4ac ("au1x: convert to platform drivers") found in Takashi's tree. Thank you! Manuel Lauss Manuel Lauss (2): ASoC: au1x: dbdma2: fix oops on soc device removal. ASoC: au1x: dbdma2: plug memleak in pcm device creation error path sound/soc/au1x/dbdma2.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-)

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[alsa-devel] ASoC: OMAP mcbsp: dai->dev is missing warning
by Peter Ujfalusi 03 Dec '09

03 Dec '09

Hello, Recently the following warnings appeared during the bootup on OMAP platforms: No device for DAI omap-mcbsp-dai-0 No device for DAI omap-mcbsp-dai-1 No device for DAI omap-mcbsp-dai-2 No device for DAI omap-mcbsp-dai-3 No device for DAI omap-mcbsp-dai-4 I have taken a look, and it is printed, because the dai->dev is NULL for OMAP McBSP DAIs. Also the comment there states, that the dai->dev will become mandatory in the future. At the moment the platform device/driver is not used for the sound/soc/omap/omap-mcbsp, it is used for the arch/arm/plat-omap/mcbsp.c, so we really don't have possibility at the moment to provide a valid dai->dev for ASoC at the moment. Mark: Do you have some estimates, when the dai->dev will become mandatory? Jarkko: Shall we start to think about the possibilities to provide a valid dai- >dev for the mcbsp DAIs? -- Péter

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