May 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: include/sound/asound.h whitespace fixups
by Daniel Mack 11 May '10

11 May '10

This fixes some whitespace/indentation flaws I stumbled over. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> --- include/sound/asound.h | 20 ++++++++++---------- 1 files changed, 10 insertions(+), 10 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 0985955..9f1eecf 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -574,7 +574,7 @@ enum { #define SNDRV_TIMER_FLG_SLAVE (1<<0) /* cannot be controlled */ struct snd_timer_id { - int dev_class; + int dev_class; int dev_sclass; int card; int device; @@ -762,7 +762,7 @@ struct snd_ctl_elem_id { snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ - unsigned char name[44]; /* ASCII name of item */ + unsigned char name[44]; /* ASCII name of item */ unsigned int index; /* index of item */ }; @@ -809,7 +809,7 @@ struct snd_ctl_elem_info { struct snd_ctl_elem_value { struct snd_ctl_elem_id id; /* W: element ID */ unsigned int indirect: 1; /* W: indirect access - obsoleted */ - union { + union { union { long value[128]; long *value_ptr; /* obsoleted */ @@ -827,15 +827,15 @@ struct snd_ctl_elem_value { unsigned char *data_ptr; /* obsoleted */ } bytes; struct snd_aes_iec958 iec958; - } value; /* RO */ + } value; /* RO */ struct timespec tstamp; - unsigned char reserved[128-sizeof(struct timespec)]; + unsigned char reserved[128-sizeof(struct timespec)]; }; struct snd_ctl_tlv { - unsigned int numid; /* control element numeric identification */ - unsigned int length; /* in bytes aligned to 4 */ - unsigned int tlv[0]; /* first TLV */ + unsigned int numid; /* control element numeric identification */ + unsigned int length; /* in bytes aligned to 4 */ + unsigned int tlv[0]; /* first TLV */ }; #define SNDRV_CTL_IOCTL_PVERSION _IOR('U', 0x00, int) @@ -886,8 +886,8 @@ struct snd_ctl_event { unsigned int mask; struct snd_ctl_elem_id id; } elem; - unsigned char data8[60]; - } data; + unsigned char data8[60]; + } data; }; /* -- 1.7.1

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[alsa-devel] MUTE broken on hda nvhdmi
by VDR User 11 May '10

11 May '10

Hi. Mute seems to be broken since the nvhdmi rewrite. I'm currently using alsa-driver-1.0.23.0.g30ba4.0.g0c8e7.tar.bz2 -- anyone got plans to fix this on their TODO list or could please look into it? Thanks.

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Re: [alsa-devel] USB transfer_buffer allocations on 64bit systems
by Alan Stern 11 May '10

11 May '10

On Tue, 11 May 2010, FUJITA Tomonori wrote: > > > > > At one point we tried an experiment, printing out the buffer and DMA > > > > > addresses. I don't recall seeing anything obviously wrong, but if an > > > > > IOMMU was in use then that might not mean anything. Is it possible > > > > > that the IOMMU mappings sometimes get messed up for addresses above 4 > > > > > GB? > > > > > > > > You mean that an IOMMU could allocate an address above 4GB wrongly? If > > > > so, IIRC, all the IOMMU implementations use dev->dma_mask and > > > > dev->coherent_dma_mask properly. And the DMA address space of the > > > > majority of IOMMUs are limited less than 4GB. > > > > > > The Intel IOMMU code will use dev->dma_mask and dev->coherent_dma_mask > > > properly. It is not limited to 4GiB, but it will tend to give virtual > > > DMA addresses below 4GiB even when a device is capable of more; it'll > > > only give out higher addresses when the address space below 4GiB is > > > exhausted. > > > > What I meant was: Is it possible that the IOMMU code will return a > > virtual DMA address before 4 GB but will somehow forget to actually map > > that address to the data buffer? > > Then, the IOMMU is completely broken. Then, we would get tons of DMA > bugs not only about USB, I guess. So I'm not sure. Yes, you're right about that. > > The problem goes away when Pedro boots with mem=4G. And the dma_mask > > value is set properly (in fact, the ehci-hcd driver currently doesn't > > use 64-bit DMA at all). > > > > If anyone wants to see the debug log entries showing the buffer and DMA > > addresses, they are attached to this email message: > > > > http://marc.info/?l=linux-kernel&m=127076841801054&w=2 > > > > Either the data isn't getting written to the buffer correctly or else > > the buffer isn't getting sent to the device correctly. Can anybody > > suggest a means of determining which is the case? > > I can't say anything about this log that including only DMA addresses. > I'm not familiar with how the USB core does DMA stuff. And the USB > stack design that the USB core does DMA stuff (allocating, mappings, > etc) makes debugging DMA issues really difficult. The DMA stuff is simple enough in this case. The urb->transfer_buffer address is passed to dma_map_single(), and the DMA address it returns is stored in urb->transfer_dma. Those are the two values printed out by the debugging patch. Alan Stern

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 11 May '10

11 May '10

The following changes since commit 896060c76bdfd8a45eb33b3dd1a8307fe37f6c04: Mark Brown (1): ASoC: Use more idiomatic driver name for WM8731 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 A bunch of minor WM8993/4 enhancements and Peter's invert handling. Mark Brown (4): ASoC: Add register write logging for WM8994 ASoC: Add additional WM hubs DC servo trace ASoC: Reorder power down sequence for WM hubs devices ASoC: Don't restart unconfigured WM8994 FLLs Peter Ujfalusi (1): ASoC: core: Fix for the volume limiting when invert is in use include/sound/soc.h | 23 ++++++++++++----------- sound/soc/codecs/wm8994.c | 5 +++++ sound/soc/codecs/wm_hubs.c | 12 ++++++++---- sound/soc/soc-core.c | 30 +++++++++++++++++++++--------- 4 files changed, 46 insertions(+), 24 deletions(-)

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Re: [alsa-devel] WM8993: amp control, amplifying pops.
by Mark Brown 11 May '10

11 May '10

On Tue, May 11, 2010 at 11:07:38AM +0800, Zhang JieJing wrote: > 2010/5/10 Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Hi, Our kernel version is 2.6.28, but port the wm8993 driver to this > version's kernel. Normally in these situations I would recommend backporting the entire ASoC subsystem - it's fairly well isolated from the rest of the kernel so this tends to be reasonably straightforward. > the amplifying pop is when system resume, the speaker will hear one > short pop noise, > also, this short pop noise will hear the first time driver is opened. When exactly do you hear this noise? The WM8993 driver will not power up the speaker outputs unless they are in use and there are no audio paths configured to the speakers by default. Are you configuring an audio path in the CODEC at system startup? Which mode do you have the WM8993 speaker drivers in - class D or class A/B? It might be helpful to capture all the register writes done when bringing up the WM8993. Also, if you are configuring an audio path from an analogue input you should check that there is no DC offset present in the input signal - if there is one then the WM8993 will reproduce it into the output path, which would result in an audible noise. > In 53242c68333570631a15a69842851b458eca3d99, > there was a line: > > + ret = wm8993_set_fll(codec->dai, 0, wm8993->fll_src, > + fll_fref, fll_fout); > + > but in lastest version of wm8993 driver, wm8993_set_fll will do > nothing we the second parameter(fll_id) is 0, so, I don't know what > this function want to do, > Cound you give me some idea ? This parameter specifies which of the FLLs/PLLs in the device to configure. Since the WM8993 has only one FLL it just ignores this parameter, other devices with multiple FLLs such as the WM8994 need this to select between the FLLs.

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[alsa-devel] snd-hda-intel - wallclk patch
by Jaroslav Kysela 11 May '10

11 May '10

Hi, I would like to ask HDA gurus to check this patch (I will include it to my tree once acked). The patch uses WALLCLK from HDA chips (marked as required in the HDA specification) to check for bogus - too early - interrups which confuses the upper PCM layer (sound skipping issues). More details about patch testing on problematic hardware can be found at: https://bugzilla.kernel.org/show_bug.cgi?id=15912 Thanks, Jaroslav diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 236b4ca..714978f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -174,7 +174,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #define ICH6_GSTS_FSTS (1 << 1) /* flush status */ #define ICH6_REG_INTCTL 0x20 #define ICH6_REG_INTSTS 0x24 -#define ICH6_REG_WALCLK 0x30 +#define ICH6_REG_WALLCLK 0x30 /* 24Mhz source */ #define ICH6_REG_SYNC 0x34 #define ICH6_REG_CORBLBASE 0x40 #define ICH6_REG_CORBUBASE 0x44 @@ -340,8 +340,8 @@ struct azx_dev { unsigned int period_bytes; /* size of the period in bytes */ unsigned int frags; /* number for period in the play buffer */ unsigned int fifo_size; /* FIFO size */ - unsigned long start_jiffies; /* start + minimum jiffies */ - unsigned long min_jiffies; /* minimum jiffies before position is valid */ + unsigned long start_wallclk; /* start + minimum wallclk */ + unsigned long period_wallclk; /* wallclk for period */ void __iomem *sd_addr; /* stream descriptor pointer */ @@ -361,7 +361,6 @@ struct azx_dev { unsigned int opened :1; unsigned int running :1; unsigned int irq_pending :1; - unsigned int start_flag: 1; /* stream full start flag */ /* * For VIA: * A flag to ensure DMA position is 0 @@ -1674,8 +1673,9 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) return err; } - azx_dev->min_jiffies = (runtime->period_size * HZ) / - (runtime->rate * 2); + /* wallclk has 24Mhz clock source */ + azx_dev->period_wallclk = (((runtime->period_size * 24000) / + runtime->rate) * 1000); azx_setup_controller(chip, azx_dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1; @@ -1729,14 +1729,15 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) if (s->pcm->card != substream->pcm->card) continue; azx_dev = get_azx_dev(s); - if (rstart) { - azx_dev->start_flag = 1; - azx_dev->start_jiffies = jiffies + azx_dev->min_jiffies; - } - if (start) + if (start) { + azx_dev->start_wallclk = azx_readl(chip, WALLCLK); + if (!rstart) + azx_dev->start_wallclk -= + azx_dev->period_wallclk; azx_stream_start(chip, azx_dev); - else + } else { azx_stream_stop(chip, azx_dev); + } azx_dev->running = start; } spin_unlock(&chip->reg_lock); @@ -1880,12 +1881,13 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) */ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) { + u32 wallclk; unsigned int pos; - if (azx_dev->start_flag && - time_before_eq(jiffies, azx_dev->start_jiffies)) + wallclk = azx_readl(chip, WALLCLK); + if ((wallclk - azx_dev->start_wallclk) < + (azx_dev->period_wallclk * 2) / 3) return -1; /* bogus (too early) interrupt */ - azx_dev->start_flag = 0; pos = azx_get_position(chip, azx_dev); if (chip->position_fix == POS_FIX_AUTO) { @@ -1899,13 +1901,12 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) chip->position_fix = POS_FIX_POSBUF; } - if (!bdl_pos_adj[chip->dev_index]) - return 1; /* no delayed ack */ if (WARN_ONCE(!azx_dev->period_bytes, "hda-intel: zero azx_dev->period_bytes")) return 0; /* this shouldn't happen! */ if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) return 0; /* NG - it's below the period boundary */ + azx_dev->start_wallclk = wallclk; return 1; /* OK, it's fine */ } @@ -1915,7 +1916,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) static void azx_irq_pending_work(struct work_struct *work) { struct azx *chip = container_of(work, struct azx, irq_pending_work); - int i, pending; + int i, pending, ok; if (!chip->irq_pending_warned) { printk(KERN_WARNING @@ -1934,11 +1935,14 @@ static void azx_irq_pending_work(struct work_struct *work) !azx_dev->substream || !azx_dev->running) continue; - if (azx_position_ok(chip, azx_dev)) { + ok = azx_position_ok(chip, azx_dev); + if (ok > 0) { azx_dev->irq_pending = 0; spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(azx_dev->substream); spin_lock(&chip->reg_lock); + } else if (ok < 0) { + pending = 0; /* too early */ } else pending++; } ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH] ASoC: core: Fix for the volume limiting when invert is in use
by Peter Ujfalusi 11 May '10

