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

[alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Joe Perches 06 Jun '12

06 Jun '12

KERN_<LEVEL> currently takes up 3 bytes. Shrink the kernel size by using an ASCII SOH and then the level byte. Remove the need for KERN_CONT. Convert directly embedded uses of <.> to KERN_<LEVEL> Joe Perches (8): printk: Add generic functions to find KERN_<LEVEL> headers printk: Add kern_levels.h to make KERN_<LEVEL> available for asm use arch: Remove direct definitions of KERN_<LEVEL> uses btrfs: Use printk_get_level and printk_skip_level, add __printf, fix fallout sound: Use printk_get_level and printk_skip_level staging: wlags49_h2: Remove direct declarations of KERN_<LEVEL> prefixes printk: Convert the format for KERN_<LEVEL> to a 2 byte pattern printk: Remove the now unnecessary "C" annotation for KERN_CONT arch/arm/lib/io-acorn.S | 3 +- arch/arm/vfp/vfphw.S | 7 +++-- arch/frv/kernel/kernel_thread.S | 2 +- drivers/staging/wlags49_h2/hcf.c | 8 +++--- drivers/staging/wlags49_h2/wl_main.c | 4 +- fs/btrfs/ctree.h | 13 ++++++++++ fs/btrfs/disk-io.c | 2 +- fs/btrfs/relocation.c | 2 +- fs/btrfs/super.c | 41 +++++++++++++++++++++++++++++----- include/linux/kern_levels.h | 25 ++++++++++++++++++++ include/linux/printk.h | 41 +++++++++++++++++++-------------- kernel/printk.c | 14 +++++++---- sound/core/misc.c | 13 +++++++--- 13 files changed, 130 insertions(+), 45 deletions(-) create mode 100644 include/linux/kern_levels.h -- 1.7.8.111.gad25c.dirty

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Re: [alsa-devel] [PATCH 0/8] Rework KERN_<LEVEL>
by Andrew Morton 06 Jun '12

06 Jun '12

On Wed, 06 Jun 2012 00:55:00 +0200 Kay Sievers <kay(a)vrfy.org> wrote: > On Tue, 2012-06-05 at 14:28 -0700, Andrew Morton wrote: > > > devkmsg_writev() does weird and wonderful things with > > facilities/levels. That function incorrectly returns "success" when > > copy_from_user() faults, btw. > > Oh. Better? > > Thanks, > Kay > > > From: Kay Sievers <kay(a)vrfy.org> > Subject: kmsg: /dev/kmsg - properly return possible copy_from_user() failure > > Reported-By: Andrew Morton <akpm(a)linux-foundation.org> > Signed-off-by: Kay Sievers <kay(a)vrfy.org > --- > printk.c | 4 +++- > 1 file changed, 3 insertions(+), 1 deletion(-) > > diff --git a/kernel/printk.c b/kernel/printk.c > index 32462d2..6bdacab 100644 > --- a/kernel/printk.c > +++ b/kernel/printk.c > @@ -365,8 +365,10 @@ static ssize_t devkmsg_writev(struct kiocb *iocb, const struct iovec *iv, > > line = buf; > for (i = 0; i < count; i++) { > - if (copy_from_user(line, iv[i].iov_base, iv[i].iov_len)) > + if (copy_from_user(line, iv[i].iov_base, iv[i].iov_len)) { > + ret = -EFAULT; > goto out; > + } > line += iv[i].iov_len; > } Strictly speaking, when write() encounters an error it should return number-of-bytes-written, or an errno if it wrote zero bytes. So something like --- a/kernel/printk.c~a +++ a/kernel/printk.c @@ -355,7 +355,7 @@ static ssize_t devkmsg_writev(struct kio int level = default_message_loglevel; int facility = 1; /* LOG_USER */ size_t len = iov_length(iv, count); - ssize_t ret = len; + ssize_t ret; if (len > LOG_LINE_MAX) return -EINVAL; @@ -365,8 +365,12 @@ static ssize_t devkmsg_writev(struct kio line = buf; for (i = 0; i < count; i++) { - if (copy_from_user(line, iv[i].iov_base, iv[i].iov_len)) + if (copy_from_user(line, iv[i].iov_base, iv[i].iov_len)) { + ret = line - buf; + if (!ret) + ret = -EFAULT; goto out; + } line += iv[i].iov_len; } @@ -396,6 +400,7 @@ static ssize_t devkmsg_writev(struct kio line[len] = '\0'; printk_emit(facility, level, NULL, 0, "%s", line); + ret = 0; out: kfree(buf); return ret; _

