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[alsa-devel] problem in set frames and buffer size‏
by Haiping Wang 01 Dec '08

01 Dec '08

hello all,I have write a alsa app from a example from internet.The original code is good running. format 16bitchannel 2rate 44100/* This example reads from the default PCM device and writes to standard output for 5 seconds of data. */ /* Use the newer ALSA API */ #define ALSA_PCM_NEW_HW_PARAMS_API #include <alsa/asoundlib.h> int main() { long loops; int rc; int size; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; snd_pcm_uframes_t frames; char *buffer; /* Open PCM device for recording (capture). */ rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0); if (rc < 0) { fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); exit(1); } /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&params); /* Fill it in with default values. */ snd_pcm_hw_params_any(handle, params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Signed 16-bit little-endian format */ snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(handle, params, 2); /* 44100 bits/second sampling rate (CD quality) */ val = 44100; snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); /* Set period size to 32 frames. */ frames = 32; snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); if (rc < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); exit(1); } /* Use a buffer large enough to hold one period */ snd_pcm_hw_params_get_period_size(params, &frames, &dir); size = frames * 4; /* 2 bytes/sample, 2 channels */ buffer = (char *) malloc(size); /* We want to loop for 5 seconds */ snd_pcm_hw_params_get_period_time(params, &val, &dir); loops = 5000000 / val; while (loops > 0) { loops--; rc = snd_pcm_readi(handle, buffer, frames); if (rc == -EPIPE) { /* EPIPE means overrun */ fprintf(stderr, "overrun occurred\n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr, "error from read: %s\n", snd_strerror(rc)); } else if (rc != (int)frames) { fprintf(stderr, "short read, read %d frames\n", rc); } rc = write(1, buffer, size); if (rc != size) fprintf(stderr, "short write: wrote %d bytes\n", rc); } snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer); return 0; } now I want to change channel and format to write my own programm.format = 8bit, 24bit and 32bitchannel = 1 or 2how can i change the parameters in frames and buffer size.for each format 8/16/24/32bit how to set frames = ?and alsa channel 1/2 how to set buffersize??? I try many times and failed .I want your help. thanks!! _________________________________________________________________ Spielspaß für Jedermann! Mit tausenden von Spielen für Windows Vista! http://redirect.gimas.net/?n=M0812xSpieleVistaDE

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[alsa-devel] Merged upstream to topic/asoc branch
by Takashi Iwai 01 Dec '08

01 Dec '08

Hi, FYI, I merged the current Linus tree back to topic/asoc branch on sound git tree for clean-up works including the latest commit in sound/soc/fsl. If you have any problems with it, please let me know. thanks, Takashi

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[alsa-devel] (SoC) Does i.MX27's SSI work as I2S clock master?
by ngc＠drvlabo.jp 01 Dec '08

01 Dec '08

Hello, lists. Now I'm porting WM8728 driver with i.MX27ADS, based on Freescale's BSP kernel(2.6.22) and refering to ALSA SoC git repository. I'm connecting the WM8728 to the SSI2 controller via SSI-3 signal set. Then, my ADS board has altered: i.MX27 WM8728 --------------------- SSI3_CLK BCLKIN SSI3_FS LRCIN SSI3_TXDAT DIN TOUT/SSI2_MCLK MCLK Of course, I added some codes to configure GPIO pins as appropriate into board specific sources. Then, I'm trying to play sounds via WM8728 and have not hear any sounds yet. I checked SSI-3 signal set with an oscilloscope, no clock signal appears on SSI3_CLK. I tried the various SSI setting, I2S normal, I2S master, asychronous/synchronous, normal mode/network mode, ... But, I can't see clock signal, yet. I want to know an example setting to act i.MX' SSI as I2S clock master. Regards, -------------- NOGUCHI Hiroshi ngc(a)drvlabo.jp

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Re: [alsa-devel] [ath5k-devel] Bugs on aspire one A150
by Maxim Levitsky 01 Dec '08

01 Dec '08

On Tue, 2008-11-11 at 00:12 +0200, Nick Kossifidis wrote: > 2008/11/10 Maxim Levitsky <maximlevitsky(a)gmail.com>: > > Bob Copeland wrote: > >> On Fri, Nov 7, 2008 at 10:26 PM, Maxim Levitsky <maximlevitsky(a)gmail.com> wrote: > >>> ** 2 - wireless: not to mention the fact that ath5k wasn't installed by > >>> default in ubuntu... > >>> wireless more or less works, but kernel log is full of backtraces. > >> > >> As you've seen, these should at least be gone now... > >> > >>> Was able to connect to my WPA2 access point. > >>> Sometimes wireless fails completely, especially after suspend to ram. > >> > >> Did you get noise calibration failures in dmesg at this point? Felix posted > >> a patch recently that handles them better. > > Yep, but they seems to be gone in latest -git too. > > Wireless work fine now, it seems like my iwl3945. > > > >> > >>> Advanced features like monitor/injection work, but when I changed the card's > >>> mac address it stopped working. > >>> I also noticed that if I then start airodump, then wireless works with new > >>> mac. > > Why it doesn't accept new mac?, can this be fixed? > > > > > >> > >>> ** 4 - wireless led doesn't work. > >>> ath5k devs, can you fix this? > >> > >> Currently all the LED code we have in there is for setting a particular gpio > >> for particular laptops (and they differ between models). We only have quirks > >> for IBM and HP, none for Acer. Also, the legacy-hal seems to have a different > >> strategy for 2425 chips. I can try to hack up some code for you to test. > >> > > > > I have seen on the web that madwifi did support the led with some gpio settings > > > > sysctl -w dev.wifi0.ledpin=3 > > sysctl -w dev.wifi0.softled=1 > > > > from https://help.ubuntu.com/community/AspireOne > > > > Best regards, > > Maxim Levitsky > > i think that some led settings are stored on EEPROM, let me check it out... Hi, Any update on that? Best regards, Maxim Levitsky > >

1 0

[alsa-devel] Unused TWL4030 macros
by Takashi Iwai 01 Dec '08

01 Dec '08

Hi, after the commit b0bd53a7399f65e2d1b37cd44c5003e55b886c1e, we get compile warnings as: sound/soc/codecs/twl4030.c:233: warning: 'snd_soc_get_volsw_twl4030' defined but not used sound/soc/codecs/twl4030.c:262: warning: 'snd_soc_put_volsw_twl4030' defined but not used sound/soc/codecs/twl4030.c:291: warning: 'snd_soc_get_volsw_r2_twl4030' defined but not used sound/soc/codecs/twl4030.c:318: warning: 'snd_soc_put_volsw_r2_twl4030' defined but not used Will the new macros ever used? Takashi

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[alsa-devel] [PATCH] ALSA: Add WM9715 to AC97 IDs
by Mark Brown 01 Dec '08

01 Dec '08

The WM9715 is software compatible with the WM9711 and WM9712. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/pci/ac97/ac97_codec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6704acb..e8375ff 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -175,7 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL}, { 0x574d4C05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL}, { 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL}, -{ 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL}, +{ 0x574d4C12, 0xffffffff, "WM9711,WM9712,WM9715", patch_wolfson11, NULL}, { 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF}, { 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL }, { 0x594d4802, 0xffffffff, "YMF752", NULL, NULL }, -- 1.5.6.5

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[alsa-devel] [PATCH 0/4] ASoC: TWL4030: Gain control updates, adding pre-DAC routing controls
by Peter Ujfalusi 01 Dec '08

01 Dec '08

Hello, This series adds 'proper' gain controls for: Per-DAC common gain controls. There are three gain controls asspciated wit a DAC in TWL, There are four DAC can be used (in OPT_MODE=1). Digital fine gain, Digital coarse gain, Analog gain. Every output, except the HandsFree has their own gain controls per channel (L/R) Pre-DAC routings: Digital input namings: SDRL1: TDM #1 SDRR1: TDM #2 SDRL2: TDM #3, I2S Left SDRR2: TDM #4, I2S Right Mono mixing: SDRM1: mixed SDRL1 and SDRR1 SDRM2: mixed SDRL2 and SDRR2 The possible mux settings for the DAC inputs: DACL1 <- SDRL1, SDRM1, SDRM2, SDRL2 DACR1 <- SDRR1, SDRM1, SDRM2, SDRR2 DACL2 <- SDRL2, SDRM2 DACR2 <- SDRR2, SDRM2 --- Peter Ujfalusi (4): ASoC: TWL4030: Change the capture volume control to TLV ASoC: TWL4030: Change the common playback volume controls ASoC: TWL4030: Add volume controls for outputs ASoC: TWL4030: Add pre-DAC digital mux controls sound/soc/codecs/twl4030.c | 59 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 59 insertions(+), 0 deletions(-)

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[alsa-devel] Help with Creative's X-Fi drivers
by James Pearson 01 Dec '08

01 Dec '08

I know the ALSA team don't support Creative's X-Fi drivers, but maybe someone could help with this issue: I've managed to 'backport' Creative's recently GPL'd drivers to run on a CentOS4 kernel with ALSA 1.0.15 - the changes to the code were minor, but one thing I'm not sure about is, in ctpcm.c it has: static int ct_pcm_timer_stop(struct ct_atc_pcm *apcm) { unsigned long flags; CTDPF("%s is called\n", __func__); spin_lock_irqsave(&apcm->timer_lock, flags); apcm->stop_timer = 1; del_timer(&apcm->timer); spin_unlock_irqrestore(&apcm->timer_lock, flags); try_to_del_timer_sync(&apcm->timer); return 0; } The CentOS4/RHEL4 2.6.9 kernel doesn't have try_to_del_timer_sync() - so, should I replace it with del_timer() or del_timer_sync() ? I'm currently using del_timer_sync() - and the driver works (I get sound out the head phone jack) - is this OK? Thanks James Pearson