11 May '10

If the register for the volume needs invert, than the inversion need to be done from the chip maximum, and not from the platform dependent limit. Introduce soc_mixer_control.platform_max value, which initially equals to chip maximum. The snd_soc_limit_volume function only modify the platform_max, all volsw_info call returns this as well. The .max value holds the chip default (maximum), and it is used for the inversion, if it is needed. Additional check in the volsw_info call has been added to check the validity of the platform_max in case, when custom macros used by codec drivers are not initializing it correctly. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello, Normal, and inverted volume controls now can be limited, and both behave correctly. Has been tested on twl4030, tpa6130a2, and tlv320dac33 codec drivers. The patch is generated against Takashi's topic/asoc branch, but I think it should be fine with Mark's tree as well. Peter include/sound/soc.h | 23 ++++++++++++----------- sound/soc/soc-core.c | 30 +++++++++++++++++++++--------- 2 files changed, 33 insertions(+), 20 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9f306f0..aac97d7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -30,10 +30,10 @@ #define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = xshift, .rshift = xshift, .max = xmax, \ - .invert = xinvert}) + .platform_max = xmax, .invert = xinvert}) #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .max = xmax, .invert = xinvert}) + {.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert}) #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ @@ -53,14 +53,14 @@ .put = snd_soc_put_volsw, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -70,7 +70,7 @@ .put = snd_soc_put_volsw, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = shift_left, .rshift = shift_right,\ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -80,7 +80,7 @@ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ @@ -89,7 +89,8 @@ .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ .put = snd_soc_put_volsw_s8, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .min = xmin, .max = xmax} } + {.reg = xreg, .min = xmin, .max = xmax, \ + .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .max = xmax, .texts = xtexts } @@ -126,7 +127,7 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -146,7 +147,7 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -157,7 +158,7 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ @@ -569,7 +570,7 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { - int min, max; + int min, max, platform_max; unsigned int reg, rreg, shift, rshift, invert; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4079223..9340dcd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1954,18 +1954,22 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; + int platform_max; unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = shift == rshift ? 1 : 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = max; + uinfo->value.integer.max = platform_max; return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw); @@ -2063,16 +2067,20 @@ int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; + int platform_max; - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = max; + uinfo->value.integer.max = platform_max; return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); @@ -2173,13 +2181,17 @@ int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; + int platform_max; int min = mc->min; + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = max-min; + uinfo->value.integer.max = platform_max - min; return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); @@ -2268,7 +2280,7 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, if (found) { mc = (struct soc_mixer_control *)kctl->private_value; if (max <= mc->max) { - mc->max = max; + mc->platform_max = max; ret = 0; } } -- 1.7.1

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 11 May '10

11 May '10

The following changes since commit d0bbc24d2ab78d9efc61bc80cfc85f59626ba5af: Jassi Brar (1): ASoC: SMDK64XX: Switch to IISv4 CPU driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Mark Brown (8): ASoC: Remove unneeded suspend bias managment from CODEC drivers ASoC: Remove unused DAPM suspend flag ASoC: Refactor DAPM suspend handling ASoC: Support leaving paths enabled over system suspend ASoC: Allow active paths from the GSM modem while the GTA02 is suspended ASoC: Allow DAI links to be kept active over suspend ASoC: Refactor WM8731 regulator management into bias management ASoC: Use more idiomatic driver name for WM8731 drivers/mfd/wm831x-irq.c | 30 ++++++++++ include/sound/soc-dapm.h | 3 +- include/sound/soc.h | 3 + sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/stac9766.c | 3 - sound/soc/codecs/tlv320aic23.c | 1 - sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 3 - sound/soc/codecs/twl4030.c | 1 - sound/soc/codecs/twl6040.c | 1 - sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm8350.c | 3 - sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 54 +++++++++-------- sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm9712.c | 3 - sound/soc/codecs/wm9713.c | 3 - sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 8 +++ sound/soc/soc-core.c | 73 ++++++++++++++++++++++-- sound/soc/soc-dapm.c | 92 +++++++++++++++++++----------- 27 files changed, 206 insertions(+), 94 deletions(-)

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[alsa-devel] [PATCH] hda_intel: Fixes distorted recording on US15W chipset
by Shahin Ghazinouri 11 May '10

11 May '10

The HDA controller in US15W (Poulsbo) reports inaccurate position values for capture streams when using the LPIB read method, resulting in distorted recordings. However, using the position buffer is broken for playback streams, resulting in a fallback to the LPIB method with the current driver. This patch works around the issue by independently detecting the read position method for capture and playback streams. The patch will not have any effect if the position fix method is explicitly set. Signed-off-by: Shahin Ghazinouri <shahin.ghazinouri(a)pelagicore.com> --- --- linux-2.6.34-rc6/sound/pci/hda/hda_intel.c 2010-04-30 05:02:05.000000000 +0200 +++ linux-2.6.34-rc6_patch/sound/pci/hda/hda_intel.c 2010-04-30 11:31:57.000000000 +0200 @@ -425,7 +425,8 @@ struct snd_dma_buffer posbuf; /* flags */ - int position_fix; + int playback_position_fix; + int capture_position_fix; int poll_count; unsigned int running :1; unsigned int initialized :1; @@ -1302,8 +1303,10 @@ azx_sd_writel(azx_dev, SD_BDLPU, upper_32_bits(azx_dev->bdl.addr)); /* enable the position buffer */ - if (chip->position_fix == POS_FIX_POSBUF || - chip->position_fix == POS_FIX_AUTO || + if (chip->capture_position_fix == POS_FIX_POSBUF || + chip->capture_position_fix == POS_FIX_AUTO || + chip->playback_position_fix == POS_FIX_POSBUF || + chip->playback_position_fix == POS_FIX_AUTO || chip->via_dmapos_patch) { if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE)) azx_writel(chip, DPLBASE, @@ -1843,13 +1846,24 @@ if (chip->via_dmapos_patch) pos = azx_via_get_position(chip, azx_dev); - else if (chip->position_fix == POS_FIX_POSBUF || - chip->position_fix == POS_FIX_AUTO) { - /* use the position buffer */ - pos = le32_to_cpu(*azx_dev->posbuf); + else if (azx_dev->index < chip->playback_index_offset) { + /* Capture stream */ + if (chip->capture_position_fix == POS_FIX_POSBUF || + chip->capture_position_fix == POS_FIX_AUTO) + /* use the position buffer */ + pos = le32_to_cpu(*azx_dev->posbuf); + else + /* read LPIB */ + pos = azx_sd_readl(azx_dev, SD_LPIB); } else { - /* read LPIB */ - pos = azx_sd_readl(azx_dev, SD_LPIB); + /* Playback stream */ + if (chip->playback_position_fix == POS_FIX_POSBUF || + chip->playback_position_fix == POS_FIX_AUTO) + /* use the position buffer */ + pos = le32_to_cpu(*azx_dev->posbuf); + else + /* read LPIB */ + pos = azx_sd_readl(azx_dev, SD_LPIB); } if (pos >= azx_dev->bufsize) pos = 0; @@ -1884,15 +1898,30 @@ azx_dev->start_flag = 0; pos = azx_get_position(chip, azx_dev); - if (chip->position_fix == POS_FIX_AUTO) { - if (!pos) { - printk(KERN_WARNING - "hda-intel: Invalid position buffer, " - "using LPIB read method instead.\n"); - chip->position_fix = POS_FIX_LPIB; - pos = azx_get_position(chip, azx_dev); - } else - chip->position_fix = POS_FIX_POSBUF; + if (azx_dev->index < chip->playback_index_offset) { + /* Capture stream */ + if (chip->capture_position_fix == POS_FIX_AUTO) { + if (!pos) { + printk(KERN_WARNING + "hda-intel: Invalid position buffer, " + "using LPIB read method instead.\n"); + chip->capture_position_fix = POS_FIX_LPIB; + pos = azx_get_position(chip, azx_dev); + } else + chip->capture_position_fix = POS_FIX_POSBUF; + } + } else { + /* Playback stream */ + if (chip->playback_position_fix == POS_FIX_AUTO) { + if (!pos) { + printk(KERN_WARNING + "hda-intel: Invalid position buffer, " + "using LPIB read method instead.\n"); + chip->playback_position_fix = POS_FIX_LPIB; + pos = azx_get_position(chip, azx_dev); + } else + chip->playback_position_fix = POS_FIX_POSBUF; + } } if (!bdl_pos_adj[chip->dev_index]) @@ -2431,7 +2460,10 @@ chip->dev_index = dev; INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); - chip->position_fix = check_position_fix(chip, position_fix[dev]); + chip->capture_position_fix = + check_position_fix(chip, position_fix[dev]); + chip->playback_position_fix = + check_position_fix(chip, position_fix[dev]); check_probe_mask(chip, dev); chip->single_cmd = single_cmd;