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[alsa-devel] [PATCH] ALSA: hda - add power states information in proc
by Wang Xingchao 06 Jun '12

06 Jun '12

add more power states information: - reset status - clock stop ok - power states error Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_proc.c | 6 ++++++ 2 files changed, 9 insertions(+) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 4fc3960..71864cdd 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -323,6 +323,9 @@ enum { #define AC_PWRST_D1 0x01 #define AC_PWRST_D2 0x02 #define AC_PWRST_D3 0x03 +#define AC_PWRST_ERROR (1<<8) +#define AC_PWRST_CLK_STOP_OK (1<<9) +#define AC_PWRST_SETTING_RESET (1<<10) /* Processing capabilies */ #define AC_PCAP_BENIGN (1<<0) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index e59e2f0..5511de1 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -459,6 +459,12 @@ static void print_power_state(struct snd_info_buffer *buffer, get_pwr_state(pwr & AC_PWRST_SETTING), get_pwr_state((pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT)); + if (pwr & AC_PWRST_ERROR) + snd_iprintf(buffer, " Power state error\n"); + if (pwr & AC_PWRST_CLK_STOP_OK) + snd_iprintf(buffer, " PS: clock stop ok\n"); + if (pwr & AC_PWRST_SETTING_RESET) + snd_iprintf(buffer, " Power state setting reset\n"); } static void print_unsol_cap(struct snd_info_buffer *buffer, -- 1.7.9.5

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[alsa-devel] Confusing about Playback/Capture, CODEC/CODEC links, and snd_soc_dapm_link_dai_widgets()
by Stephen Warren 05 Jun '12