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[alsa-devel] Restore state around suspend/resume
by Dan Nicholson 01 Dec '08

01 Dec '08

Hi, Currently, suspending the computer via pm-suspend (as done from HAL), causes the ALSA state to be stored and restored via alsactl. Here's what the hook looks like: http://cgit.freedesktop.org/pm-utils/tree/pm/sleep.d/65alsa Is this still necessary on recent kernels? Is there a list of drivers which properly maintain state across suspend/resume? We're trying to cut down on the amount of work done in userspace which slows down suspend cycles. Thanks. -- Dan -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [ARM] pxa: explicit #include <mach/dma.h> in various drivers
by Eric Miao 01 Dec '08

01 Dec '08

Where 'pxa_dma_desc' and 'pxa_{request,free}_dma' are referenced. Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- drivers/mtd/nand/pxa3xx_nand.c | 2 +- drivers/net/smc911x.h | 3 +++ sound/arm/pxa2xx-pcm.h | 2 +- 3 files changed, 5 insertions(+), 2 deletions(-) diff --git a/drivers/mtd/nand/pxa3xx_nand.c b/drivers/mtd/nand/pxa3xx_nand.c index c0fa9c9..64b3be1 100644 --- a/drivers/mtd/nand/pxa3xx_nand.c +++ b/drivers/mtd/nand/pxa3xx_nand.c @@ -20,8 +20,8 @@ #include <linux/mtd/partitions.h> #include <linux/io.h> #include <linux/irq.h> -#include <asm/dma.h> +#include <mach/dma.h> #include <mach/pxa-regs.h> #include <mach/pxa3xx_nand.h> diff --git a/drivers/net/smc911x.h b/drivers/net/smc911x.h index cc7d85b..870b4c3 100644 --- a/drivers/net/smc911x.h +++ b/drivers/net/smc911x.h @@ -200,6 +200,9 @@ static inline void SMC_outsl(struct smc911x_local *lp, int reg, #ifdef SMC_USE_PXA_DMA + +#include <mach/dma.h> + /* * Define the request and free functions * These are unfortunately architecture specific as no generic allocation diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h index 5c4a4d3..65f86b5 100644 --- a/sound/arm/pxa2xx-pcm.h +++ b/sound/arm/pxa2xx-pcm.h @@ -9,7 +9,7 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ -#include <asm/dma.h> +#include <mach/dma.h> struct pxa2xx_runtime_data { int dma_ch; -- 1.6.0.4

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[alsa-devel] 1.0.18-tools : ac3dec won't compile
by Klaus Schulz 30 Nov '08

30 Nov '08

Hi there. Just tried to get the tools 1.0.18 compiled. Following error message was printed when I tried to make ac3dec: gcc -DHAVE_CONFIG_H -I. -I.. -Wall -Werror -O3 -g -MT extract_ac3.o -MD -MP -MF .deps/extract_ac3.Tpo -c -o extract_ac3.o extract_ac3.c cc1: warnings being treated as errors extract_ac3.c: In function 'read_write_next_bytes': extract_ac3.c:163: error: ignoring return value of 'write', declared with attribute warn_unused_result extract_ac3.c:174: error: ignoring return value of 'write', declared with attribute warn_unused_result make[2]: *** [extract_ac3.o] Error 1 make[2]: Leaving directory `/usr/src/Alsa-1.0.18a/alsa-tools-1.0.18/ac3dec/tools' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/src/Alsa-1.0.18a/alsa-tools-1.0.18/ac3dec' make: *** [all] Error 2 Any ideas? THX

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Re: [alsa-devel] [BUG] Long Resume with Riptide
by nokos＠gmx.net 29 Nov '08