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[alsa-devel] [PATCH] ALSA: hda: Fix 0 dB for Lenovo models using Conexant CX20549 (Venice)
by Daniel Chen 11 May '10

11 May '10

Hi, Attached please find a patch against sound-2.6 (also applicable to stable trees) that fixes the reported distortion for Lenovo models on CX20549. Best, -Dan

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[alsa-devel] [PATCH] ALSA: hda: Fix 0 dB for Packard Bell models using Conexant CX20549 (Venice)
by Daniel Chen 11 May '10

11 May '10

Hi, Attached is a patch against sound-2.6 master HEAD (also applicable to 2.6.32.12 and 2.6.33.3) that fixes 0 dB for Packard Bell machines using Conexant CX20549. Without it, sound distorts horribly. Best, -Dan

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[alsa-devel] Mute LED on HP Pavilion DV7-1160eg
by Thoralf Freitag 11 May '10

11 May '10

I tried to find a way for switching/control Mute LED on an HP DV 7 notebook. I found a solution which is very ugly. According the solution described here (http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016126.html) I changed the value for LED control in this manner: diff -u alsa-driver-1.0.23/sound/pci/hda/patch_sigmatel.c alsa-driver-1.0.23_b/sound/pci/hda/patch_sigmatel.c --- alsa-driver-1.0.23/sound/pci/hda/patch_sigmatel.c 2010-04-16 13:10:10.000000000 +0200 +++ alsa-driver-1.0.23_b/sound/pci/hda/patch_sigmatel.c 2010-05-01 21:37:36.719063655 +0200 @@ -4952,9 +4952,9 @@ } } if (muted) - spec->gpio_data &= ~spec->gpio_led; /* orange */ + spec->gpio_data &= ~0x01; /* orange */ else - spec->gpio_data |= spec->gpio_led; /* white */ + spec->gpio_data |= 0x01; /* white */ if (!spec->gpio_led_polarity) { /* LED state is inverted on these systems */ I hate this solution. Maybe someone of the programming experts could give me a way to make it better. On which position should I make which changes ? Thanx for your guidance. Thoralf

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[alsa-devel] [RFC] G.723-24 format patch
by Ben Collins 10 May '10

10 May '10

I'm writing a driver for a card that only has hardware support for G.723-24 ADPCM format. I am working around the lack of support in alsa at the moment for this format, but wanted to take a crack at implementing it properly. G.723-24 is a 3-bit signed sample at a fixed 8khz mono sample rate (24000 bit rate). Basically means you need to take in at least 3 bytes (8 samples) minimum for processing the standard packed version. I don't have hardware that supports it, but since I was in the code, I added a G723_24_1B format for producing 1 sample per byte (3 lower bits are actual sample), but for my use I am producing the standard packed stream. Only question I had is that since this is signed, should I expect or want the _1B version to be a 1 byte sign extended version of the 3-bit sample, or just expect the lower 3 bits of the byte to be the g723-24 sample? Looking at other unpacked examples, I assume the latter rather than the former. diff --git a/include/sound/asound.h b/include/sound/asound.h index 1f57bb9..16fc633 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -212,7 +212,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_S18_3BE ((__force snd_pcm_format_t) 41) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3LE ((__force snd_pcm_format_t) 42) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3BE ((__force snd_pcm_format_t) 43) /* in three bytes */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_U18_3BE +#define SNDRV_PCM_FORMAT_G723_24 ((__force snd_pcm_format_t) 44) /* 8 samples in 3 bytes */ +#define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_24_1B #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/include/sound/pcm.h b/include/sound/pcm.h index de6d981..031e2cd 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -173,6 +173,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_U18_3LE (1ULL << SNDRV_PCM_FORMAT_U18_3LE) #define SNDRV_PCM_FMTBIT_S18_3BE (1ULL << SNDRV_PCM_FORMAT_S18_3BE) #define SNDRV_PCM_FMTBIT_U18_3BE (1ULL << SNDRV_PCM_FORMAT_U18_3BE) +#define SNDRV_PCM_FMTBIT_G723_24 (1ULL << SNDRV_PCM_FORMAT_G723_24) +#define SNDRV_PCM_FMTBIT_G723_24_1B (1ULL << SNDRV_PCM_FORMAT_G723_24_1B) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index ea2bf82..c7f8da4 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -128,6 +128,10 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { .width = 4, .phys = 4, .le = -1, .signd = -1, .silence = {}, }, + [SNDRV_PCM_FORMAT_G723_24] = { + .width = 3, .phys = 3, .le = -1, .signd = 1, + .silence = {}, + }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, @@ -186,6 +190,10 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { .width = 18, .phys = 24, .le = 0, .signd = 0, .silence = { 0x02, 0x00, 0x00 }, }, + [SNDRV_PCM_FORMAT_G723_24_1B] = { + .width = 3, .phys = 8, .le = -1, .signd = 1, + .silence = {}, + }, };

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Re: [alsa-devel] ALSA Control Interface - default values?
by Mark Brown 10 May '10

10 May '10

On Mon, May 10, 2010 at 04:25:07PM +0200, Max Klein wrote: > thanks for fast reply, may you tell me how the driver can set the defaults? > EDIT: whoops, I guess it was an error in reasoning from myself...as I have the control of all those variables outputted by the get and put callbacks, I just have to initialize those variables before initializing the control interface, right? Yes, the driver just needs to make sure that whatever it uses to return the current setting when queried is going to give the right answer from the time the control is set up, normally by initialising variables though if you're doing things like read data back from hardware then you may be able to rely on the hardware defaults. [Re-adding CC to alsa-devel - it's normal to do a reply to all for Linux lists.]

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Re: [alsa-devel] [PATCH] ALSA: hda: enable SPDIF output for Conexant 5051/Lenovo docking stations
by Takashi Iwai 10 May '10

10 May '10

At Mon, 10 May 2010 10:21:50 -0400, Aristeu Rozanski wrote: > > Hi Pierre, > > this patch should work for all CXT5051-based Lenovo docking stations. > > Aristeu, you added a quirk for the X200 docking station (commit > > 0e6e6db212b0c97c5d4f5f55b674ec2d3276c16c), > > can you check if you see any adverse effects with this patch? > the patch doesn't change or break anything on X200 or X200 + docking Thanks for checking! Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda: enable SPDIF output for Conexant 5051/Lenovo docking stations
by pl bossart 10 May '10

10 May '10

this patch should work for all CXT5051-based Lenovo docking stations. Aristeu, you added a quirk for the X200 docking station (commit 0e6e6db212b0c97c5d4f5f55b674ec2d3276c16c), can you check if you see any adverse effects with this patch? Thanks - Pierre

2 3

[alsa-devel] ALSA Control Interface - default values?
by Max Klein 10 May '10

10 May '10

Hello, when writing an alsa driver, in the info callback of the control I can specify which range of values this control does accept (uinfo->value.integer.min and .max). If I put 1 as minimum and 10 as maximum for example, I would expect that the default value of this control is something in between the range, e.g. 1, after the driver is loaded. But I figured out, that ALSA initializes the control with a value of 0, which is not in between the range. Can someone please tell me why ALSA initializes the Control with a value of zero, even if the minimum in the info callback is 1 ? Is there any way to initialize the control with some "default" values, aside of snd_ctl_add()? Best regards and thanks for your help! Max -- "Feel free" - 10 GB Mailbox, 100 FreeSMS/Monat ... Jetzt GMX TopMail testen: http://www.gmx.net/de/go/topmail

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Re: [alsa-devel] WM8993: amp control, amplifying pops.
by Mark Brown 10 May '10

10 May '10

On Mon, May 10, 2010 at 10:10:53AM +0800, Zhang JieJing wrote: > I have a question about WM8993 codec's amplifying pops control. Could you please be more specific about what you mean when you say "amplifying pops control"? What exactly is an "amplifying pop" (do you mean that you hear an unwanted noise, or is there something else)? Please ensure that you are using the latest version of the WM8993 driver (including wm_hubs.c) to make sure that you have all the latest bug fixes and enhancements. If you are not using single ended line outputs from the device you should also mark the line outputs as being in differential mode for maximum power savings. > Our board using a WM8993 codec, and I We have problem about ampifying pops, > when the system resume, We will heard a ampifying pop, (but it's silence on > the suspend processing), We also noticed that when system boot, also can > hear one ampifying pop. System resume and power up are identical from the point of view of the CODEC driver. > From I know, the WM8993's codec have a internal amplifier, I don't know how > to control this amplifier. The WM8993 has a large number of amplifiers, there are many audio paths in the device and most of them have some amplification control. Which outputs are you concerned about?

1 0

[alsa-devel] [PATCH] ASoC: core: Support for limiting the volume
by Peter Ujfalusi 10 May '10

10 May '10

Add support for the core to limit the maximum volume on an existing control. The function will modify the soc_mixer_control.max value of the given control. The new value must be lower than the original one (chip maximum) If there is a need for limiting a gain on a given control, than machine drivers can do the following in their snd_soc_dai_link.init function: snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21); This will modify the original 31 (chip maximum) to 21, so user space will not be able to set the gain higher than this. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 39 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 41 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 01e9c66..9f306f0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -326,6 +326,8 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_limit_volume(struct snd_soc_codec *codec, + const char *name, int max); /** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d59076e..4079223 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2238,6 +2238,45 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); /** + * snd_soc_limit_volume - Set new limit to an existing volume control. + * + * @codec: where to look for the control + * @name: Name of the control + * @max: new maximum limit + * + * Return 0 for success, else error. + */ +int snd_soc_limit_volume(struct snd_soc_codec *codec, + const char *name, int max) +{ + struct snd_card *card = codec->card; + struct snd_kcontrol *kctl; + struct soc_mixer_control *mc; + int found = 0; + int ret = -EINVAL; + + /* Sanity check for name and max */ + if (unlikely(!name || max <= 0)) + return -EINVAL; + + list_for_each_entry(kctl, &card->controls, list) { + if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { + found = 1; + break; + } + } + if (found) { + mc = (struct soc_mixer_control *)kctl->private_value; + if (max <= mc->max) { + mc->max = max; + ret = 0; + } + } + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_limit_volume); + +/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID -- 1.7.1

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Re: [alsa-devel] USB transfer_buffer allocations on 64bit systems
by David Woodhouse 10 May '10