05 Jun '12

Mark, Liam, recall the Tegra AHUB structure: > +-------------+ +-----+ +--------------+ +-----+ +------+ > | FIFO pair 0 |<->| CIF |<->| Cross-bar |<->| CIF |<->| I2S0 |<-> External IO > +-------------+ +-----+ | (the "AHUB") | +-----+ +------+ > . . . | | . . . > +-------------+ +-----+ | | +-----+ +------+ > | FIFO pair 3 |<->| CIF |<->| |<->| CIF |<->| I2S4 |<-> External IO > +-------------+ +-----+ | | +-----+ +------+ > | | > | | +-----+ +-------+ > | |<->| CIF |<->| SPDIF |<-> External IO > | | +-----+ +-------+ > | | > | | +-----+ +-------+ > | |<->| CIF |<->| DAM | > | | +-----+ +-------+ > +--------------+ (CIF is AHUB XBAR Client InterFace; basically the DAI or DAI link) I have created a driver for the 4 FIFOs on the left, which exposes 4 (CPU) DAIs. I have created a driver for the AHUB XBAR, which is an ASoC CODEC, and has a DAI for each of the CIFs. There's a CODEC/CODEC link between each FIFO DAI and the relevant AHUB CIF DAI. In order to connect the I2S devices to the AHUB XBAR, I had to make that a CODEC too. This exposes two DAIs; the CIF DAI to connect to the AHUB, and the DAP (Digital Audio Port) DAI to represent the I/O pins on the Tegra package that are connected to the external codec. I have an issue with the paths that snd_soc_dapm_link_dai_widgets() sets up inside both the AHUB XBAR and the I2S. I'll use the I2S to explain since it's internally much simpler. The two DAIs in I2S are roughly: > static const struct snd_soc_dai_driver tegra30_i2s_dais[] = { > { > .playback = { > .stream_name = "DAP Playback", > }, > .capture = { > .stream_name = "DAP Capture", > }, > }, > { > .playback = { > .stream_name = "CIF Playback", > }, > .capture = { > .stream_name = "CIF Capture", > }, > }, > }; For the CIF-side DAI, I instantiated widgets for the AIFs: > static const struct snd_soc_dapm_widget tegra30_i2s_widgets[] = { > SND_SOC_DAPM_AIF_IN("CIF RX", "CIF Playback", 0, SND_SOC_NOPM, 0, 0), > SND_SOC_DAPM_AIF_OUT("CIF TX", "CIF Capture", 0, SND_SOC_NOPM, 0, 0), This works fine; snd_soc_dapm_link_dai_widgets() create a route from the DAI widget "DAP Playback" that feeds into the AIF_IN widget "CIF RX". "works fine" means that the ASoC debugfs files in "asoc/$card/tegra30-i2s.1/dapm/CIF {Playback,RX}" show the expected connections. The DAPM routes inside the I2S are: > static const struct snd_soc_dapm_route tegra30_i2s_routes[] = { > { "DAP TX", NULL, "CIF RX" }, > { "CIF TX", NULL, "DAP RX" }, > }; This also works fine. Now, I instantiated two AIF widgets to interface with the DAP-side DAI: > SND_SOC_DAPM_AIF_IN("DAP RX", "DAP Capture", 0, SND_SOC_NOPM, 0, 0), > SND_SOC_DAPM_AIF_OUT("DAP TX", "DAP Playback", 0, SND_SOC_NOPM, 0, 0), However, this causes snd_soc_dapm_link_dai_widgets() to set up some unexpected paths; "DAP Playback" ends up feeding *into* "DAP TX", rather than feeding from/out of it... > # cat CIF\ Playback > CIF Playback: Off in 0 out 0 > stream CIF Playback inactive > out "static" "CIF RX" > out "static" "CIF RX" > # cat CIF\ RX > CIF RX: Off in 0 out 0 > stream CIF Playback inactive > in "static" "CIF Playback" > in "static" "CIF Playback" > out "static" "DAP TX" > # cat DAP\ TX > DAP TX: Off in 0 out 0 > stream DAP Playback inactive > in "static" "DAP Playback" <<<<<<<<<< > in "static" "DAP Playback" <<<<<<<<<< > in "static" "CIF RX" > # cat DAP\ Playback > DAP Playback: Off in 0 out 0 > stream DAP Playback inactive > out "static" "DAP TX" > out "static" "DAP TX" Now, I can fake this out by swapping the stream names on the DAP-side AIF widgets: > SND_SOC_DAPM_AIF_IN("DAP RX", "DAP Playback", 0, SND_SOC_NOPM, 0, 0), > SND_SOC_DAPM_AIF_OUT("DAP TX", "DAP Capture", 0, SND_SOC_NOPM, 0, 0), Which