29 Nov '08

Hi, can you enable the sound debug options: CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERSOSE=y (be warned it generates lots of log entries) and search for "send cmd ...failed" you will also find some "send cmd" lines containing timing statistics. Then try to reduce the timeoutvalue in sound/pci/riptide/riptide.c CMDIF_TIMEOUT to something near mintime/maxtime (it defaults to 5 sec [500000*10us]) If the error is hard to reproduce you might want to try this: --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -945,8 +945,8 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm, errout: cif->errcnt++; spin_unlock_irqrestore(&cif->lock, irqflags); - snd_printdd - ("send cmd %d hw: 0x%x flag: 0x%x cmd: 0x%x parm: 0x%x ret: 0x%x 0x%x CMDE: %d DATF: %d failed %d\n", + snd_printk(KERN_ERR + "send cmd %d hw: 0x%x flag: 0x%x cmd: 0x%x parm: 0x%x ret: 0x%x 0x%x CMDE: %d DATF: %d failed %d\n", cif->cmdcnt, (int)((void *)&(cmdport->stat) - (void *)hwport), flags, cmd, parm, ret ? ret->retlongs[0] : 0, ret ? ret->retlongs[1] : 0, IS_CMDE(cmdport), IS_DATF(cmdport), and disable the SND_DEBUG stuff (less log spam but you see which commands are failing and together with the reduced CMDIF_TIMEOUT you also see which commands are stalling) Peter Am Freitag, den 28.11.2008, 07:48 +0100 schrieb Eric Sesterhenn: > Hi, > > Sometimes during bootup on this box i get a warning that resume took too long > (I have suspend testing enabled). As far as I can see the riptide driver > takes ~5 seconds to resume. Kernel is 2.6.28-rc5 > > [ 23.857632] PM: test RTC wakeup from 'standby' suspend > [ 23.859062] PM: Syncing filesystems ... done. > [ 23.866226] Freezing user space processes ... (elapsed 0.00 seconds) done. > [ 23.866671] Freezing remaining freezable tasks ... (elapsed 0.00 seconds) done. > [ 23.867527] Suspending console(s) (use no_console_suspend to debug) > [ 24.317553] sd 2:0:0:0: [sda] Synchronizing SCSI cache > [ 24.318470] sd 2:0:0:0: [sda] Stopping disk > [ 24.431264] serial 00:08: disabled > [ 24.432568] RIPTIDE 0000:00:12.0: PCI INT A disabled > [ 24.465203] aic7xxx 0000:00:0c.0: PCI INT A disabled > [ 24.480466] uhci_hcd 0000:00:07.2: PCI INT D disabled > [ 24.481411] PM: suspend devices took 0.616 seconds > [ 24.481663] ACPI: Preparing to enter system sleep state S1 > [ 24.490826] ACPI: Waking up from system sleep state S1 > [ 24.493282] ACPI Warning (nspredef-0852): \_WAK: Return type mismatch - found Integer, expected Package [20080926] > [ 24.494394] uhci_hcd 0000:00:07.2: PCI INT D -> Link[LNKD] -> GSI 9 (level, low) -> IRQ 9 > [ 24.512439] aic7xxx 0000:00:0c.0: enabling device (0116 -> 0117) > [ 24.512471] aic7xxx 0000:00:0c.0: PCI INT A -> Link[LNKB] -> GSI 5 (level, low) -> IRQ 5 > [ 24.532081] eth0: link up, 100Mbps, full-duplex, lpa 0x41E1 > [ 24.548248] RIPTIDE 0000:00:12.0: PCI INT A -> Link[LNKA] -> GSI 11 (level, low) -> IRQ 11 > [ 29.954494] matroxfb 0000:01:00.0: PCI INT A -> Link[LNKA] -> GSI 11 (level, low) -> IRQ 11 > [ 29.957755] serial 00:08: activated > [ 29.962016] sd 2:0:0:0: [sda] Starting disk > [ 29.968733] ata1.00: ACPI cmd ef/03:0c:00:00:00:a0 filtered out > [ 29.968747] ata1.00: ACPI cmd ef/03:22:00:00:00:a0 filtered out > [ 29.969379] ata2.00: ACPI cmd ef/03:0c:00:00:00:a0 filtered out > [ 29.969392] ata2.00: ACPI cmd ef/03:42:00:00:00:a0 filtered out > [ 29.976984] ata1.00: configured for UDMA/33 > [ 29.992727] ata2.00: configured for UDMA/33 > [ 30.016716] ata2.00: configured for UDMA/33 > [ 30.016731] ata2: EH complete > [ 30.360069] CPA self-test: > [ 30.365709] 4k 65533 large 0 gb 0 x 65533[c0000000-cfffc000] miss 0 > [ 30.380333] 4k 65533 large 0 gb 0 x 65533[c0000000-cfffc000] miss 0 > [ 30.392794] 4k 65533 large 0 gb 0 x 65533[c0000000-cfffc000] miss 0 > [ 30.392803] ok. > [ 33.253318] sd 2:0:0:0: [sda] 40031712 512-byte hardware sectors: (20.4 GB/19.0 GiB) > [ 33.253543] sd 2:0:0:0: [sda] Write Protect is off > [ 33.253842] sd 2:0:0:0: [sda] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA > [ 33.254073] sda: detected capacity change from 0 to 20496236544 > [ 33.254398] sd 2:0:0:0: [sda] 40031712 512-byte hardware sectors: (20.4 GB/19.0 GiB) > [ 33.254563] sd 2:0:0:0: [sda] Write Protect is off > [ 33.254858] sd 2:0:0:0: [sda] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA > [ 33.254920] sda: detected capacity change from 0 to 20496236544 > [ 33.298085] PM: resume devices took 8.804 seconds > [ 33.298105] ------------[ cut here ]------------ > [ 33.298114] WARNING: at kernel/power/main.c:177 suspend_test_finish+0x85/0x90() > [ 33.298124] Component: resume devices > [ 33.298130] Modules linked in: > [ 33.298142] Pid: 1, comm: swapper Not tainted 2.6.28-rc5 #60 > [ 33.298151] Call Trace: > [ 33.298176] [<c012a235>] warn_slowpath+0x65/0x90 > [ 33.298195] [<c014d1ef>] ? trace_hardirqs_off_caller+0x1f/0xc0 > [ 33.298208] [<c014d29b>] ? trace_hardirqs_off+0xb/0x10 > [ 33.298232] [<c05c57ff>] ? _spin_unlock_irqrestore+0x4f/0x60 > [ 33.298246] [<c05c5832>] ? _spin_unlock+0x22/0x30 > [ 33.298263] [<c014f475>] ? trace_hardirqs_on_caller+0x115/0x160 > [ 33.298276] [<c014f4cb>] ? trace_hardirqs_on+0xb/0x10 > [ 33.298289] [<c05c2f0a>] ? printk+0x1d/0x23 > [ 33.298301] [<c015ab55>] suspend_test_finish+0x85/0x90 > [ 33.298314] [<c015ac7b>] suspend_devices_and_enter+0xfb/0x1f0 > [ 33.298327] [<c015af64>] enter_state+0x1c4/0x280 > [ 33.298338] [<c015b0fe>] pm_suspend+0x1e/0x30 > [ 33.298352] [<c08d21f9>] test_suspend+0x167/0x19f > [ 33.298366] [<c08d2092>] ? test_suspend+0x0/0x19f > [ 33.298379] [<c010111b>] do_one_initcall+0x2b/0x180 > [ 33.298391] [<c014f4cb>] ? trace_hardirqs_on+0xb/0x10 > [ 33.298412] [<c01ec557>] ? proc_register+0x117/0x1f0 > [ 33.298426] [<c05c5832>] ? _spin_unlock+0x22/0x30 > [ 33.298438] [<c01ec557>] ? proc_register+0x117/0x1f0 > [ 33.298450] [<c01ec111>] ? __proc_create+0xf1/0x110 > [ 33.298463] [<c01ec739>] ? create_proc_entry+0x59/0xa0 > [ 33.298478] [<c016d8b7>] ? register_irq_proc+0x87/0xa0 > [ 33.298491] [<c016d909>] ? init_irq_proc+0x39/0x50 > [ 33.298510] [<c08ba4f5>] kernel_init+0x83/0xdf > [ 33.298521] [<c08ba472>] ? kernel_init+0x0/0xdf > [ 33.298538] [<c010440b>] kernel_thread_helper+0x7/0x10 > [ 33.298547] ---[ end trace 9997c7f634daff1f ]--- > > On other bootups I see the following before the freezer tests, > since in both cases the device gets initialized i guess this > is the same issue. > > [ 23.522685] RIPTIDE 0000:00:12.0: enabling device (0104 -> 0105) > [ 23.524005] RIPTIDE 0000:00:12.0: PCI INT A -> Link[LNKA] -> GSI 11 (level, low) -> IRQ 11 > [ 23.664021] RCU detected CPU 0 stall (t=4294899535/1358 jiffies) > [ 23.664021] Pid: 1, comm: swapper Not tainted 2.6.28-rc5 #60 > [ 23.664021] Call Trace: > [ 23.664021] [<c05c2f0a>] ? printk+0x1d/0x23 > [ 23.664021] [<c016e26d>] __rcu_pending+0x11d/0x1c0 > [ 23.664021] [<c016e327>] rcu_pending+0x17/0x40 > [ 23.664021] [<c0133e2f>] update_process_times+0x2f/0x60 > [ 23.664021] [<c014b0ea>] tick_sched_timer+0x5a/0xd0 > [ 23.664021] [<c014b090>] ? tick_sched_timer+0x0/0xd0 > [ 23.664021] [<c014b090>] ? tick_sched_timer+0x0/0xd0 > [ 23.664021] [<c014257c>] __run_hrtimer+0xac/0xc0 > [ 23.664021] [<c014b090>] ? tick_sched_timer+0x0/0xd0 > [ 23.664021] [<c0142dff>] hrtimer_interrupt+0x12f/0x1a0 > [ 23.664021] [<c0106770>] timer_interrupt+0x40/0x50 > [ 23.664021] [<c016b4c1>] handle_IRQ_event+0x31/0x60 > [ 23.664021] [<c016d18b>] handle_level_irq+0x6b/0xe0 > [ 23.664021] [<c016d120>] ? handle_level_irq+0x0/0xe0 > [ 23.664021] <IRQ> [<c01042bd>] ? common_interrupt+0x21/0x30 > [ 23.664021] [<c01042c4>] ? common_interrupt+0x28/0x30 > [ 23.664021] [<c014007b>] ? posix_cpu_timer_schedule+0xcb/0x130 > [ 23.664021] [<c05c57ec>] ? _spin_unlock_irqrestore+0x3c/0x60 > [ 23.664021] [<c04f91c7>] ? sendcmd+0x217/0x240 > [ 23.664021] [<c04f9841>] ? riptide_reset+0x371/0x6e0 > [ 23.664021] [<c014f475>] ? trace_hardirqs_on_caller+0x115/0x160 > [ 23.664021] [<c04f9bdd>] ? snd_riptide_initialize+0x2d/0xc0 > [ 23.664021] [<c05c15af>] ? snd_card_riptide_probe+0x17f/0x580 > [ 23.664021] [<c010442a>] ? ftrace_call+0x5/0x8 > [ 23.664021] [<c02c2d29>] ? pci_match_id+0x9/0xa0 > [ 23.664021] [<c02c2f5d>] ? pci_match_device+0xbd/0xd0 > [ 23.664021] [<c02c375e>] ? pci_device_probe+0x5e/0x80 > [ 23.664021] [<c0383ec1>] ? driver_probe_device+0xa1/0x2f0 > [ 23.664021] [<c014f4cb>] ? trace_hardirqs_on+0xb/0x10 > [ 23.664021] [<c03841a1>] ? __driver_attach+0x91/0xa0 > [ 23.664021] [<c03835eb>] ? bus_for_each_dev+0x5b/0x80 > [ 23.664021] [<c02c36a0>] ? pci_device_remove+0x0/0x40 > [ 23.664021] [<c0383c3e>] ? driver_attach+0x1e/0x20 > [ 23.664021] [<c0384110>] ? __driver_attach+0x0/0xa0 > [ 23.664021] [<c0382e47>] ? bus_add_driver+0x1c7/0x290 > [ 23.664021] [<c02c36a0>] ? pci_device_remove+0x0/0x40 > [ 23.664021] [<c0384381>] ? driver_register+0x71/0x150 > [ 23.664021] [<c02c39c3>] ? __pci_register_driver+0x63/0xb0 > [ 23.664021] [<c08ea9eb>] ? alsa_card_riptide_init+0x0/0x1f > [ 23.664021] [<c08ea9ff>] ? alsa_card_riptide_init+0x14/0x1f > [ 23.664021] [<c010111b>] ? do_one_initcall+0x2b/0x180 > [ 23.664021] [<c014f4cb>] ? trace_hardirqs_on+0xb/0x10 > [ 23.664021] [<c01ec557>] ? proc_register+0x117/0x1f0 > [ 23.664021] [<c05c5832>] ? _spin_unlock+0x22/0x30 > [ 23.664021] [<c01ec557>] ? proc_register+0x117/0x1f0 > [ 23.664021] [<c01ec111>] ? __proc_create+0xf1/0x110 > [ 23.664021] [<c01ec739>] ? create_proc_entry+0x59/0xa0 > [ 23.664021] [<c016d8b7>] ? register_irq_proc+0x87/0xa0 > [ 23.664021] [<c016d909>] ? init_irq_proc+0x39/0x50 > [ 23.664021] [<c08ba4f5>] ? kernel_init+0x83/0xdf > [ 23.664021] [<c08ba472>] ? kernel_init+0x0/0xdf > [ 23.664021] [<c010440b>] ? kernel_thread_helper+0x7/0x10 > > If you need more information or need me to test patches, just let me know. > > Greetings, Eric

1 0

[alsa-devel] HDA Analyzer
by Jaroslav Kysela 28 Nov '08

28 Nov '08

Hi HDA developers, I am trying to write a small tool for control and analyze HDA codecs (something like more powerful GUI brother for Takashi's hda-verb). For rapid development, I chose python and pygtk2 gtk bindings. At this time, only basic things are implemented and not all widget types are supported in detail, but routing and volume control is implemented. If you are interested to check this preview and send me ideas to make this tool better, the code is available at URL bellow (or in alsa.git repo - see www.alsa-project.org). http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer Planned features: - support all widget types in detail - allow assign custom names to nodes for better orientation - raw verb r/w ops - save a text output of codec configuration for driver developers (maybe some diff) - run aplay or speaker-test directly from GUI - run arecord directly from GUI with some simple scope to analyze input - sysfs support for actual HDA driver Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 7

[alsa-devel] simplest volume control possible
by John Kacur 28 Nov '08

28 Nov '08

I'm looking at the dapm.txt documentation, and I'm looking at the code in codecs like wm8753.c, and I'm thinking that all of this is tremendous overkill for the very simple codec I have. Basically I don't care about power, the thing can be powered on all the time. I put a one pulse on a gpio for volume up and a zero pulse on a gpio for volume down, 0 - 63 pulses though. Question, then, what would be an appropriate widget to use?