10 May '10

On Mon, 2010-05-10 at 11:50 +0900, FUJITA Tomonori wrote: > On Fri, 7 May 2010 10:51:10 -0400 (EDT) > Alan Stern <stern(a)rowland.harvard.edu> wrote: > > > On Fri, 7 May 2010, Daniel Mack wrote: > > > > > > At least the audio class and ua101 drivers don't do this and fill the > > > > buffers before they are submitted. > > > > > > Gnaa, you're right. I _thought_ my code does it the way I described, but > > > what I wrote is how I _wanted_ to do it, not how it's currently done. I > > > have a plan to change this in the future. > > > > > > So unfortunately, that doesn't explain it either. Sorry for the noise. > > > > At one point we tried an experiment, printing out the buffer and DMA > > addresses. I don't recall seeing anything obviously wrong, but if an > > IOMMU was in use then that might not mean anything. Is it possible > > that the IOMMU mappings sometimes get messed up for addresses above 4 > > GB? > > You mean that an IOMMU could allocate an address above 4GB wrongly? If > so, IIRC, all the IOMMU implementations use dev->dma_mask and > dev->coherent_dma_mask properly. And the DMA address space of the > majority of IOMMUs are limited less than 4GB. The Intel IOMMU code will use dev->dma_mask and dev->coherent_dma_mask properly. It is not limited to 4GiB, but it will tend to give virtual DMA addresses below 4GiB even when a device is capable of more; it'll only give out higher addresses when the address space below 4GiB is exhausted. -- dwmw2

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[alsa-devel] [PATCH 1/2] es1688: allocate snd_es1688 structure as a part of snd_card structure
by Krzysztof Helt 10 May '10

10 May '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Allocate the snd_es1688 during the snd_card allocation. This allows to remove the card pointer from the snd_es1688 structure. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the second version. It removes kfree(chip) as well. >> Allocate the snd_es1688 during the snd_card allocation. >> This allows to remove the card pointer from the snd_es1688 structure. > Any other merit? There is no real purpose of this change aside saving one allocation, removing one pointer from the structure and making code shorter. include/sound/es1688.h | 10 ++++---- sound/isa/es1688/es1688.c | 23 ++++++++++++--------- sound/isa/es1688/es1688_lib.c | 44 +++++++++++++--------------------------- sound/isa/gus/gusextreme.c | 26 +++++++++++++----------- 4 files changed, 46 insertions(+), 57 deletions(-) diff --git a/include/sound/es1688.h b/include/sound/es1688.h index 10fcf14..4c29572 100644 --- a/include/sound/es1688.h +++ b/include/sound/es1688.h @@ -44,7 +44,6 @@ struct snd_es1688 { unsigned char pad; unsigned int dma_size; - struct snd_card *card; struct snd_pcm *pcm; struct snd_pcm_substream *playback_substream; struct snd_pcm_substream *capture_substream; @@ -108,14 +107,15 @@ struct snd_es1688 { void snd_es1688_mixer_write(struct snd_es1688 *chip, unsigned char reg, unsigned char data); int snd_es1688_create(struct snd_card *card, + struct snd_es1688 *chip, unsigned long port, unsigned long mpu_port, int irq, int mpu_irq, int dma8, - unsigned short hardware, - struct snd_es1688 ** rchip); -int snd_es1688_pcm(struct snd_es1688 *chip, int device, struct snd_pcm ** rpcm); -int snd_es1688_mixer(struct snd_es1688 *chip); + unsigned short hardware); +int snd_es1688_pcm(struct snd_card *card, struct snd_es1688 *chip, int device, + struct snd_pcm **rpcm); +int snd_es1688_mixer(struct snd_card *card, struct snd_es1688 *chip); #endif /* __SOUND_ES1688_H */ diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 07df201..2816794 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -79,8 +79,8 @@ static int __devinit snd_es1688_match(struct device *dev, unsigned int n) return enable[n]; } -static int __devinit snd_es1688_legacy_create(struct snd_card *card, - struct device *dev, unsigned int n, struct snd_es1688 **rchip) +static int __devinit snd_es1688_legacy_create(struct snd_card *card, + struct snd_es1688 *chip, struct device *dev, unsigned int n) { static long possible_ports[] = {0x220, 0x240, 0x260}; static int possible_irqs[] = {5, 9, 10, 7, -1}; @@ -104,14 +104,14 @@ static int __devinit snd_es1688_legacy_create(struct snd_card *card, } if (port[n] != SNDRV_AUTO_PORT) - return snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip); + return snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_AUTO); i = 0; do { port[n] = possible_ports[i]; - error = snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip); + error = snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_AUTO); } while (error < 0 && ++i < ARRAY_SIZE(possible_ports)); return error; @@ -125,19 +125,22 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) struct snd_pcm *pcm; int error; - error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + error = snd_card_create(index[n], id[n], THIS_MODULE, + sizeof(struct snd_es1688), &card); if (error < 0) return error; - error = snd_es1688_legacy_create(card, dev, n, &chip); + chip = card->private_data; + + error = snd_es1688_legacy_create(card, chip, dev, n); if (error < 0) goto out; - error = snd_es1688_pcm(chip, 0, &pcm); + error = snd_es1688_pcm(card, chip, 0, &pcm); if (error < 0) goto out; - error = snd_es1688_mixer(chip); + error = snd_es1688_mixer(card, chip); if (error < 0) goto out; diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index c76bb00..fdd4404 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -620,7 +620,6 @@ static int snd_es1688_free(struct snd_es1688 *chip) disable_dma(chip->dma8); free_dma(chip->dma8); } - kfree(chip); return 0; } @@ -638,23 +637,20 @@ static const char *snd_es1688_chip_id(struct snd_es1688 *chip) } int snd_es1688_create(struct snd_card *card, + struct snd_es1688 *chip, unsigned long port, unsigned long mpu_port, int irq, int mpu_irq, int dma8, - unsigned short hardware, - struct snd_es1688 **rchip) + unsigned short hardware) { static struct snd_device_ops ops = { .dev_free = snd_es1688_dev_free, }; - struct snd_es1688 *chip; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (chip == NULL) return -ENOMEM; chip->irq = -1; @@ -662,25 +658,21 @@ int snd_es1688_create(struct snd_card *card, if ((chip->res_port = request_region(port + 4, 12, "ES1688")) == NULL) { snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4); - snd_es1688_free(chip); return -EBUSY; } if (request_irq(irq, snd_es1688_interrupt, IRQF_DISABLED, "ES1688", (void *) chip)) { snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq); - snd_es1688_free(chip); return -EBUSY; } chip->irq = irq; if (request_dma(dma8, "ES1688")) { snd_printk(KERN_ERR "es1688: can't grab DMA8 %d\n", dma8); - snd_es1688_free(chip); return -EBUSY; } chip->dma8 = dma8; spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); - chip->card = card; chip->port = port; mpu_port &= ~0x000f; if (mpu_port < 0x300 || mpu_port > 0x330) @@ -689,23 +681,16 @@ int snd_es1688_create(struct snd_card *card, chip->mpu_irq = mpu_irq; chip->hardware = hardware; - if ((err = snd_es1688_probe(chip)) < 0) { - snd_es1688_free(chip); + err = snd_es1688_probe(chip); + if (err < 0) return err; - } - if ((err = snd_es1688_init(chip, 1)) < 0) { - snd_es1688_free(chip); - return err; - } - /* Register device */ - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_es1688_free(chip); + err = snd_es1688_init(chip, 1); + if (err < 0) return err; - } - *rchip = chip; - return 0; + /* Register device */ + return snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); } static struct snd_pcm_ops snd_es1688_playback_ops = { @@ -730,12 +715,14 @@ static struct snd_pcm_ops snd_es1688_capture_ops = { .pointer = snd_es1688_capture_pointer, }; -int snd_es1688_pcm(struct snd_es1688 * chip, int device, struct snd_pcm ** rpcm) +int snd_es1688_pcm(struct snd_card *card, struct snd_es1688 *chip, + int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "ESx688", device, 1, 1, &pcm)) < 0) + err = snd_pcm_new(card, "ESx688", device, 1, 1, &pcm); + if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_es1688_playback_ops); @@ -1009,18 +996,15 @@ static unsigned char snd_es1688_init_table[][2] = { { ES1688_REC_DEV, 0x17 } }; -int snd_es1688_mixer(struct snd_es1688 *chip) +int snd_es1688_mixer(struct snd_card *card, struct snd_es1688 *chip) { - struct snd_card *card; unsigned int idx; int err; unsigned char reg, val; - if (snd_BUG_ON(!chip || !chip->card)) + if (snd_BUG_ON(!chip || !card)) return -EINVAL; - card = chip->card; - strcpy(card->mixername, snd_es1688_chip_id(chip)); for (idx = 0; idx < ARRAY_SIZE(snd_es1688_controls); idx++) { diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 65e4b18..008e8e5 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -95,7 +95,7 @@ static int __devinit snd_gusextreme_match(struct device *dev, unsigned int n) } static int __devinit snd_gusextreme_es1688_create(struct snd_card *card, - struct device *dev, unsigned int n, struct snd_es1688 **rchip) + struct snd_es1688 *chip, struct device *dev, unsigned int n) { static long possible_ports[] = {0x220, 0x240, 0x260}; static int possible_irqs[] = {5, 9, 10, 7, -1}; @@ -119,14 +119,14 @@ static int __devinit snd_gusextreme_es1688_create(struct snd_card *card, } if (port[n] != SNDRV_AUTO_PORT) - return snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_1688, rchip); + return snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_1688); i = 0; do { port[n] = possible_ports[i]; - error = snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_1688, rchip); + error = snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_1688); } while (error < 0 && ++i < ARRAY_SIZE(possible_ports)); return error; @@ -206,9 +206,8 @@ static int __devinit snd_gusextreme_detect(struct snd_gus_card *gus, return 0; } -static int __devinit snd_gusextreme_mixer(struct snd_es1688 *chip) +static int __devinit snd_gusextreme_mixer(struct snd_card *card) { - struct snd_card *card = chip->card; struct snd_ctl_elem_id id1, id2; int error; @@ -241,17 +240,20 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) struct snd_opl3 *opl3; int error; - error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + error = snd_card_create(index[n], id[n], THIS_MODULE, + sizeof(struct snd_es1688), &card); if (error < 0) return error; + es1688 = card->private_data; + if (mpu_port[n] == SNDRV_AUTO_PORT) mpu_port[n] = 0; if (mpu_irq[n] > 15) mpu_irq[n] = -1; - error = snd_gusextreme_es1688_create(card, dev, n, &es1688); + error = snd_gusextreme_es1688_create(card, es1688, dev, n); if (error < 0) goto out; @@ -280,11 +282,11 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) } gus->codec_flag = 1; - error = snd_es1688_pcm(es1688, 0, NULL); + error = snd_es1688_pcm(card, es1688, 0, NULL); if (error < 0) goto out; - error = snd_es1688_mixer(es1688); + error = snd_es1688_mixer(card, es1688); if (error < 0) goto out; @@ -300,7 +302,7 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) if (error < 0) goto out; - error = snd_gusextreme_mixer(es1688); + error = snd_gusextreme_mixer(card); if (error < 0) goto out; -- 1.6.4 ---------------------------------------------------------------------- Sprawdź jak dać rodzinie poczucie bezpieczeństwa. http://clk.tradedoubler.com/click?p=191431&a=1694818&g=18733532