yields the expected paths in debugfs: > # cat CIF\ Playback > CIF Playback: Off in 0 out 0 > stream CIF Playback inactive > out "static" "CIF RX" > out "static" "CIF RX" > # cat CIF\ RX > CIF RX: Off in 0 out 0 > stream CIF Playback inactive > in "static" "CIF Playback" > in "static" "CIF Playback" > out "static" "DAP TX" > # cat DAP\ TX > DAP TX: Off in 0 out 0 > stream DAP Capture inactive > in "static" "CIF RX" > out "static" "DAP Capture" > out "static" "DAP Capture" > # cat DAP\ Capture > DAP Capture: Off in 0 out 0 > stream DAP Capture inactive > in "static" "DAP TX" > in "static" "DAP TX" But I'm not sure if that's quite correct. In the DAI definitions, is the .playback sub-structure always meant to represent: 1) Playback from the CPU's perspective, in which case the first set of DAP AIF definitions above would be correct or: 2) Is it more that Playback==RX_into_codec, Capture==TX_from_codec? This appears to be supported by the fact that the paths get set up correctly with the second set of AIF definitions above. But if (2) is correct, I wonder why soc_dapm_stream_event()'s first if statement appears to consider the playback_widget of both sides of the DAI to be coupled; wouldn't one side's playback widget be coupled to the other side's capture widget? As an aside, I'm not sure if it's conceptually correct to talk about playback or capture any more (beyond the initial CPU DAI) with arbitrary CODEC/CODEC links, since who knows what kind of routing/CPU->CPU loopbacks/external CODEC->CODEC loopbacks/... might exist in the CODECs? Either way though, something is still not working correctly even when the expected paths show up in debugfs. When I start playback on a PCM exposed by the DMA FIFO driver, I see the relevant AHUB XBAR's stream turned on, and a route exists to the relevant AIF_IN widget, but that widget is not considered to have any outward connections by is_connected_output_ep(), which I think is what is causing it and none of the rest of the path to turn on: (from asoc/$card/asoc/tegra30-ahub-xbar/dapm) > # cat APBIF0\ Playback > APBIF0 Playback: On in 1 out 1 > stream APBIF0 Playback active > out "static" "APBIF0 RX" > out "static" "APBIF0 RX" > # cat APBIF0\ RX > APBIF0 RX: Off in 2 out 0 <<<<< last value shouldn't be 0? > stream APBIF0 Playback inactive <<<<< still inactive? > in "static" "APBIF0 Playback" > in "static" "APBIF0 Playback" > out "APBIF0 RX" "I2S0 TX Mux" even though when I look at all the debugfs files, all the expected paths are there, at least within the AHUB XBAR and I2S drivers, and the external DAI links in the machine driver all probed as expected and bound all the components together. Probably related, the I2S and WM8903 (DAI) drivers aren't being called by the ASoC core to initialize themselves for playback either; pretty much all that happens is that DMA is started. Is the machine driver responsible for this? At this point, I'm not sure whether I have a gross mis-understanding of how this is supposed to work, or whether there are simply bits missing from the DAPM code to fully support CODEC/CODEC links, rather than CPU/CODEC links. Probably slightly related to all of this, but is the following code correct: > int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, > struct snd_soc_dapm_widget_list **list) ... > if (stream == SNDRV_PCM_STREAM_PLAYBACK) > paths = is_connected_output_ep(dai->playback_widget, list); > else > paths = is_connected_input_ep(dai->playback_widget, list); I would have expected this to use capture_widget on the final line, but I haven't thought about this in detail, just noticed the lack of symmetry by very brief inspection. Thanks for any help. Sorry for the long email.