2 2

Re: [alsa-devel] Creative X-Fi Xtreme Audio (CA0110 chip)
by Vedran Miletić 28 Nov '08

28 Nov '08

2008/11/21 Dmitri Seletski <drjoms(a)gmail.com>: > Hi Vedran, once again, could I get experimental drivers for Creative XFI pro > gamer drivers?(link sent to manuel seemed to be broken) > Regards > > On Thu, Nov 20, 2008 at 1:51 PM, Vedran Miletić <rivanvx(a)gmail.com> wrote: >> >> CA0110 HDA mode is supported in this tree: >> >> http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-unstable-2.6.git;a=su… >> CA0111 (PCI version) is supported by snd-ca0106, not sure if CA0106 >> mode can be supported that way too. >> >> 2008/11/19 Manuel C. <manuel(a)ekerazha.com>: >> > Hi, >> > I want to bring to your attention the Creative X-Fi Xtreme Audio card >> > with CA0110 chip, it is also bundled with some MSI motherboards. >> > >> > CA0110 web page: http://www.creative.com/oem/products/chips/CA0110.asp >> > >> > >From that web page: >> > "CA0110 is a PCI Express audio chip using the CA0111 audio mode or High >> > Definition Audio mode. >> > >> > [...] >> > >> > CA0106 mode >> > - Same as original CA0111." >> > >> > So it should be a PCI Express version of the CA0111 chip (and CA0111 >> > could be similar to the CA0106 chip). >> > >> > The CA0110 chip isn't even listed in the Creative products matrix on the >> > wiki. Is a CA0110 driver under development? Thank you. >> > >> > _______________________________________________ >> > Alsa-devel mailing list >> > Alsa-devel(a)alsa-project.org >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> > >> >> >> >> -- >> Vedran Miletić >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > Now I noticed that for some reason link got broken. Anyway, the correct link is: http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-unstable-2.6.git;a=su… -- Vedran Miletić

2 1

[alsa-devel] Changelog wanted for changes from 1.0.17 to 1.0.18
by Nigel Henry 28 Nov '08

28 Nov '08

Is the changelog for changes from alsa driver 1.0.17 to 1.0.18 still available somewhere? Currently I can access the changelog for changes from 1.0.18 to 1.0.18a. I would like to download this info, and save it for future use. How do I do that? Any help most gratefully received. Nigel.

3 2

[alsa-devel] [PATCHv2 0/1] OMAP_TWL4030 SoC Audio driver.
by Stanley.Miao 27 Nov '08

27 Nov '08

Changes from v1: 1, change the name from omap-general to omap_twl4030. 2, proune off board related defines. 3, add a extended interface omap_twl4030_specific_init() to support some board specific features. If some board specific features are added into a board, for example, omap3evm board. we can create a new file omap3evm.c and define a funciton omap_twl4030_specific_init() in it, and select SND_OMAP_TWL4030_SPECIFIC in config SND_OMAP_SOC_OMAP3EVM.

2 4

[alsa-devel] [PATCH 0/3] ASoC updates
by Mark Brown 27 Nov '08

27 Nov '08

The following changes since commit 5c0d7bb797a975691ca8bbc38e53da03c6e151bb: Dmitry Baryshkov (1): ASoC: tosa: move gpio probing to machine callbacks are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Daniel Mack (1): ASoC: Allow more routing features for tlv320aic3x Mark Brown (1): ASoC: Fix word wrapping in OMAP Kconfig Qinghuang Feng (1): ASoC: Clean up kernel-doc for snd_soc_dai_set_fmt sound/soc/codecs/tlv320aic3x.c | 96 +++++++++++++++++++++++++++------------ sound/soc/codecs/tlv320aic3x.h | 12 +++++ sound/soc/omap/Kconfig | 5 +- sound/soc/soc-core.c | 1 - 4 files changed, 80 insertions(+), 34 deletions(-)

2 4

Re: [alsa-devel] [PATCH] sound/pci/mixart/mixart.c: Add missing snd_card_free
by Takashi Iwai 27 Nov '08

27 Nov '08

At Thu, 27 Nov 2008 15:40:40 +0100 (CET), Julia Lawall wrote: > > From: Julia Lawall <julia(a)diku.dk> > > The function snd_mixart_create creates a link between mgr and card that > allows snd_mixart_free to free card as well. But if snd_mixart_create > fails, then the link has not been created and card has to be freed explicitly. > > > The semantic match that finds the problem is as follows: > (http://www.emn.fr/x-info/coccinelle/) (snip) > Signed-off-by: Julia Lawall <julia(a)diku.dk> Thanks, applied now. Takashi

1 0

Re: [alsa-devel] [PATCH] sound/pci/mixart/mixart.c: Add missing snd_card_free
by Takashi Iwai 27 Nov '08

27 Nov '08

At Thu, 27 Nov 2008 15:01:51 +0100 (CET), Julia Lawall wrote: > > On Thu, 27 Nov 2008, Takashi Iwai wrote: > > > At Thu, 27 Nov 2008 14:33:30 +0100 (CET), > > Julia Lawall wrote: > > > > > > Oops, please ignore this patch. There is a problem, but the modification > > > is not correct. > > > > How about the patch below? > > My solution was to move the initialization of mgr->chip[idx] down below > the last error return in snd_mixart_create. I think it depends on what is > going to happen to the structure ops. In your solution, if ops.dev_free > is used later on, then the link between mgr and card will be broken, and > no one will know to free card. > > There is the same code and the same problem in sound/pci/pcxhr/pcxhr.c > > I attach a proposed patch for both below. Looks good. Could you repost with a readable comment? thanks, Takashi > > julia > > > Takashi > > > > --- > > diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c > > index ae7601f..3cccfed 100644 > > --- a/sound/pci/mixart/mixart.c > > +++ b/sound/pci/mixart/mixart.c > > @@ -989,6 +989,7 @@ static int snd_mixart_pcm_digital(struct snd_mixart *chip) > > > > static int snd_mixart_chip_free(struct snd_mixart *chip) > > { > > + chip->mgr->chip[chip->chip_idx] = NULL; > > kfree(chip); > > return 0; > > } > > -- > > To unsubscribe from this list: send the line "unsubscribe kernel-janitors" in > > the body of a message to majordomo(a)vger.kernel.org > > More majordomo info at http://vger.kernel.org/majordomo-info.html > > > > --- > sound/pci/mixart/mixart.c | 4 +++- > sound/pci/pcxhr/pcxhr.c | 4 +++- > 2 files changed, 6 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c > index ae7601f..f23a735 100644 > --- a/sound/pci/mixart/mixart.c > +++ b/sound/pci/mixart/mixart.c > @@ -1010,7 +1010,7 @@ static int __devinit snd_mixart_create(struct mixart_mgr *mgr, struct snd_card * > .dev_free = snd_mixart_chip_dev_free, > }; > > - mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL); > + chip = kzalloc(sizeof(*chip), GFP_KERNEL); > if (! chip) { > snd_printk(KERN_ERR "cannot allocate chip\n"); > return -ENOMEM; > @@ -1025,6 +1025,7 @@ static int __devinit snd_mixart_create(struct mixart_mgr *mgr, struct snd_card * > return err; > } > > + mgr->chip[idx] = chip; > snd_card_set_dev(card, &mgr->pci->dev); > > return 0; > @@ -1377,6 +1378,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, > sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i); > > if ((err = snd_mixart_create(mgr, card, i)) < 0) { > + snd_card_free(card); > snd_mixart_free(mgr); > return err; > } > diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c > index 73de6e9..7d2b136 100644 > --- a/sound/pci/pcxhr/pcxhr.c > +++ b/sound/pci/pcxhr/pcxhr.c > @@ -1024,7 +1024,7 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, > .dev_free = pcxhr_chip_dev_free, > }; > > - mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL); > + chip = kzalloc(sizeof(*chip), GFP_KERNEL); > if (! chip) { > snd_printk(KERN_ERR "cannot allocate chip\n"); > return -ENOMEM; > @@ -1050,6 +1050,7 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, > return err; > } > > + mgr->chip[idx] = chip; > snd_card_set_dev(card, &mgr->pci->dev); > > return 0; > @@ -1310,6 +1311,7 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id > sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i); > > if ((err = pcxhr_create(mgr, card, i)) < 0) { > + snd_card_free(card); > pcxhr_free(mgr); > return err; > } >

1 0

Re: [alsa-devel] [PATCH] sound/pci/mixart/mixart.c: Add missing snd_card_free
by Takashi Iwai 27 Nov '08

27 Nov '08

At Thu, 27 Nov 2008 14:33:30 +0100 (CET), Julia Lawall wrote: > > Oops, please ignore this patch. There is a problem, but the modification > is not correct. How about the patch below? Takashi --- diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index ae7601f..3cccfed 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -989,6 +989,7 @@ static int snd_mixart_pcm_digital(struct snd_mixart *chip) static int snd_mixart_chip_free(struct snd_mixart *chip) { + chip->mgr->chip[chip->chip_idx] = NULL; kfree(chip); return 0; }