2 1

[alsa-devel] [PATCH 2.6.34-rc5 v2] HID: Prodikeys PC-MIDI HID Driver
by Don Prince 10 May '10

10 May '10

A specialised HID driver for the Creative Prodikeys PC-MIDI USB Keyboard. The Prodikeys PC-MIDI is a multifunction keyboard comprising a qwerty keyboard, multimedia keys and a touch sensitive musical keyboard. The specialised HID driver adds full support for the musical keyboard and extra multimedia keys which are not currently handled by the default HID driver. The specialised HID driver interfaces with Alsa, and presents the midi keyboard as a Rawmidi device. Sustain duration, octave shifting and the midi output channel can be read/written form userspace via sysfs. Signed-off-by: Don Prince <dhprince-devel(a)yahoo.co.uk> --- Applied fixes to original patch submission suggested by Clemens Ladisch. The keyboard can be viewed here http://www.prodikeys.com/products/Prodikeys_PC-MIDI/ The driver (older version) and a simple wxWidgets based UI application is available (in tar.gz format) at the Sourceforge project page http://sourceforge.net/projects/pc-midi-linux/ Apply patch with patch -p1 < <path to the driver patch> --- diff -X /opt/scripts/diff-exclude.txt -Npaur a/Documentation/ABI/testing/sysfs-driver-hid-prodikeys b/Documentation/ABI/testing/sysfs-driver-hid-prodikeys --- a/Documentation/ABI/testing/sysfs-driver-hid-prodikeys 1970-01-01 01:00:00.000000000 +0100 +++ b/Documentation/ABI/testing/sysfs-driver-hid-prodikeys 2010-05-09 16:14:33.948863321 +0100 @@ -0,0 +1,29 @@ +What: /sys/bus/hid/drivers/prodikeys/.../channel +Date: April 2010 +KernelVersion: 2.6.34 +Contact: Don Prince <dhprince.devel(a)yahoo.co.uk> +Description: + Allows control (via software) the midi channel to which + that the pc-midi keyboard will output.midi data. + Range: 0..15 + Type: Read/write +What: /sys/bus/hid/drivers/prodikeys/.../sustain +Date: April 2010 +KernelVersion: 2.6.34 +Contact: Don Prince <dhprince.devel(a)yahoo.co.uk> +Description: + Allows control (via software) the sustain duration of a + note held by the pc-midi driver. + 0 means sustain mode is disabled. + Range: 0..5000 (milliseconds) + Type: Read/write +What: /sys/bus/hid/drivers/prodikeys/.../octave +Date: April 2010 +KernelVersion: 2.6.34 +Contact: Don Prince <dhprince.devel(a)yahoo.co.uk> +Description: + Controls the octave shift modifier in the pc-midi driver. + The octave can be shifted via software up/down 2 octaves. + 0 means the no ocatve shift. + Range: -2..2 (minus 2 to plus 2) + Type: Read/Write diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/hid-core.c b/drivers/hid/hid-core.c --- a/drivers/hid/hid-core.c 2010-04-22 18:52:46.445384754 +0100 +++ b/drivers/hid/hid-core.c 2010-04-25 19:10:31.317248975 +0100 @@ -1297,6 +1297,7 @@ static const struct hid_device_id hid_bl { HID_USB_DEVICE(USB_VENDOR_ID_BELKIN, USB_DEVICE_ID_FLIP_KVM) }, { HID_USB_DEVICE(USB_VENDOR_ID_CHERRY, USB_DEVICE_ID_CHERRY_CYMOTION) }, { HID_USB_DEVICE(USB_VENDOR_ID_CHICONY, USB_DEVICE_ID_CHICONY_TACTICAL_PAD) }, + { HID_USB_DEVICE(USB_VENDOR_ID_CREATIVELABS, USB_DEVICE_ID_PRODIKEYS_PCMIDI) }, { HID_USB_DEVICE(USB_VENDOR_ID_CYPRESS, USB_DEVICE_ID_CYPRESS_BARCODE_1) }, { HID_USB_DEVICE(USB_VENDOR_ID_CYPRESS, USB_DEVICE_ID_CYPRESS_BARCODE_2) }, { HID_USB_DEVICE(USB_VENDOR_ID_CYPRESS, USB_DEVICE_ID_CYPRESS_BARCODE_3) }, diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/hid-ids.h b/drivers/hid/hid-ids.h --- a/drivers/hid/hid-ids.h 2010-04-22 18:52:46.455385039 +0100 +++ b/drivers/hid/hid-ids.h 2010-04-25 19:10:31.319248829 +0100 @@ -147,6 +147,9 @@ #define USB_DEVICE_ID_CODEMERCS_IOW_FIRST 0x1500 #define USB_DEVICE_ID_CODEMERCS_IOW_LAST 0x15ff +#define USB_VENDOR_ID_CREATIVELABS 0x041e +#define USB_DEVICE_ID_PRODIKEYS_PCMIDI 0x2801 + #define USB_VENDOR_ID_CYGNAL 0x10c4 #define USB_DEVICE_ID_CYGNAL_RADIO_SI470X 0x818a diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/hid-prodikeys.c b/drivers/hid/hid-prodikeys.c --- a/drivers/hid/hid-prodikeys.c 1970-01-01 01:00:00.000000000 +0100 +++ b/drivers/hid/hid-prodikeys.c 2010-05-09 16:20:42.569736755 +0100 @@ -0,0 +1,910 @@ +/* + * HID driver for the Prodikeys PC-MIDI Keyboard + * providing midi & extra multimedia keys functionality + * + * Copyright (c) 2009 Don Prince <dhprince.devel(a)yahoo.co.uk> + * + * Controls for Octave Shift Up/Down, Channel, and + * Sustain Duration available via sysfs. + * + */ + +/* + * 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/device.h> +#include <linux/module.h> +#include <linux/usb.h> +#include <linux/mutex.h> +#include <linux/hid.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> +#include "usbhid/usbhid.h" +#include "hid-ids.h" + + +#define pk_debug(format, arg...) \ + pr_debug("hid-prodikeys: " format "\n" , ## arg) +#define pk_error(format, arg...) \ + pr_err("hid-prodikeys: " format "\n" , ## arg) + +struct pcmidi_snd; + +struct pk_device { + unsigned long quirks; + + struct hid_device *hdev; + struct pcmidi_snd *pm; /* pcmidi device context */ +}; + +struct pcmidi_snd; + +struct pcmidi_sustain { + unsigned long in_use; + struct pcmidi_snd *pm; + struct timer_list timer; + unsigned char status; + unsigned char note; + unsigned char velocity; +}; + +#define PCMIDI_SUSTAINED_MAX 32 +struct pcmidi_snd { + struct pk_device *pk; + unsigned short ifnum; + struct hid_report *pcmidi_report6; + struct input_dev *input_ep82; + unsigned short midi_mode; + unsigned short midi_sustain_mode; + unsigned short midi_sustain; + unsigned short midi_channel; + short midi_octave; + struct pcmidi_sustain sustained_notes[PCMIDI_SUSTAINED_MAX]; + unsigned short fn_state; + unsigned short last_key[24]; + spinlock_t rawmidi_in_lock; + struct snd_card *card; + struct snd_rawmidi *rwmidi; + struct snd_rawmidi_substream *in_substream; + struct snd_rawmidi_substream *out_substream; + unsigned long in_triggered; + unsigned long out_active; +}; + +#define PK_QUIRK_NOGET 0x00010000 +#define PCMIDI_MIDDLE_C 60 +#define PCMIDI_CHANNEL_MIN 0 +#define PCMIDI_CHANNEL_MAX 15 +#define PCMIDI_OCTAVE_MIN (-2) +#define PCMIDI_OCTAVE_MAX 2 +#define PCMIDI_SUSTAIN_MIN 0 +#define PCMIDI_SUSTAIN_MAX 5000 + +static const char shortname[] = "PC-MIDI"; +static const char longname[] = "Prodikeys PC-MIDI Keyboard"; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +module_param_array(id, charp, NULL, 0444); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for the PC-MIDI virtual audio driver"); +MODULE_PARM_DESC(id, "ID string for the PC-MIDI virtual audio driver"); +MODULE_PARM_DESC(enable, "Enable for the PC-MIDI virtual audio driver"); + + +/* Output routine for the sysfs channel file */ +static ssize_t show_channel(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + dbg_hid("pcmidi sysfs read channel=%u\n", pk->pm->midi_channel); + + return sprintf(buf, "%u (min:%u, max:%u)\n", pk->pm->midi_channel, + PCMIDI_CHANNEL_MIN, PCMIDI_CHANNEL_MAX); +} + +/* Input routine for the sysfs channel file */ +static ssize_t store_channel(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + unsigned channel = 0; + + if (sscanf(buf, "%u", &channel) > 0 && channel <= PCMIDI_CHANNEL_MAX) { + dbg_hid("pcmidi sysfs write channel=%u\n", channel); + pk->pm->midi_channel = channel; + return strlen(buf); + } + return -EINVAL; +} + +static DEVICE_ATTR(channel, S_IRUGO | S_IWUGO, show_channel, + store_channel); + +static struct device_attribute *sysfs_device_attr_channel = { + &dev_attr_channel, + }; + +/* Output routine for the sysfs sustain file */ +static ssize_t show_sustain(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + dbg_hid("pcmidi sysfs read sustain=%u\n", pk->pm->midi_sustain); + + return sprintf(buf, "%u (off:%u, max:%u (ms))\n", pk->pm->midi_sustain, + PCMIDI_SUSTAIN_MIN, PCMIDI_SUSTAIN_MAX); +} + +/* Input routine for the sysfs sustain file */ +static ssize_t store_sustain(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + unsigned sustain = 0; + + if (sscanf(buf, "%u", &sustain) > 0 && sustain <= PCMIDI_SUSTAIN_MAX) { + dbg_hid("pcmidi sysfs write sustain=%u\n", sustain); + pk->pm->midi_sustain = sustain; + pk->pm->midi_sustain_mode = + (0 == sustain || !pk->pm->midi_mode) ? 