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[alsa-devel] dapm_widget_power_read_file() symmetry issue?
by Stephen Warren 05 Jun '12

05 Jun '12

In dapm_widget_power_read_file(), there is first: > list_for_each_entry(p, &w->sources, list_sink) { > if (p->connected && !p->connected(w, p->sink)) > continue; then right after that: > list_for_each_entry(p, &w->sinks, list_source) { > if (p->connected && !p->connected(w, p->sink)) > continue; Should that last call to p->connected() be p->connected(p->source, w), since the function's parameters are (source, sink)?

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[alsa-devel] [PATCH] ALSA: au88x0 - Remove unused "Master Mono" Playback Volume and Playback Switch of ac97 codec
by Raymond Yau 05 Jun '12

05 Jun '12

Remove "Master Mono Playback Volume" and "Master Mono Playback Switch" of ac97 mixer since au88x0 does no use "Master Mono Pin" of AC97 codec even au88x0 support mono playback

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[alsa-devel] To disable dma_area caching (twice)
by MASAO TAKAHASHI 05 Jun '12

05 Jun '12

Hi everybody. Please advice me to solve my problem. 1. My environment Hardware Alpha-project MS104-SH4AG with SH7764 kernel sh-linux-2.6.29 ALSA driver sh7764-pcm.c developped by Manuel Lauss <mano(a)roarinelk.homelinux.net> 2. My problem I am using the above ALSA driver for SH7764 SSI. I have found the runtime->dma_area address refers DMA area via D-cache. So, playback start is too late after executing snd_pcm_writei(). Until D-cache writeback is done, sound data stays in D-cache. I want to read and write from/to main memory without D-cache. 3. My idea to solve this problem 3-1 change allocating type of DMA area as follows snd_pcm_lib_preallocate_pages_for_all( pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), 64*1024, 64*1024); is changed to snd_pcm_lib_preallocate_pages_for_all( pcm, SNDRV_DMA_TYPE_DEV, <----------- here snd_dma_continuous_data(GFP_KERNEL), 64*1024, 64*1024); to let dma_addr pointing P2SEG area without caching 3-2 program internal DMA controller start address with dma_addr ssidma_outl(sd, (virt_to_phys)(u32)runtime->dma_addr, SSIRDMADR); 3-3 add mmap_data_fault routine static int sh7764_mmap_data_fault( structvm_area_struct*area, struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; unsigned long offset; struct page * page; void *vaddr; size_t dma_bytes; pr_debug("mmap_data_fault\n"); if (substream == NULL) return VM_FAULT_SIGBUS; runtime = substream->runtime; offset = vmf->pgoff << PAGE_SHIFT; dma_bytes = PAGE_ALIGN(runtime->dma_bytes); if (offset > dma_bytes - PAGE_SIZE) return VM_FAULT_SIGBUS; if (substream->ops->page) { page = substream->ops->page(substream, offset); if (!page) return VM_FAULT_SIGBUS; } else { vaddr = runtime->dma_addr + offset; page = virt_to_page(vaddr); } get_page(page); vmf->page = page; pr_debug("mmap_data_fault:page = %x\n", (unsigned int)page); return 0; } static struct vm_operations_struct sh7764_pcm_vm_ops_data = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, .fault = sh7764_mmap_data_fault, }; 3-3 add sh7764_pcm_mmap routine static int sh7764_pcm_mmap( struct snd_pcm_substream *substream, struct vm_area_struct *area) { unsigned long offset; unsigned long mp; unsigned long virt; int map_size; struct page *map, *mapend, *ip; struct snd_pcm_runtime *runtime; int ret; #ifdef pgprot_noncached area->vm_page_prot = pgprot_noncached(area->vm_page_prot); #endif offset = area->vm_pgoff << PAGE_SHIFT; runtime = substream->runtime; map_size = runtime->dma_bytes; mp = virt_to_phys((unsigned long)runtime->dma_addr)); map = virt_to_page(mp); mapend = virt_to_page(mp+map_size - 1); for (ip = map; ip <= mapend; ip++){ SetPageReserved(ip); area->vm_ops = &sh7764_pcm_vm_ops_data; area->vm_private_data = substream; area->vm_flags |= VM_IO; ret = io_remap_pfn_range(area, area->vm_start, mp>>PAGE_SHIFT, area->vm_end - area->vm_start, area->vm_page_prot); if (ret ){ pr_debug("sh7764-pcm.c:remap_pfn_range returned EAGAIN\n"); return -EAGAIN; } atomic_inc(&substream->mmap_count); pr_debug("mmap_count=%u\n", substream->mmap_count); return 0; } static struct snd_pcm_ops sh7764_ssidma_ops = { .open = sh7764_ssidma_open, .close = sh7764_ssidma_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = sh7764_hw_params, .hw_free = sh7764_hw_free, .prepare = sh7764_prepare, .trigger = sh7764_trigger, .pointer = sh7764_pos, .mmap = sh7764_pcm_mmap, }; ------------------------------------------------------------ Sorry, so long contents. My solution is right? or Is there any better solution ? Best regards ---------- MASAO TAKAHASHI <masao-takahashi(a)kanno.co.jp> KANNO WORKS CO., LTD. JAPAN TEL 093-436-2330

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[alsa-devel] [PATCH v2 resend] ASoC: omap-abe-twl6040: Add device tree support
by Peter Ujfalusi 05 Jun '12