1 0

[alsa-devel] [PATCH] OMAP general SOC driver.
by Stanley.Miao 27 Nov '08

27 Nov '08

Add a shared omap SoC driver to avoid reduplicate code among omap soc drivers. Signed-off-by: Stanley.Miao <stanley.miao(a)windriver.com> --- sound/soc/omap/Kconfig | 49 +++++++++++++ sound/soc/omap/Makefile | 13 ++++ sound/soc/omap/omap-general.c | 155 +++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-general.h | 70 ++++++++++++++++++ 4 files changed, 287 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap-general.c create mode 100644 sound/soc/omap/omap-general.h diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 8b7766b..92a557f 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -14,6 +14,14 @@ config SND_OMAP_SOC_N810 help Say Y if you want to add support for SoC audio on Nokia N810. +config SND_OMAP_SOC_OMAP3_BEAGLE + tristate "SoC Audio support for OMAP3 Beagle" + depends on SND_OMAP_SOC && MACH_OMAP3_BEAGLE + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the Beagleboard. + config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" depends on SND_OMAP_SOC && MACH_OMAP_OSK @@ -21,3 +29,44 @@ config SND_OMAP_SOC_OSK5912 select SND_SOC_TLV320AIC23 help Say Y if you want to add support for SoC audio on osk5912. + +config SND_OMAP_SOC_OVERO + tristate "SoC Audio support for Gumstix Overo" + depends on SND_OMAP_SOC && MACH_OVERO + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the Gumstix Overo. + +config SND_OMAP_SOC_3430SDP + tristate "SoC Audio support for OMAP 3430SDP" + depends on SND_OMAP_SOC && MACH_OMAP_3430SDP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP 3430SDP. + +config SND_OMAP_SOC_LDP + tristate "SoC Audio support for OMAP ZOOM SDK" + depends on SND_OMAP_SOC && MACH_OMAP_LDP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP ZOOM MDK. + +config SND_OMAP_SOC_OMAP2EVM + tristate "SoC Audio support for OMAP2EVM" + depends on SND_OMAP_SOC && MACH_OMAP2EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP2EVM. + +config SND_OMAP_SOC_OMAP3EVM + tristate "SoC Audio support for OMAP3EVM" + depends on SND_OMAP_SOC && MACH_OMAP3EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3EVM. + diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index e09d1f2..de9163b 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -7,7 +7,20 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-omap3beagle-objs := omap-general.o snd-soc-osk5912-objs := osk5912.o +snd-soc-overo-objs := omap-general.o +snd-soc-3430sdp-objs := omap-general.o +snd-soc-ldp-objs := omap-general.o +snd-soc-omap2evm-objs := omap-general.o +snd-soc-omap3evm-objs := omap-general.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o +obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o +obj-$(CONFIG_SND_OMAP_SOC_3430SDP) += snd-soc-3430sdp.o +obj-$(CONFIG_SND_OMAP_SOC_LDP) += snd-soc-ldp.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o + diff --git a/sound/soc/omap/omap-general.c b/sound/soc/omap/omap-general.c new file mode 100644 index 0000000..a3fa647 --- /dev/null +++ b/sound/soc/omap/omap-general.c @@ -0,0 +1,155 @@ +/* + * omap-general.c -- OMAP SoC general machine file. + * + * Author: Stanley Miao <stanley.miao(a)windriver.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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "omap-general.h" + +static int general_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_SYS_CLOCK, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops general_ops = { + .startup = SOC_OPS_STARTUP, + .shutdown = SOC_OPS_SHUTDOWN, + .hw_params = general_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link general_dai = { + .name = CODEC_NAME, + .stream_name = STREAM_NAME, + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = CODEC_DAI, + .ops = &general_ops, + .init = DAI_LINK_INIT, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_omap_general = { + .name = "omap_general", + .dai_link = &general_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device general_snd_devdata = { + .machine = &snd_soc_omap_general, + .platform = &omap_soc_platform, + .codec_dev = CODEC_DEV, + .codec_data = CODEC_SETUP_DATA, +}; + +static struct platform_device *general_snd_device; + +static int __init general_soc_init(void) +{ + int ret; + + if (!MACHINE_IS_OMAP_GENERAL) { + pr_debug("Not supported board!\n"); + return -ENODEV; + } + printk(KERN_INFO "OMAP General SoC init\n"); + + general_snd_device = platform_device_alloc("soc-audio", -1); + if (!general_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(general_snd_device, &general_snd_devdata); + general_snd_devdata.dev = &general_snd_device->dev; + *(unsigned int *)general_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(general_snd_device); + if (ret) + goto err1; + ret = omap_board_soc_init(general_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + platform_device_put(general_snd_device); + + return ret; +} +module_init(general_soc_init); + +static void __exit general_soc_exit(void) +{ + omap_board_soc_exit(general_snd_device); + platform_device_unregister(general_snd_device); +} +module_exit(general_soc_exit); + +MODULE_AUTHOR("Stanley Miao <stanley.miao(a)windriver.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP GENERAL"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-general.h b/sound/soc/omap/omap-general.h new file mode 100644 index 0000000..8b9515c --- /dev/null +++ b/sound/soc/omap/omap-general.h @@ -0,0 +1,70 @@ +/* + * omap-general.h + * + * Copyright (C) 2008 Wind River Systems, Inc. + * + * Author: Stanley Miao <stanley.miao(a)windriver.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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifdef CONFIG_SND_OMAP_SOC_N810 +#include "../codecs/tlv320aic3x.h" +#define CODEC_SYS_CLOCK 12000000 +#define CODEC_NAME "TLV320AIC33" +#define STREAM_NAME "AIC33" +#define CODEC_DEV (&soc_codec_dev_aic3x) +#define CODEC_DAI (&aic3x_dai) +#define CODEC_SETUP_DATA (&&n810_aic33_setup) +#define DAI_LINK_INIT (n810_aic33_init) +#define SOC_OPS_STARTUP (n810_startup) +#define SOC_OPS_SHUTDOWN (n810_shutdown) +#else +#include "../codecs/twl4030.h" +#define CODEC_SYS_CLOCK 26000000 +#define CODEC_NAME "TWL4030" +#define STREAM_NAME "TWL4030" +#define CODEC_DEV (&soc_codec_dev_twl4030) +#define CODEC_DAI (&twl4030_dai) +#define CODEC_SETUP_DATA NULL +#define DAI_LINK_INIT NULL +#define SOC_OPS_STARTUP NULL +#define SOC_OPS_SHUTDOWN NULL +#endif /* CONFIG_SND_OMAP_SOC_N810 */ + +#ifdef CONFIG_SND_OMAP_SOC_N810 +#define MACHINE_IS_OMAP_GENERAL (machine_is_nokia_n810() || machine_is_nokia_n810_wimax()) +#elif defined(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) +#define MACHINE_IS_OMAP_GENERAL machine_is_omap3_beagle() +#elif defined(CONFIG_SND_OMAP_SOC_OVERO) +#define MACHINE_IS_OMAP_GENERAL machine_is_overo() +#elif defined(CONFIG_SND_OMAP_SOC_3430SDP) +#define MACHINE_IS_OMAP_GENERAL machine_is_omap_3430sdp() +#elif defined(CONFIG_SND_OMAP_SOC_LDP) +#define MACHINE_IS_OMAP_GENERAL machine_is_omap_ldp() +#elif defined(CONFIG_SND_OMAP_SOC_OMAP2EVM) +#define MACHINE_IS_OMAP_GENERAL machine_is_omap2evm() +#elif defined(CONFIG_SND_OMAP_SOC_OMAP3EVM) +#define MACHINE_IS_OMAP_GENERAL machine_is_omapevm() +#endif + +#ifndef omap_board_soc_init +#define omap_board_soc_init(x) 0 +#endif +#ifndef omap_board_soc_exit +#define omap_board_soc_exit +#endif + -- 1.5.6.3

8 15

Re: [alsa-devel] [PATCH] OMAP general SOC driver.
by Mark Brown 27 Nov '08

27 Nov '08

On Thu, Nov 27, 2008 at 12:42:12PM +0200, Grazvydas Ignotas wrote: > Pandora board uses different master/slave configuration for McBSP bus, > compared to most other boards. It also has another audio path (in > addition to TWL4030) - another McBSP link connected to external DAC. > The DAC needs 256*Fs clock from TWL4030 to function, and is used for > main audio output. TWL4030 is to be used for 2 MIC + line inputs only, > and has no outputs connected. This definitely needs to be a separate machine driver. > I have main audio path working (McBSP2 to DAC), but not inputs (McBSP4 > to TWL4030). How can this be implemented using current framework? > Should I register 2 platform devices? Sorry for possible derail or > stupid questions. This should all be one ASoC device, especially given that the fact that all the paths use the TWL4030 somehow. Have a look at how, eg, the neo1973 driver handles the bluetooth. For now there's not *particularly* sane support for multiple codecs. Does the DAC have any software control? If not it should be fairly straightforward to handle.