0 : 1; + return strlen(buf); + } + return -EINVAL; +} + +static DEVICE_ATTR(sustain, S_IRUGO | S_IWUGO, show_sustain, + store_sustain); + +static struct device_attribute *sysfs_device_attr_sustain = { + &dev_attr_sustain, + }; + +/* Output routine for the sysfs octave file */ +static ssize_t show_octave(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + dbg_hid("pcmidi sysfs read octave=%d\n", pk->pm->midi_octave); + + return sprintf(buf, "%d (min:%d, max:%d)\n", pk->pm->midi_octave, + PCMIDI_OCTAVE_MIN, PCMIDI_OCTAVE_MAX); +} + +/* Input routine for the sysfs octave file */ +static ssize_t store_octave(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + int octave = 0; + + if (sscanf(buf, "%d", &octave) > 0 && + octave >= PCMIDI_OCTAVE_MIN && octave <= PCMIDI_OCTAVE_MAX) { + dbg_hid("pcmidi sysfs write octave=%d\n", octave); + pk->pm->midi_octave = octave; + return strlen(buf); + } + return -EINVAL; +} + +static DEVICE_ATTR(octave, S_IRUGO | S_IWUGO, show_octave, + store_octave); + +static struct device_attribute *sysfs_device_attr_octave = { + &dev_attr_octave, + }; + + +static void pcmidi_send_note(struct pcmidi_snd *pm, + unsigned char status, unsigned char note, unsigned char velocity) +{ + unsigned long flags; + unsigned char buffer[3]; + + buffer[0] = status; + buffer[1] = note; + buffer[2] = velocity; + + spin_lock_irqsave(&pm->rawmidi_in_lock, flags); + + if (!pm->in_substream) + goto drop_note; + if (!test_bit(pm->in_substream->number, &pm->in_triggered)) + goto drop_note; + + snd_rawmidi_receive(pm->in_substream, buffer, 3); + +drop_note: + spin_unlock_irqrestore(&pm->rawmidi_in_lock, flags); + + return; +} + +void pcmidi_sustained_note_release(unsigned long data) +{ + struct pcmidi_sustain *pms = (struct pcmidi_sustain *)data; + + pcmidi_send_note(pms->pm, pms->status, pms->note, pms->velocity); + pms->in_use = 0; +} + +void init_sustain_timers(struct pcmidi_snd *pm) +{ + struct pcmidi_sustain *pms; + unsigned i; + + for (i = 0; i < PCMIDI_SUSTAINED_MAX; i++) { + pms = &pm->sustained_notes[i]; + pms->in_use = 0; + pms->pm = pm; + setup_timer(&pms->timer, pcmidi_sustained_note_release, + (unsigned long)pms); + } +} + +void stop_sustain_timers(struct pcmidi_snd *pm) +{ + struct pcmidi_sustain *pms; + unsigned i; + + for (i = 0; i < PCMIDI_SUSTAINED_MAX; i++) { + pms = &pm->sustained_notes[i]; + pms->in_use = 1; + del_timer_sync(&pms->timer); + } +} + +static int pcmidi_get_output_report(struct pcmidi_snd *pm) +{ + struct hid_device *hdev = pm->pk->hdev; + struct hid_report *report; + + list_for_each_entry(report, + &hdev->report_enum[HID_OUTPUT_REPORT].report_list, list) { + if (!(6 == report->id)) + continue; + + if (report->maxfield < 1) { + dev_err(&hdev->dev, "output report is empty\n"); + break; + } + if (report->field[0]->report_count != 2) { + dev_err(&hdev->dev, "field count too low\n"); + break; + } + pm->pcmidi_report6 = report; + return 0; + } + /* should never get here */ + return -ENODEV; +} + +static void pcmidi_submit_output_report(struct pcmidi_snd *pm, int state) +{ + struct hid_device *hdev = pm->pk->hdev; + struct hid_report *report = pm->pcmidi_report6; + report->field[0]->value[0] = 0x01; + report->field[0]->value[1] = state; + + usbhid_submit_report(hdev, report, USB_DIR_OUT); +} + +static int pcmidi_handle_report1(struct pcmidi_snd *pm, u8 *data) +{ + u32 bit_mask; + + bit_mask = data[1]; + bit_mask = (bit_mask << 8) | data[2]; + bit_mask = (bit_mask << 8) | data[3]; + + dbg_hid("pcmidi mode: %d\n", pm->midi_mode); + + /*KEY_MAIL or octave down*/ + if (pm->midi_mode && bit_mask == 0x004000) { + /* octave down */ + pm->midi_octave--; + if (pm->midi_octave < -2) + pm->midi_octave = -2; + dbg_hid("pcmidi mode: %d octave: %d\n", + pm->midi_mode, pm->midi_octave); + return 1; + } + /*KEY_WWW or sustain*/ + else if (pm->midi_mode && bit_mask == 0x000004) { + /* sustain on/off*/ + pm->midi_sustain_mode ^= 0x1; + return 1; + } + + return 0; /* continue key processing */ +} + +static int pcmidi_handle_report3(struct pcmidi_snd *pm, u8 *data, int size) +{ + struct pcmidi_sustain *pms; + unsigned i, j; + unsigned char status, note, velocity; + + unsigned num_notes = (size-1)/2; + for (j = 0; j < num_notes; j++) { + note = data[j*2+1]; + velocity = data[j*2+2]; + + if (note < 0x81) { /* note on */ + status = 128 + 16 + pm->midi_channel; /* 1001nnnn */ + note = note - 0x54 + PCMIDI_MIDDLE_C + + (pm->midi_octave * 12); + if (0 == velocity) + velocity = 1; /* force note on */ + } else { /* note off */ + status = 128 + pm->midi_channel; /* 1000nnnn */ + note = note - 0x94 + PCMIDI_MIDDLE_C + + (pm->midi_octave*12); + + if (pm->midi_sustain_mode) { + for (i = 0; i < PCMIDI_SUSTAINED_MAX; i++) { + pms = &pm->sustained_notes[i]; + if (!pms->in_use) { + pms->status = status; + pms->note = note; + pms->velocity = velocity; + pms->in_use = 1; + + mod_timer(&pms->timer, + jiffies + + msecs_to_jiffies(pm->midi_sustain)); + return 1; + } + } + } + } + pcmidi_send_note(pm, status, note, velocity); + } + + return 1; +} + +static int pcmidi_handle_report4(struct pcmidi_snd *pm, u8 *data) +{ + unsigned key; + u32 bit_mask; + u32 bit_index; + + bit_mask = data[1]; + bit_mask = (bit_mask << 8) | data[2]; + bit_mask = (bit_mask << 8) | data[3]; + + /* break keys */ + for (bit_index = 0; bit_index < 24; bit_index++) { + key = pm->last_key[bit_index]; + if (!((0x01 << bit_index) & bit_mask)) { + input_event(pm->input_ep82, EV_KEY, + pm->last_key[bit_index], 0); + pm->last_key[bit_index] = 0; + } + } + + /* make keys */ + for (bit_index = 0; bit_index < 24; bit_index++) { + key = 0; + switch ((0x01 << bit_index) & bit_mask) { + case 0x000010: /* Fn lock*/ + pm->fn_state ^= 0x000010; + if (pm->fn_state) + pcmidi_submit_output_report(pm, 0xc5); + else + pcmidi_submit_output_report(pm, 0xc6); + continue; + case 0x020000: /* midi launcher..send a key (qwerty) or not? */ + pcmidi_submit_output_report(pm, 0xc1); + pm->midi_mode ^= 0x01; + + dbg_hid("pcmidi mode: %d\n", pm->midi_mode); + continue; + case 0x100000: /* KEY_MESSENGER or octave up */ + dbg_hid("pcmidi mode: %d\n", pm->midi_mode); + if (pm->midi_mode) { + pm->midi_octave++; + if (pm->midi_octave > 2) + pm->midi_octave = 2; + dbg_hid("pcmidi mode: %d octave: %d\n", + pm->midi_mode, pm->midi_octave); + continue; + } else + key = KEY_MESSENGER; + break; + case 0x400000: + key = KEY_CALENDAR; + break; + case 0x080000: + key = KEY_ADDRESSBOOK; + break; + case 0x040000: + key = KEY_DOCUMENTS; + break; + case 0x800000: + key = KEY_WORDPROCESSOR; + break; + case 0x200000: + key = KEY_SPREADSHEET; + break; + case 0x010000: + key = KEY_COFFEE; + break; + case 0x000100: + key = KEY_HELP; + break; + case 0x000200: + key = KEY_SEND; + break; + case 0x000400: + key = KEY_REPLY; + break; + case 0x000800: + key = KEY_FORWARDMAIL; + break; + case 0x001000: + key = KEY_NEW; + break; + case 0x002000: + key = KEY_OPEN; + break; + case 0x004000: + key = KEY_CLOSE; + break; + case 0x008000: + key = KEY_SAVE; + break; + case 0x000001: + key = KEY_UNDO; + break; + case 0x000002: + key = KEY_REDO; + break; + case 0x000004: + key = KEY_SPELLCHECK; + break; + case 0x000008: + key = KEY_PRINT; + break; + } + if (key) { + input_event(pm->input_ep82, EV_KEY, key, 1); + pm->last_key[bit_index] = key; + } + } + + return 1; +} + +int pcmidi_handle_report( + struct pcmidi_snd *pm, unsigned report_id, u8 *data, int size) +{ + int ret = 0; + + switch (report_id) { + case 0x01: /* midi keys (qwerty)*/ + ret = pcmidi_handle_report1(pm, data); + break; + case 0x03: /* midi keyboard (musical)*/ + ret = pcmidi_handle_report3(pm, data, size); + break; + case 0x04: /* multimedia/midi keys (qwerty)*/ + ret = pcmidi_handle_report4(pm, data); + break; + } + return ret; +} + +void pcmidi_setup_extra_keys(struct pcmidi_snd *pm, struct input_dev *input) +{ + /* reassigned functionality for N/A keys + MY PICTURES => KEY_WORDPROCESSOR + MY MUSIC=> KEY_SPREADSHEET + */ + unsigned int keys[] = { + KEY_FN, + KEY_MESSENGER, KEY_CALENDAR, + KEY_ADDRESSBOOK, KEY_DOCUMENTS, + KEY_WORDPROCESSOR, + KEY_SPREADSHEET, + KEY_COFFEE, + KEY_HELP, KEY_SEND, + KEY_REPLY, KEY_FORWARDMAIL, + KEY_NEW, KEY_OPEN, + KEY_CLOSE, KEY_SAVE, + KEY_UNDO, KEY_REDO, + KEY_SPELLCHECK, KEY_PRINT, + 0 + }; + + unsigned int *pkeys = &keys[0]; + unsigned short i; + + if (pm->ifnum != 1) /* only set up ONCE for interace 1 */ + return; + + pm->input_ep82 = input; + + for (i = 0; i < 24; i++) + pm->last_key[i] = 0; + + while (*pkeys != 0) { + set_bit(*pkeys, pm->input_ep82->keybit); + ++pkeys; + } +} + +static int pcmidi_set_operational(struct pcmidi_snd *pm) +{ + if (pm->ifnum != 1) + return 0; /* only set up ONCE for interace 1 */ + + pcmidi_get_output_report(pm); + pcmidi_submit_output_report(pm, 0xc1); + return 0; +} + +static int pcmidi_snd_free(struct snd_device *dev) +{ + return 0; +} + +static int pcmidi_in_open(struct snd_rawmidi_substream *substream) +{ + struct pcmidi_snd *pm = substream->rmidi->private_data; + + dbg_hid("pcmidi in open\n"); + pm->in_substream = substream; + return 0; +} + +static int pcmidi_in_close(struct snd_rawmidi_substream *substream) +{ + dbg_hid("pcmidi in close\n"); + return 0; +} + +static void pcmidi_in_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct pcmidi_snd *pm = substream->rmidi->private_data; + + dbg_hid("pcmidi in trigger %d\n", up); + + pm->in_triggered = up; +} + +static struct snd_rawmidi_ops pcmidi_in_ops = { + .open = pcmidi_in_open, + .close = pcmidi_in_close, + .trigger = pcmidi_in_trigger +}; + +int pcmidi_snd_initialise(struct pcmidi_snd *pm) +{ + static int dev; + struct snd_card *card; + struct snd_rawmidi *rwmidi; + int err; + + static struct snd_device_ops ops = { + .dev_free = pcmidi_snd_free, + }; + + if (pm->ifnum != 1) + return 0; /* only set up midi device ONCE for interace 1 */ + + if (dev >= SNDRV_CARDS) + return -ENODEV; + + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* Setup sound card */ + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) { + pk_error("failed to create pc-midi sound card\n"); + err = -ENOMEM; + goto fail; + } + pm->card = card; + + /* Setup sound device */ + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, pm, &ops); + if (err < 0) { + pk_error("failed to create pc-midi sound device: error %d\n", + err); + goto fail; + } + + strncpy(card->driver, shortname, sizeof(card->driver)); + strncpy(card->shortname, shortname, sizeof(card->shortname)); + strncpy(card->longname, longname, sizeof(card->longname)); + + /* Set up rawmidi */ + err = snd_rawmidi_new(card, card->shortname, 0, + 0, 1, &rwmidi); + if (err < 0) { + pk_error("failed to create pc-midi rawmidi device: error %d\n", + err); + goto fail; + } + pm->rwmidi = rwmidi; + strncpy(rwmidi->name, card->shortname, sizeof(rwmidi->name)); + rwmidi->info_flags = SNDRV_RAWMIDI_INFO_INPUT; + rwmidi->private_data = pm; + + snd_rawmidi_set_ops(rwmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &pcmidi_in_ops); + + snd_card_set_dev(card, &pm->pk->hdev->dev); + + /* create sysfs variables */ + err = device_create_file(&pm->pk->hdev->dev, + sysfs_device_attr_channel); + if (err < 0) { + pk_error("failed to create sysfs attribute channel: error %d\n", + err); + goto fail; + } + + err = device_create_file(&pm->pk->hdev->dev, + sysfs_device_attr_sustain); + if (err < 0) { + pk_error("failed to create sysfs attribute sustain: error %d\n", + err); + goto fail_attr_sustain; + } + + err = device_create_file(&pm->pk->hdev->dev, + sysfs_device_attr_octave); + if (err < 0) { + pk_error("failed to create sysfs attribute octave: error %d\n", + err); + goto fail_attr_octave; + } + + spin_lock_init(&pm->rawmidi_in_lock); + + init_sustain_timers(pm); + pcmidi_set_operational(pm); + + /* register it */ + err = snd_card_register(card); + if (err < 0) { + pk_error("failed to register pc-midi sound card: error %d\n", + err); + goto fail_register; + } + + dbg_hid("pcmidi_snd_initialise finished ok\n"); + return 0; + +fail_register: + stop_sustain_timers(pm); + device_remove_file(&pm->pk->hdev->dev, sysfs_device_attr_octave); +fail_attr_octave: + device_remove_file(&pm->pk->hdev->dev, sysfs_device_attr_sustain); +fail_attr_sustain: + device_remove_file(&pm->pk->hdev->dev, sysfs_device_attr_channel); +fail: + if (pm->card) { + snd_card_free(pm->card); + pm->card = NULL; + } + return err; +} + +int pcmidi_snd_terminate(struct pcmidi_snd *pm) +{ + if (pm->card) { + stop_sustain_timers(pm); + + device_remove_file(&pm->pk->hdev->dev, + sysfs_device_attr_channel); + device_remove_file(&pm->pk->hdev->dev, + sysfs_device_attr_sustain); + device_remove_file(&pm->pk->hdev->dev, + sysfs_device_attr_octave); + + snd_card_disconnect(pm->card); + snd_card_free_when_closed(pm->card); + } + + return 0; +} + +/* + * PC-MIDI report descriptor for report id is wrong. + */ +static void pk_report_fixup(struct hid_device *hdev, __u8 *rdesc, + unsigned int rsize) +{ + if (rsize == 178 && + rdesc[111] == 0x06 && rdesc[112] == 0x00 && + rdesc[113] == 0xff) { + dev_info(&hdev->dev, "fixing up pc-midi keyboard report " + "descriptor\n"); + + rdesc[144] = 0x18; /* report 4: was 0x10 report count */ + } +} + +static int pk_input_mapping(struct hid_device *hdev, struct hid_input *hi, + struct hid_field *field, struct hid_usage *usage, + unsigned long **bit, int *max) +{ + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + struct pcmidi_snd *pm; + + pm = pk->pm; + + if (HID_UP_MSVENDOR == (usage->hid & HID_USAGE_PAGE) && + 1 == pm->ifnum) { + pcmidi_setup_extra_keys(pm, hi->input); + return 0; + } + + return 0; +} + + +static int pk_raw_event(struct hid_device *hdev, struct hid_report *report, + u8 *data, int size) +{ + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + int ret = 0; + + if (1 == pk->pm->ifnum) { + if (report->id == data[0]) + switch (report->id) { + case 0x01: /* midi keys (qwerty)*/ + case 0x03: /* midi keyboard (musical)*/ + case 0x04: /* extra/midi keys (qwerty)*/ + ret = pcmidi_handle_report(pk->pm, + report->id, data, size); + break; + } + } + + return ret; +} + +static int pk_probe(struct hid_device *hdev, const struct hid_device_id *id) +{ + int ret; + struct usb_interface *intf = to_usb_interface(hdev->dev.parent); + unsigned short ifnum = intf->cur_altsetting->desc.bInterfaceNumber; + unsigned long quirks = id->driver_data; + struct pk_device *pk; + struct pcmidi_snd *pm = NULL; + + pk = kzalloc(sizeof(*pk), GFP_KERNEL); + if (pk == NULL) { + dev_err(&hdev->dev, "prodikeys: can't alloc descriptor\n"); + return -ENOMEM; + } + + pk->hdev = hdev; + + pm = kzalloc(sizeof(*pm), GFP_KERNEL); + if (pm == NULL) { + dev_err(&hdev->dev, + "prodikeys: can't alloc descriptor\n"); + ret = -ENOMEM; + goto err_free; + } + + pm->pk = pk; + pk->pm = pm; + pm->ifnum = ifnum; + + hid_set_drvdata(hdev, pk); + + ret = hid_parse(hdev); + if (ret) { + dev_err(&hdev->dev, "prodikeys: hid parse failed\n"); + goto err_free; + } + + if (quirks & PK_QUIRK_NOGET) { /* hid_parse cleared all the quirks */ + hdev->quirks |= HID_QUIRK_NOGET; + } + + ret = hid_hw_start(hdev, HID_CONNECT_DEFAULT); + if (ret) { + dev_err(&hdev->dev, "prodikeys: hw start failed\n"); + goto err_free; + } + + ret = pcmidi_snd_initialise(pm); + if (ret < 0) + goto err_stop; + + return 0; +err_stop: + hid_hw_stop(hdev); +err_free: + if (pm != NULL) + kfree(pm); + + kfree(pk); + return ret; +} + +static void pk_remove(struct hid_device *hdev) +{ + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + struct pcmidi_snd *pm; + + pm = pk->pm; + if (pm) { + pcmidi_snd_terminate(pm); + kfree(pm); + } + + hid_hw_stop(hdev); + + kfree(pk); +} + +static const struct hid_device_id pk_devices[] = { + {HID_USB_DEVICE(USB_VENDOR_ID_CREATIVELABS, + USB_DEVICE_ID_PRODIKEYS_PCMIDI), + .driver_data = PK_QUIRK_NOGET}, + { } +}; +MODULE_DEVICE_TABLE(hid, pk_devices); + +static struct hid_driver pk_driver = { + .name = "prodikeys", + .id_table = pk_devices, + .report_fixup = pk_report_fixup, + .input_mapping = pk_input_mapping, + .raw_event = pk_raw_event, + .probe = pk_probe, + .remove = pk_remove, +}; + +static int pk_init(void) +{ + int ret; + + ret = hid_register_driver(&pk_driver); + if (ret) + printk(KERN_ERR "can't register prodikeys driver\n"); + + return ret; +} + +static void pk_exit(void) +{ + hid_unregister_driver(&pk_driver); +} + +module_init(pk_init); +module_exit(pk_exit); +MODULE_LICENSE("GPL"); diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/Kconfig b/drivers/hid/Kconfig --- a/drivers/hid/Kconfig 2010-03-09 13:53:05.930183131 +0000 +++ b/drivers/hid/Kconfig 2010-05-06 14:00:57.880373988 +0100 @@ -100,6 +100,22 @@ config HID_CHICONY ---help--- Support for Chicony Tactical pad. +config HID_PRODIKEYS + tristate "Prodikeys PC-MIDI Keyboard support" if EMBEDDED + depends on USB_HID && SND + select SND_RAWMIDI + default !EMBEDDED + ---help--- + Support for Prodikeys PC-MIDI Keyboard device support. + Say Y here to enable support for this device. + - Prodikeys PC-MIDI keyboard. + The Prodikeys PC-MIDI acts as a USB Audio device, with one MIDI + input and one MIDI output. These MIDI jacks appear as + a sound "card" in the ALSA sound system. + Note: if you say N here, this device will still function as a basic + multimedia keyboard, but will lack support for the musical keyboard + and some additional multimedia keys. + config HID_CYPRESS tristate "Cypress" if EMBEDDED depends on USB_HID diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/Makefile b/drivers/hid/Makefile --- a/drivers/hid/Makefile 2010-03-09 13:53:06.036326844 +0000 +++ b/drivers/hid/Makefile 2010-05-05 12:57:45.986487233 +0100 @@ -41,6 +41,7 @@ obj-$(CONFIG_HID_MONTEREY) += hid-monter obj-$(CONFIG_HID_MOSART) += hid-mosart.o obj-$(CONFIG_HID_NTRIG) += hid-ntrig.o obj-$(CONFIG_HID_ORTEK) += hid-ortek.o +obj-$(CONFIG_HID_PRODIKEYS) += hid-prodikeys.o obj-$(CONFIG_HID_QUANTA) += hid-quanta.o obj-$(CONFIG_HID_PANTHERLORD) += hid-pl.o obj-$(CONFIG_HID_PETALYNX) += hid-petalynx.o ___________________________________________________________ Copy addresses and emails from any email account to Yahoo! 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2 1