05 Jun '12

When the board boots with device tree the driver will receive the name of the card, DAPM routing map, phandle for the audio components described in the dts file, mclk speed, and the possibility of detecting the jack detection. The card will be set up based on this information. Since the routing is provided via DT we can mark the card fully routed so core can take care of disconnecting the unused pins. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Liam. This patch was part of the series to add the Device Tree support for ABE+TWL6040 based audio. The intro mail for the series was: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-May/052103.html Thank you, Peter .../devicetree/bindings/sound/omap-abe-twl6040.txt | 91 ++++++++++++ sound/soc/omap/omap-abe-twl6040.c | 145 ++++++++++++++++---- 2 files changed, 206 insertions(+), 30 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt diff --git a/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt new file mode 100644 index 0000000..65dec87 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt @@ -0,0 +1,91 @@ +* Texas Instruments OMAP4+ and twl6040 based audio setups + +Required properties: +- compatible: "ti,abe-twl6040" +- ti,model: Name of the sound card ( for example "SDP4430") +- ti,mclk-freq: MCLK frequency for HPPLL operation +- ti,mcpdm: phandle for the McPDM node +- ti,twl6040: phandle for the twl6040 core node +- ti,audio-routing: List of connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. + +Optional properties: +- ti,dmic: phandle for the OMAP dmic node if the machine have it connected +- ti,jack_detection: Need to be set to <1> if the board capable to detect jack + insertion, removal. + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Earphone Spk + * Ext Spk + * Line Out + * Vibrator + * Headset Mic + * Main Handset Mic + * Sub Handset Mic + * Line In + * Digital Mic + +twl6040 pins: + * HSOL + * HSOR + * EP + * HFL + * HFR + * AUXL + * AUXR + * VIBRAL + * VIBRAR + * HSMIC + * MAINMIC + * SUBMIC + * AFML + * AFMR + + * Headset Mic Bias + * Main Mic Bias + * Digital Mic1 Bias + * Digital Mic2 Bias + +Digital mic pins: + * DMic + +Example: + +sound { + compatible = "ti,abe-twl6040"; + ti,model = "SDP4430"; + + ti,jack-detection = <1>; + ti,mclk-freq = <38400000>; + + ti,mcpdm = <&mcpdm>; + ti,dmic = <&dmic>; + + ti,twl6040 = <&twl6040>; + + /* Audio routing */ + ti,audio-routing = + "Headset Stereophone", "HSOL", + "Headset Stereophone", "HSOR", + "Earphone Spk", "EP", + "Ext Spk", "HFL", + "Ext Spk", "HFR", + "Line Out", "AUXL", + "Line Out", "AUXR", + "Vibrator", "VIBRAL", + "Vibrator", "VIBRAR", + "HSMIC", "Headset Mic", + "Headset Mic", "Headset Mic Bias", + "MAINMIC", "Main Handset Mic", + "Main Handset Mic", "Main Mic Bias", + "SUBMIC", "Sub Handset Mic", + "Sub Handset Mic", "Main Mic Bias", + "AFML", "Line In", + "AFMR", "Line In", + "DMic", "Digital Mic", + "Digital Mic", "Digital Mic1 Bias"; +}; diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 9d93793..1995a7f 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -25,6 +25,7 @@ #include <linux/mfd/twl6040.h> #include <linux/platform_data/omap-abe-twl6040.h> #include <linux/module.h> +#include <linux/of.h> #include <sound/core.h> #include <sound/pcm.h> @@ -43,6 +44,8 @@ struct abe_twl6040 { int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ + + struct platform_device *dmic_codec_dev; }; static int omap_abe_hw_params(struct snd_pcm_substream *substream, @@ -185,17 +188,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) int hs_trim; int ret = 0; - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - 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_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - /* * Configure McPDM offset cancellation based on the HSOTRIM value from * twl6040. @@ -216,6 +208,24 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } + /* + * NULL pdata means we booted with DT. In this case the routing is + * provided and the card is fully routed, no need to mark pins. + */ + if (!pdata) + return ret; + + /* Disable not connected paths if not used */ + twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); + 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_hsmic, "Headset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); + return ret; } @@ -270,52 +280,116 @@ static struct snd_soc_card omap_abe_card = { static __devinit int omap_abe_probe(struct platform_device *pdev) { struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); + struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; struct abe_twl6040 *priv; int num_links = 0; - int ret; + int ret = 0; card->dev = &pdev->dev; - if (!pdata) { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; - } - priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); if (priv == NULL) return -ENOMEM; - if (pdata->card_name) { - card->name = pdata->card_name; + priv->dmic_codec_dev = ERR_PTR(-EINVAL); + + if (node) { + struct device_node *dai_node; + + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + ret = snd_soc_of_parse_audio_routing(card, + "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, + "Error while parsing DAPM routing\n"); + return -EINVAL; + } + + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_dai_of_node = dai_node; + + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_dai_of_node = dai_node; + + priv->dmic_codec_dev = platform_device_register_simple( + "dmic-codec", -1, NULL, 0); + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, + "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); + } + } else { + num_links = 1; + } + + of_property_read_u32(node, "ti,jack-detection", + &priv->jack_detection); + of_property_read_u32(node, "ti,mclk-freq", + &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; + } + + omap_abe_card.fully_routed = 1; + } else if (pdata) { + if (pdata->card_name) { + card->name = pdata->card_name; + } else { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + if (pdata->has_dmic) + num_links = 2; + else + num_links = 1; + + priv->jack_detection = pdata->jack_detection; + priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Card name is not provided\n"); + dev_err(&pdev->dev, "Missing pdata\n"); return -ENODEV; } - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; - if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); - return -ENODEV; + ret = -ENODEV; + goto err_unregister; } - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - card->dai_link = abe_twl6040_dai_links; card->num_links = num_links; snd_soc_card_set_drvdata(card, priv); ret = snd_soc_register_card(card); - if (ret) + if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + goto err_unregister; + } + + return 0; + +err_unregister: + if (!IS_ERR(priv->dmic_codec_dev)) + platform_device_unregister(priv->dmic_codec_dev); return ret; } @@ -323,17 +397,28 @@ static __devinit int omap_abe_probe(struct platform_device *pdev) static int __devexit omap_abe_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); + struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); snd_soc_unregister_card(card); + if (!IS_ERR(priv->dmic_codec_dev)) + platform_device_unregister(priv->dmic_codec_dev); + return 0; } +static const struct of_device_id omap_abe_of_match[] = { + {.compatible = "ti,abe-twl6040", }, + { }, +}; +MODULE_DEVICE_TABLE(of, omap_abe_of_match); + static struct platform_driver omap_abe_driver = { .driver = { .name = "omap-abe-twl6040", .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, + .of_match_table = omap_abe_of_match, }, .probe = omap_abe_probe, .remove = __devexit_p(omap_abe_remove), -- 1.7.8.6