1 0

[alsa-devel] Alsa Driver Development - General Concepts
by karl＠lil-voodoo.co.uk 27 Nov '08

27 Nov '08

Hi, I've got a couple of audio/midi devices knocking around that I'd quite like to add ALSA support for (for use in Linux sequencers/DAWs) so I've been reading up on Linux device drivers and Alsa driver development. I do have a programming background but unfortunately, I'm a little confused on a couple of points and I was hoping someone wouldn't mind helping me out. I'm kind of basing my thoughts on the idea that I have a USB device which is not recognised by the kernel and does not expose a known midi interface (so requires a completely new Alsa driver?). The goal would be for this device to be usable in Linux midi sequencers. - How does the Alsa driver communicate with the hardware? Would I need to write a Kernel driver (e.g. a character device) for the Alsa driver to talk to? - Would my Alsa driver basically create a card instance and then a RawMidi component? I'm assuming the use of RawMidi would then allow applications to recognise the Alsa driver as a seemingly ordinary midi port??? Then implement _input_trigger() and _output_trigger() to exchange data with the application, whilst passing data to/from the kernel driver to talk to the hardware? Sorry if I'm way off base. I'm trying to get in on this thing to help spread some extra Linux love in music world. Many thanks.

4 3

[alsa-devel] [PATCH] cleanup controllers in snd-usb-caiaq
by Daniel Mack 27 Nov '08

27 Nov '08

Hi, this patch does some code cleanup in snd-usb-caiaq. No functional change whatsoever. Version number bumped to 1.3.9. Best regards, Daniel

2 5

[alsa-devel] Alsa driver won't compile on Ubuntu kernel 2.6.24-16-generic
by Klaus Schulz 27 Nov '08

27 Nov '08

Hi there. I issued this TT some time ago. Could anybody have a look into it - please! There a quite some people running Ubuntu and still 2.6.24 kernels, who'd love to get an ALSA upgrade in place THX Klaus

2 2

[alsa-devel] [PATCH] Allow more routing features for tlv320aic3x
by Daniel Mack 26 Nov '08

26 Nov '08

Hi, this patch adds more routing possibilities to the tlv320aic3x code driver. Stereo mixers are, as dicussed earlier, not appropriate due to the versatile channel functions of this rather complex chip. Best regards, Daniel

2 6

Re: [alsa-devel] Current state of the XFi driver?
by Takashi Iwai 26 Nov '08

26 Nov '08

At Wed, 26 Nov 2008 10:02:54 -0600, William Zumwalt wrote: > > So, the group was working on an alsa driver for the XFi, then CL released > another driver for linux IIRC, in Nov. ... did the alsa group stop working on > this driver and now we should go back to trying to get the CL drivers to work? > > Does one work better than the other currently? Or is there a recommended > driver between the two? Honestly, I don't care at all if anyone will continue working on my previous driver code, but I have no interest right now... Takashi

1 0

[alsa-devel] Bug in sound-unstable-2.6 ice1724
by Vedran Miletić 26 Nov '08

26 Nov '08

I have: 01:0a.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01) Subsystem: TERRATEC Electronic GmbH PHASE 22 Flags: bus master, medium devsel, latency 64, IRQ 16 I/O ports at c880 [size=32] I/O ports at c800 [size=128] Capabilities: [80] Power Management version 1 Kernel driver in use: ICE1724 Kernel modules: snd-ice1724 I do this: mplayer -ao alsa:device=hw=1.0 -srate 44100 -format s32le get_video Before the song ends, last sample loops for about 20-30 times, and then mplayer quits with error: alsa-lib: pcm_hw.c:488:(snd_pcm_hw_drain) SNDRV_PCM_IOCTL_DRAIN failed: Input/output error This doesn't happen on sound-2.6 and doesn't happen on sound-unstable-2.6 with E-mu 1212m. -- Vedran Miletić

3 27

[alsa-devel] Creative X-Fi driver integration
by The Source 26 Nov '08

26 Nov '08

Hi, is it possible to include Creative driver into alsa drivers package, so it will be available right after kernel installation without additional compiling? Besides, you can react faster to kernel changes and make driver compatible with latest kernels when necessary, if you include it.

4 4

[alsa-devel] floating point exception triggered by speaker-test on pc-speaker
by Alexander E. Patrakov 26 Nov '08

26 Nov '08

Hello, I have hit a strange bug. To reproduce: 1. Build snd-pcsp.ko. 2. alsamixer -c pcsp, set the base frequency to 37286 Hz 3. $ speaker-test -r 22050 -P 4 -D plughw:pcsp speaker-test 1.0.18 Playback device is plughw:pcsp Stream parameters are 22050Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 22050Hz (requested 22050Hz) Buffer size range from 37 to 38756 Period size range from 18 to 19379 Using max buffer size 38756 Periods = 4 was set period_size = 0 was set buffer_size = 38756 0 - Front Left Floating point exception I.e., as you see, period size somehow became 0. While debugging this with printfs, I noticed that the interval for periods in snd_pcm_hw_param_set_near() starts with 4, but doesn't include it (i.e., is open). Also, I found that snd_interval_refine_first() and snd_interval_refine_last(), when given an open interval, refine it to something like (n n+1). This is obviously wrong, as there are no integer points in this interval. Changing i->max++ and i->min-- there to adjust the value by 2 did eliminate the floating point exception, but replaced it with "Unable to set nperiods 4 for playback: Invalid argument". So there must be something else also wrong, but I can't pinpoint it and give you a ready-made patch :( -- Alexander E. Patrakov

2 1

[alsa-devel] Query for running OSS application on ALSA
by Saxena, Anshuman 26 Nov '08

26 Nov '08

Hi, I have been using OSS drivers and applications. Now I have moved to ALSA SoC framework supported in the kernel. I have got the same OSS application running on this ALSA SoC framework but I have the following observation: When I run the OSS application to playback some audio through ALSA SoC framework, and I send CTRL+C signal to terminate the application, I see a delay of around 3-4 sec before the application really close. I traced and found out that it takes lot of time in snd_pcm_drain in pcm_native.c . But if I run some ALSA application, I do not see this delay in terminating the application. Is there some known issue with OSS emulation of ALSA SoC framework? I feel that some buffers are being drained out to the audio codec and hence this time is consumed. Is their some configuration for the buffer size? Can I directly stop the data transfer to codec and discard the remaining data? Your help is highly appreciated. Thanks and Regards, Anshuman

1 0

[alsa-devel] [PATCH 1/2] sound/pci/hda/hda_codec.c: cleanup kernel-doc
by Qinghuang Feng 26 Nov '08

26 Nov '08

There is no argument named @state in snd_hda_resume, remove its' comment. Signed-off-by: Qinghuang Feng <qhfeng.kernel(a)gmail.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ba1ab73..979f34a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3109,7 +3109,6 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state) /** * snd_hda_resume - resume the codecs * @bus: the HDA bus - * @state: resume state * * Returns 0 if successful. *

2 1

[alsa-devel] Intel HDA and FEDORA 9 with 2.6.27.4 kernel
by Pilipovic, Josip 26 Nov '08

26 Nov '08

Hi All ... I am running Fedora 9 with 2.6.27.4 kernel. I have download and instaled ALSA 1.0.18. Card is EUROTECH INC CATALYST with Intel ATOM processor and Intel System Controler Hub US15W. Attached is output from alsa-info.sh and /proc/asound/card0. I have run bunch of different voice test but no sound on either output on this card. I have compile alsa library pcm test and run on this board no sound. I have activated and added some dignostic to hda controler code

2 1

[alsa-devel] CVS, git, or something else?
by Keith A. Milner 26 Nov '08

26 Nov '08

I've been trying to check out the latest alsa-driver source to do some work on it. I followed the instructions at: http://www.alsa-project.org/main/index.php/GIT_Server#Occasional_Developers But when I cd alsa-drivers and gitcompile, it fails: make[1]: Entering directory `/homedirs/kamilner/Development/Workspace/alsa/alsa-driver/acore' copying file alsa-kernel/core/init.c patching file init.c Hunk #2 succeeded at 280 (offset 3 lines). Hunk #3 succeeded at 306 (offset 3 lines). Hunk #4 FAILED at 568. 1 out of 4 hunks FAILED -- saving rejects to file init.c.rej make[1]: *** [init.c] Error 1 make[1]: Leaving directory `/homedirs/kamilner/Development/Workspace/alsa/alsa-driver/acore' make: *** [dep] Error 1 Do I have the right version? Am I going about this the right way? I see a lot of posts on this list talking about CVS but none mentioning git. Maybe I should be using CVS instead? -- Keith A. Milner

3 10

[alsa-devel] Intel HDA and FEDORA 9 with 2.6.27.4 kernel
by Pilipovic, Josip 26 Nov '08

26 Nov '08

Hi All ... I am running Fedora 9 with 2.6.27.4 kernel. I have download and instaled ALSA 1.0.18. Card is EUROTECH INC CATALYST with Intel ATOM processor and Intel System Controler Hub US15W. Attached is output from alsa-info.sh and /proc/asound/card0. I have run bunch of different voice test but no sound on either output on this card. I have compile alsa library pcm test and run on this board no sound. I have activated and added some dignostic to hda controler code Here is print out from /var/log/messages Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:822: hda_codec_setup_stream: NID=0x11, stream=0x3, channel=0, format=0x4011 Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1434: azx_pcm_prepare: bufsize=0x4400, format=0x4011 Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1443: JP:azx_pcm_prepare: fifosize=0xc0 Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:822: hda_codec_setup_stream: NID=0x10, stream=0x3, channel=0, format=0x4011 Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:822: hda_codec_setup_stream: NID=0x11, stream=0x3, channel=0, format=0x4011 Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1457: JP:azx_pcm_trigger: fifosize=0x1 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1457: JP:azx_pcm_trigger: fifosize=0x0 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x10 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x11 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x10 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x11 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x10 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x11