[alsa-devel] [PATCH 0/5] Enabling ASoC paths over suspend
by Mark Brown 10 May '10

10 May '10

The following patch series implements support for keeping some paths in the audio codec enabled while the system is suspended. This is primarily intended for mobile phones, where the GSM modem is normally capable of supporting a voice call with no intervention from the CPU so power can be saved by suspending the Linux system. This is done by allowing DAPM nodes to be marked as ignoring suspend. Any audio paths between two widgets set up like this at suspend time will be kept active and the CODEC device will not be suspended. The existing testing is very light (especially for digital paths), the intention is to do more of that get this merged into Takashi's tree early next week. git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git suspend-enable Mark Brown (5): ASoC: Remove unneeded suspend bias managment from CODEC drivers ASoC: Remove unused DAPM suspend flag ASoC: Refactor DAPM suspend handling ASoC: Support leaving paths enabled over system suspend ASoC: Allow active paths from the GSM modem while the GTA02 is suspended include/sound/soc-dapm.h | 3 +- sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/stac9766.c | 3 - sound/soc/codecs/tlv320aic23.c | 1 - sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 3 - sound/soc/codecs/twl4030.c | 1 - sound/soc/codecs/twl6040.c | 1 - sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm8350.c | 3 - sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm9712.c | 3 - sound/soc/codecs/wm9713.c | 3 - sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 8 +++ sound/soc/soc-core.c | 35 ++++++++++- sound/soc/soc-dapm.c | 92 +++++++++++++++++++----------- 25 files changed, 107 insertions(+), 69 deletions(-)

3 8

[alsa-devel] Not all channels working with IDT 92HD83C1X5
by Thierry Reding 10 May '10

10 May '10

Hi, I have a board with an IDT 92HD83C1X5 codec connected to an Intel HDA controller. When running a older "reference" Ubuntu 9.04 (provided by the vendor) the codec works fine. I've been diffing the sources provided of the 2.6.28 kernel against vanilla 2.6.28 and couldn't find any significant changes regarding HDA or the patch_sigmatel.c that has the support for this codec. However, when I use kernel 2.6.34-rc6 on the same hardware, only one channel is working properly, all others remain muted. After some investigation I came up with the attached patch that makes all channels work again. I am aware that this is far from being acceptable code and that this is only a workaround. Still, I am interested in fixing this properly, but I don't really know where to start. Also attached is an extract of dmesg with the actual output of the debugging messages added by the attached patch. Cheers, Thierry

1 0

[alsa-devel] [PATCH 10/11] ASoC: S3C64XX: IISv4: Add CPU driver
by Jassi Brar 10 May '10

10 May '10

Add the CPU driver for the IISv4 block found on S3C6410. For now, the driver is almost a copy of s3c64xx-i2s.c but it should diverge as more IISv4 specific stuff is added. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 5 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c64xx-i2s-v4.c | 209 ++++++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + 4 files changed, 217 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/s3c64xx-i2s-v4.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 15fe57e..c0c7edf 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -24,6 +24,11 @@ config SND_S3C64XX_SOC_I2S select SND_S3C_I2SV2_SOC select S3C64XX_DMA +config SND_S3C64XX_SOC_I2S_V4 + tristate + select SND_S3C_I2SV2_SOC + select S3C64XX_DMA + config SND_S3C_SOC_PCM tristate diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index df071a3..81d8dc5 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -4,6 +4,7 @@ snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c-ac97-objs := s3c-ac97.o +snd-soc-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o snd-soc-s3c-pcm-objs := s3c-pcm.o @@ -12,6 +13,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S_V4) += snd-soc-s3c64xx-i2s-v4.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o diff --git a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c new file mode 100644 index 0000000..06db130 --- /dev/null +++ b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c @@ -0,0 +1,209 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s-v4.c + * + * ALSA SoC Audio Layer - S3C64XX I2Sv4 driver + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include <mach/gpio-bank-c.h> +#include <mach/gpio-bank-h.h> +#include <plat/gpio-cfg.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c64xx-i2s.h" + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2Sv4 PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2Sv4 PCM Stereo in" +}; + +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_out; +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_in; +static struct s3c_i2sv2_info s3c64xx_i2sv4; + +struct snd_soc_dai s3c64xx_i2s_v4_dai; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_v4_dai); + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2sv4_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* configure GPIO for i2s port */ + s3c_gpio_cfgpin(S3C64XX_GPC(4), S3C64XX_GPC4_I2S_V40_DO0); + s3c_gpio_cfgpin(S3C64XX_GPC(5), S3C64XX_GPC5_I2S_V40_DO1); + s3c_gpio_cfgpin(S3C64XX_GPC(7), S3C64XX_GPC7_I2S_V40_DO2); + s3c_gpio_cfgpin(S3C64XX_GPH(6), S3C64XX_GPH6_I2S_V40_BCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(7), S3C64XX_GPH7_I2S_V40_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(8), S3C64XX_GPH8_I2S_V40_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(9), S3C64XX_GPH9_I2S_V40_DI); + + return 0; +} + +static int s3c_i2sv4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + dev_dbg(cpu_dai->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: r: IISMOD: %x\n", __func__, iismod); + + iismod &= ~S3C64XX_IISMOD_BLC_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C64XX_IISMOD_BLC_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S24_LE: + iismod |= S3C64XX_IISMOD_BLC_24BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static struct snd_soc_dai_ops s3c64xx_i2sv4_dai_ops = { + .hw_params = s3c_i2sv4_hw_params, +}; + +static __devinit int s3c64xx_i2sv4_dev_probe(struct platform_device *pdev) +{ + struct s3c_i2sv2_info *i2s; + struct snd_soc_dai *dai; + int ret; + + i2s = &s3c64xx_i2sv4; + dai = &s3c64xx_i2s_v4_dai; + + if (dai->dev) { + dev_dbg(dai->dev, "%s: \ + I2Sv4 instance already registered!\n", __func__); + return -EBUSY; + } + + dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s-v4"; + dai->id = 0; + dai->symmetric_rates = 1; + dai->playback.channels_min = 2; + dai->playback.channels_max = 2; + dai->playback.rates = S3C64XX_I2S_RATES; + dai->playback.formats = S3C64XX_I2S_FMTS; + dai->capture.channels_min = 2; + dai->capture.channels_max = 2; + dai->capture.rates = S3C64XX_I2S_RATES; + dai->capture.formats = S3C64XX_I2S_FMTS; + dai->probe = s3c64xx_i2sv4_probe; + dai->ops = &s3c64xx_i2sv4_dai_ops; + + i2s->feature |= S3C_FEATURE_CDCLKCON; + + i2s->dma_capture = &s3c64xx_i2sv4_pcm_stereo_in; + i2s->dma_playback = &s3c64xx_i2sv4_pcm_stereo_out; + + i2s->dma_capture->channel = DMACH_HSI_I2SV40_RX; + i2s->dma_capture->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_HSI_I2SV40_TX; + i2s->dma_playback->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISTXD; + + i2s->dma_capture->client = &s3c64xx_dma_client_in; + i2s->dma_capture->dma_size = 4; + i2s->dma_playback->client = &s3c64xx_dma_client_out; + i2s->dma_playback->dma_size = 4; + + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(i2s->iis_cclk); + goto err; + } + + clk_enable(i2s->iis_cclk); + + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); + if (ret) + goto err_clk; + + ret = s3c_i2sv2_register_dai(dai); + if (ret != 0) + goto err_i2sv2; + + return 0; + +err_i2sv2: + /* Not implemented for I2Sv2 core yet */ +err_clk: + clk_put(i2s->iis_cclk); +err: + return ret; +} + +static __devexit int s3c64xx_i2sv4_dev_remove(struct platform_device *pdev) +{ + dev_err(&pdev->dev, "Device removal not yet supported\n"); + return 0; +} + +static struct platform_driver s3c64xx_i2sv4_driver = { + .probe = s3c64xx_i2sv4_dev_probe, + .remove = s3c64xx_i2sv4_dev_remove, + .driver = { + .name = "s3c64xx-iis-v4", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c64xx_i2sv4_init(void) +{ + return platform_driver_register(&s3c64xx_i2sv4_driver); +} +module_init(s3c64xx_i2sv4_init); + +static void __exit s3c64xx_i2sv4_exit(void) +{ + platform_driver_unregister(&s3c64xx_i2sv4_driver); +} +module_exit(s3c64xx_i2sv4_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("S3C64XX I2Sv4 SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index f27ed50..7a40f43 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -37,5 +37,6 @@ struct clk; SNDRV_PCM_FMTBIT_S24_LE) extern struct snd_soc_dai s3c64xx_i2s_dai[]; +extern struct snd_soc_dai s3c64xx_i2s_v4_dai; #endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ -- 1.6.2.5

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