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[alsa-devel] [PATCH v2 0/4] omap-abe-twl6040: Device tree support
by Peter Ujfalusi 05 Jun '12

05 Jun '12

Hello, Changes since v1: - Binding docukentation updated with available pins and connectors - dts section for abe-twl6040 sound configuration has been changed Instead of dai link pairs only the nodes probed via DT have phandle: ti,mcpdm; ti,dmic, ti,twl6040. The driver in turn is responsible to handle the provided information. To provide dai link pairs (cpu dai - codec) we would need Linux (ASoC) specific nodes in DT for the dmic codec and twl6040 ASoC codec driver. If the DT section have ti,dmic phandle the ASoC machine driver will register the device to load the ASoC dmic-codec runtime. Example dts section for audio support (SDP4430): sound { compatible = "ti,abe-twl6040"; ti,model = "SDP4430"; ti,jack-detection = <1>; ti,mclk-freq = <38400000>; ti,mcpdm = <&mcpdm>; ti,dmic = <&dmic>; ti,twl6040 = <&twl6040>; /* Audio routing */ ti,audio-routing = "Headset Stereophone", "HSOL", "Headset Stereophone", "HSOR", "Earphone Spk", "EP", "Ext Spk", "HFL", "Ext Spk", "HFR", "Line Out", "AUXL", "Line Out", "AUXR", "Vibrator", "VIBRAL", "Vibrator", "VIBRAR", "HSMIC", "Headset Mic", "Headset Mic", "Headset Mic Bias", "MAINMIC", "Main Handset Mic", "Main Handset Mic", "Main Mic Bias", "SUBMIC", "Sub Handset Mic", "Sub Handset Mic", "Main Mic Bias", "AFML", "Line In", "AFMR", "Line In", "DMic", "Digital Mic", "Digital Mic", "Digital Mic1 Bias"; }; Regards, Peter --- Peter Ujfalusi (4): ASoC: omap-abe-twl6040: Keep only one snd_soc_dai_link structure ASoC: omap-abe-twl6040: Move Digital Mic widget into dapm table ASoC: omap-abe-twl6040: Introduce driver data for runtime parameters ASoC: omap-abe-twl6040: Add device tree support .../devicetree/bindings/sound/omap-abe-twl6040.txt | 91 +++++++++ sound/soc/omap/omap-abe-twl6040.c | 195 ++++++++++++++------ 2 files changed, 230 insertions(+), 56 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt -- 1.7.8.6

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[alsa-devel] [PATCH 1/2] ASoC: wm8994: Ensure all AIFnCLK events are run from the _late variants
by Mark Brown 05 Jun '12

05 Jun '12

Ensure that all the actions get taken at appropriate times by calling the _PRE and _POST events for the aifNclk_ev functions explicitly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 993639d..6d3a865 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1190,17 +1190,19 @@ static int late_enable_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (wm8994->aif1clk_enable) { - aif1clk_ev(w, kcontrol, event); + aif1clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMU); snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, WM8994_AIF1CLK_ENA); + aif1clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMU); wm8994->aif1clk_enable = 0; } if (wm8994->aif2clk_enable) { - aif2clk_ev(w, kcontrol, event); + aif2clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMU); snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, WM8994_AIF2CLK_ENA); + aif2clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMU); wm8994->aif2clk_enable = 0; } break; @@ -1221,15 +1223,17 @@ static int late_disable_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMD: if (wm8994->aif1clk_disable) { + aif1clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMD); snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, 0); - aif1clk_ev(w, kcontrol, event); + aif1clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMD); wm8994->aif1clk_disable = 0; } if (wm8994->aif2clk_disable) { + aif2clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMD); snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, 0); - aif2clk_ev(w, kcontrol, event); + aif2clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMD); wm8994->aif2clk_disable = 0; } break; -- 1.7.10

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