1 0

[alsa-devel] via82xx: unreliable audio, strange mixer behavior
by Forest Bond 25 Nov '08

25 Nov '08

Hi, I'm using snd-via82xx on an Ubuntu 8.04 system. alsa-info.sh output is here: http://www.alsa-project.org/db/?f=1c3d5570e7b57d4d2f79b964a323dd7b4efd7e78 Here are the symptoms: Audio goes away after a semi-random interval. Sometimes it returns after `modprobe -r snd-via82xx' followed by `modprobe snd-via82xx', but sometimes it seems to take a full reboot. This looks similar to: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3D3194 Some folks think that this kind of issue is triggered by audio volume being above a certain level. I'm not entirely sure if this is true, but I think it is. One time, I apparently triggered the issue by unplugging and replugging the audio cable from the mainboard during playback. Audio can be heard even when the PCM volume level is 0 (as told by alsamixer). The volume does get louder when I turn up that mixer control, but it never goes away entirely when I turn the volume to zero. Mute works as expected. This issue is impacting a significant project for me and my company. I'm willing to spend as much time as it takes to debug, and I'm fairly technical. I can also offer a bounty of some sort, if that interests anyone. Can anyone offer any input? Thanks, Forest -- Forest Bond http://www.alittletooquiet.net http://www.pytagsfs.org

2 8

[alsa-devel] [PATCH] ASoC: Fix TW4030 Kconfig dependency
by arunks＠mistralsolutions.com 25 Nov '08

25 Nov '08

From: Arun KS <arunks(a)mistralsolutions.com> Fixes Kconfig dependency of TWL4030 audio codec driver with TWL4030 core driver on both overo and omap2evm boards Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/Kconfig | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 6c56277..0900437 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -24,7 +24,7 @@ config SND_OMAP_SOC_OSK5912 config SND_OMAP_SOC_OVERO tristate "SoC Audio support for Gumstix Overo" - depends on SND_OMAP_SOC && MACH_OVERO + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help @@ -32,7 +32,7 @@ config SND_OMAP_SOC_OVERO config SND_OMAP_SOC_OMAP2EVM tristate "SoC Audio support for OMAP2EVM board" - depends on SND_OMAP_SOC && MACH_OMAP2EVM + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP2EVM select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help -- 1.5.6.3

3 5

[alsa-devel] [PATCH 00/13] ASoC updates
by Mark Brown 25 Nov '08

25 Nov '08

The following changes since commit 4f904735c809e44c11f57cd4f82446aac1243e0e: Mark Brown (1): ALSA: ASoC: Fix typo in snd_soc_card update documentation are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Arun KS (1): ASoC: Fix TWL4030 Kconfig dependency Dmitry Baryshkov (1): ASoC: tosa: move gpio probing to machine callbacks Jarkko Nikula (2): ASoC: OMAP: Apply channel constrains to N810 machine driver ASoC: OMAP: Add support for mono audio links in McBSP DAI Mark Brown (4): ASoC: Use supplied DAI for WM9713 rather than substream ASoC: Remove DAI type information ASoC: Flag AD1980 as an AC97 interface ASoC: Lower priority of resume work logging Misael Lopez Cruz (1): ASoC: Add support for TI SDP3430 Peter Ujfalusi (4): ASoC: TWL4030: Disable soft-volume ASoC: TWL4030: Change the Master volume control to TLV ASoC: TWL4030: Add CGAIN volume control ASoC: TWL4030: Add helper function for output gain controls include/sound/soc-dai.h | 2 +- include/sound/soc.h | 8 -- sound/soc/atmel/atmel_ssc_dai.c | 3 - sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 1 - sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/blackfin/bf5xx-i2s.c | 1 - sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1980.c | 1 + sound/soc/codecs/pcm3008.c | 1 - sound/soc/codecs/twl4030.c | 180 +++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 18 +--- sound/soc/davinci/davinci-i2s.c | 1 - sound/soc/fsl/mpc5200_psc_i2s.c | 1 - sound/soc/omap/Kconfig | 11 ++- sound/soc/omap/Makefile | 3 +- sound/soc/omap/n810.c | 4 + sound/soc/omap/omap-mcbsp.c | 24 +++-- sound/soc/omap/sdp3430.c | 152 ++++++++++++++++++++++++++++++ sound/soc/pxa/pxa-ssp.c | 4 - sound/soc/pxa/pxa2xx-ac97.c | 6 +- sound/soc/pxa/pxa2xx-i2s.c | 1 - sound/soc/pxa/tosa.c | 30 +++++-- sound/soc/s3c24xx/neo1973_wm8753.c | 1 - sound/soc/s3c24xx/s3c2412-i2s.c | 1 - sound/soc/s3c24xx/s3c2443-ac97.c | 4 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 1 - sound/soc/sh/hac.c | 4 +- sound/soc/sh/ssi.c | 2 - sound/soc/soc-core.c | 39 +++------ 31 files changed, 410 insertions(+), 102 deletions(-) create mode 100644 sound/soc/omap/sdp3430.c

2 14

[alsa-devel] [PATCH 2/2] sound/soc/soc-core.c: cleanup kernel-doc
by Qinghuang Feng 25 Nov '08

25 Nov '08

There is no argument named @clk_id in snd_soc_dai_set_fmt, remove its' comment. Signed-off-by: Qinghuang Feng <qhfeng.kernel(a)gmail.com> --- diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16c7453..81ee4ba 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1805,7 +1805,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI - * @clk_id: DAI specific clock ID * @fmt: SND_SOC_DAIFMT_ format value. * * Configures the DAI hardware format and clocking.

2 1

[alsa-devel] [PATCH] [ASOC] tosa: move gpio probing to machine callbacks
by Dmitry Baryshkov 25 Nov '08

25 Nov '08

Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/tosa.c | 30 ++++++++++++++++++++++-------- 1 files changed, 22 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index afefe41..69ca79b 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -230,10 +230,32 @@ static struct snd_soc_dai_link tosa_dai[] = { }, }; +static int tosa_probe(struct platform_device *dev) +{ + int ret; + + ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack"); + if (ret) + return ret; + ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0); + if (ret) + gpio_free(TOSA_GPIO_L_MUTE); + + return ret; +} + +static int tosa_remove(struct platform_device *dev) +{ + gpio_free(TOSA_GPIO_L_MUTE); + return 0; +} + static struct snd_soc_machine tosa = { .name = "Tosa", .dai_link = tosa_dai, .num_links = ARRAY_SIZE(tosa_dai), + .probe = tosa_probe, + .remove = tosa_remove, }; static struct snd_soc_device tosa_snd_devdata = { @@ -251,11 +273,6 @@ static int __init tosa_init(void) if (!machine_is_tosa()) return -ENODEV; - ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack"); - if (ret) - return ret; - gpio_direction_output(TOSA_GPIO_L_MUTE, 0); - tosa_snd_device = platform_device_alloc("soc-audio", -1); if (!tosa_snd_device) { ret = -ENOMEM; @@ -272,15 +289,12 @@ static int __init tosa_init(void) platform_device_put(tosa_snd_device); err_alloc: - gpio_free(TOSA_GPIO_L_MUTE); - return ret; } static void __exit tosa_exit(void) { platform_device_unregister(tosa_snd_device); - gpio_free(TOSA_GPIO_L_MUTE); } module_init(tosa_init); -- 1.5.6.5

2 1

[alsa-devel] [PATCH] ALSA: SoC: Add support for TI SDP3430
by Misael Lopez Cruz 25 Nov '08

25 Nov '08

This patch add ASoC support for TI SDP3430. It's based on Gumstix Overo SoC code by Steve Sakoman. Signed-off-by: Misael Lopez Cruz <mesak82(a)gmail.com> --- sound/soc/omap/Kconfig | 7 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp3430.c | 152 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 161 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/sdp3430.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 0daeee4..e816fa4 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -38,4 +38,11 @@ config SND_OMAP_SOC_OVERO help Say Y if you want to add support for SoC audio on the Gumstix Overo. +config SND_OMAP_SOC_SDP3430 + tristate "SoC Audio support for Texas Instruments SDP3430" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 4bae404..64e560b 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -10,9 +10,11 @@ snd-soc-n810-objs := n810.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o +snd-soc-sdp3430-objs := sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o +obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c new file mode 100644 index 0000000..85fd160 --- /dev/null +++ b/sound/soc/omap/sdp3430.c @@ -0,0 +1,152 @@ +/* + * sdp3430.c -- SoC audio for TI OMAP3430 SDP + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * Based on: + * Author: Steve Sakoman <steve(a)sakoman.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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int sdp3430_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops sdp3430_ops = { + .hw_params = sdp3430_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sdp3430_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &sdp3430_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_sdp3430 = { + .name = "SDP3430", + .dai_link = &sdp3430_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device sdp3430_snd_devdata = { + .machine = &snd_soc_machine_sdp3430, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *sdp3430_snd_device; + +static int __init sdp3430_soc_init(void) +{ + int ret; + + if (!machine_is_omap_3430sdp()) { + pr_debug("Not SDP3430!\n"); + return -ENODEV; + } + printk(KERN_INFO "SDP3430 SoC init\n"); + + sdp3430_snd_device = platform_device_alloc("soc-audio", -1); + if (!sdp3430_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata); + sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev; + *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(sdp3430_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(sdp3430_snd_device); + + return ret; +} +module_init(sdp3430_soc_init); + +static void __exit sdp3430_soc_exit(void) +{ + platform_device_unregister(sdp3430_snd_device); +} +module_exit(sdp3430_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC SDP3430"); +MODULE_LICENSE("GPL"); + -- 1.5.6.3

2 1

[alsa-devel] [topic/asoc][PATCH 0/2] ASoC: OMAP: Add support for mono audio links in McBSP DAI (take #2)
by Jarkko Nikula 25 Nov '08

25 Nov '08

Hi Second version adding support for mono audio links in OMAP McBSP DAI. At this time, the channels_min is defined to 1 in omap-mcbsp.c and no hacks in machine drivers wanting to use mono link are required. If channel restrictions are required, then machine driver can use snd_pcm_hw_constraint_minmax. Now only N810 machine driver is affected since all other OMAP machine drivers are using TWL4030 or TLV320AIC23 codecs which all claim to support only 2 channels so the link remains always stereo even playing mono audio. -- Jarkko

2 3

[alsa-devel] pcm_prepare, and pcm_trigger
by John Kacur 25 Nov '08

25 Nov '08

More ASoC fundamentals questions. when doing cat 8k8bitpcm.wav > /dev/dsp I'm seeing neither myplatform_pcm_prepare, nor myplatform_pcm_trigger called. Is this normal? Who calls these functions? Thanks

2 3

[alsa-devel] problem with scheduling alsa events
by nescivi 25 Nov '08

25 Nov '08

Hiho, In the Midi output code of this opensource project I am working on, there is a problem with the timing when alsa events are scheduled in the future. The events take much longer to arrive (sometimes almost minutes later!). Below is the code snippet. Can anyone see what is wrong in here? Let me know if you need more code... This is with libasound2 1.0.16-2 (Debian unstable). sincerely, Marije int SC_AlsaMidiClient::sendEvent(int outputIndex, int uid, snd_seq_event_t* evt, float late) { snd_seq_real_time time; if ((outputIndex < 0) || (outputIndex >= mNumOutPorts)) return errIndexOutOfRange; snd_seq_ev_set_source(evt, mOutPorts[outputIndex]); if (uid == 0) { // send to all subscribed ports snd_seq_ev_set_subs(evt); } else { // send to specific port int cid, pid; SC_AlsaParseUID(uid, cid, pid); snd_seq_ev_set_dest(evt, cid, pid); } // late is the latency value in seconds in floats if (late > 0.f) { // old time calculation: // time.tv_sec = (unsigned)(late * 1.0e-6f); // time.tv_nsec = (unsigned)(late * 1.0e3f); // new time calculation. The old one was not correct time.tv_sec = (unsigned)(late); // seconds time.tv_nsec = (unsigned)(late * 1.0e9f) % 1000000000; // nanoseconds // should these be (long) instead??? } else { time.tv_sec = time.tv_nsec = 0; } post("MIDI (ALSA): sending event, time %i, %i, late %f\n", time.tv_sec, time.tv_nsec, late); snd_seq_ev_schedule_real(evt, mQueue, 1, &time); snd_seq_event_output_direct(mHandle, evt); return errNone; }

1 0

[alsa-devel] MSI 7260 patch
by Страхиња Радић 25 Nov '08

25 Nov '08

Hi, I submitted an issue with alsa-driver 1.0.18rc3 (see https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4229). Raymond advised me to send the patch, so everybody with the same motherboard could get automatic recognition of sound card model. And here it is. I just don't know if I got it right. Are the hexadecimal values ok? SND_PCI_QUIRK(0x1462, 0x7260, "MSI 7260", ALC883_TARGA_DIG), I got this from alsainfo: !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC883 Address: 0 Vendor Id: 0x10ec0883 Subsystem Id: 0x14627260 Revision Id: 0x100002 No Modem Function Group found So, I guess 0x1462 0x7260 is ok, right? If that is so, I have attached the diff with this message. Cheers, Strahinya -- _ _ ______________________________________________________ /(_____)\ | ____ __ __ __ __ | \_/~ ~\_/ |/ __// /_/ // // / http://www.gnu.org/home.sr.html | (\'_'/) / / / __ /_\ ' / http://www.stallman.org | (o_o) /_/ /_/ /_//____/ http://alas.matf.bg.ac.yu/~mr99164 | |______________________________________________________| --- a/sound/pci/hda/patch_realtek.c 2008-09-09 18:55:25.000000000 +0200 +++ b/sound/pci/hda/patch_realtek.c 2008-11-24 17:56:31.000000000 +0100 @@ -8320,6 +8320,7 @@ SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x7260, "MSI 7260", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),

2 1

[alsa-devel] [PATCH] ASoC: Add support for OMAP3 EVM
by Anuj Aggarwal 25 Nov '08

25 Nov '08

This patch adds ALSA SoC support for OMAP3 EVM using TWL4030 audio codec. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Kconfig | 7 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3evm.c | 147 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 156 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3evm.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 6c56277..c465139 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -38,5 +38,12 @@ config SND_OMAP_SOC_OMAP2EVM help Say Y if you want to add support for SoC audio on the omap2evm board. +config SND_OMAP_SOC_OMAP3EVM + tristate "SoC Audio support for OMAP3EVM board" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the omap3evm board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index f5da3cc..bbb3a77 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -10,9 +10,11 @@ snd-soc-n810-objs := n810.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o +snd-soc-omap3evm-objs := omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c new file mode 100644 index 0000000..570af55 --- /dev/null +++ b/sound/soc/omap/omap3evm.c @@ -0,0 +1,147 @@ +/* + * omap3evm.c -- ALSA SoC support for OMAP3 EVM + * + * Author: Anuj Aggarwal <anuj.aggarwal(a)ti.com> + * + * Based on sound/soc/omap/beagle.c by Steve Sakoman + * + * Copyright (C) 2008 Texas Instruments, Incorporated + * + * 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 version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, + * whether express or implied; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap3evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3evm_ops = { + .hw_params = omap3evm_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3evm_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_omap3evm = { + .name = "omap3evm", + .dai_link = &omap3evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3evm_snd_devdata = { + .machine = &snd_soc_machine_omap3evm, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3evm_snd_device; + +static int __init omap3evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap3evm()) { + pr_debug("Not OMAP3 EVM!\n"); + return -ENODEV; + } + pr_info("OMAP3 EVM SoC init\n"); + + omap3evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata); + omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev; + *(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(omap3evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3evm_snd_device); + + return ret; +} + +static void __exit omap3evm_soc_exit(void) +{ + platform_device_unregister(omap3evm_snd_device); +} + +module_init(omap3evm_soc_init); +module_exit(omap3evm_soc_exit); + +MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); +MODULE_LICENSE("GPL"); -- 1.5.6

3 2

[alsa-devel] snd-usb-audio: minimum period time as module option?
by Klaus Schulz 25 Nov '08

25 Nov '08

Hi there. I wrote a small patch which introduces an usb-audio module option called uperiod. Changing to lower values will improve sound quality in some cases. Please comment. THX Klaus --- alsa-driver-1.0.18a/sound/usb/usbaudio.c.orig 2008-11-21 19:19:11.000000000 +0100 +++ alsa-driver-1.0.18a/sound/usb/usbaudio.c 2008-11-24 13:40:12.000000000 +0100 @@ -72,6 +72,7 @@ static int nrpacks = 8; /* max. number static int async_unlink = 1; static int device_setup[SNDRV_CARDS]; /* device parameter for this card*/ static int ignore_ctl_error; +static int uperiod = 125; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); @@ -92,6 +93,8 @@ MODULE_PARM_DESC(device_setup, "Specific module_param(ignore_ctl_error, bool, 0444); MODULE_PARM_DESC(ignore_ctl_error, "Ignore errors from USB controller for mixer interfaces."); +module_param(uperiod, int, 0644); +MODULE_PARM_DESC(uperiod, "Max. size of period per URB."); /* * debug the h/w constraints @@ -108,6 +111,8 @@ MODULE_PARM_DESC(ignore_ctl_error, #define MAX_URBS 8 #define SYNC_URBS 4 /* always four urbs for sync */ #define MIN_PACKS_URB 1 /* minimum 1 packet per urb */ +#define MAX_UPERIOD 1000 +#define MIN_UPERIOD 64 struct audioformat { struct list_head list; @@ -1919,7 +1924,7 @@ static int setup_hw_info(struct snd_pcm_ * in the current code assume the 1ms period. */ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - 1000 * MIN_PACKS_URB, + uperiod * MIN_PACKS_URB, /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX); err = check_hw_params_convention(subs); @@ -3767,6 +3772,11 @@ static int __init snd_usb_audio_init(voi printk(KERN_WARNING "invalid nrpacks value.\n"); return -EINVAL; } + + if (uperiod < MIN_UPERIOD || uperiod > MAX_UPERIOD) { + printk(KERN_WARNING "invalid uperiod value.\n"); + return -EINVAL; + } return usb_register(&usb_audio_driver); }

1 0

[alsa-devel] [PATCH 0/9] ASoC: TWL4030: Add mixer controls for outputs
by Peter Ujfalusi 25 Nov '08

25 Nov '08

Hello, I have tried to clean up the changes that I have at the momnent. After the series you will have 'ultimate' control on how the audio is routed to the outputs of TWL. One can select the possible inputs (for the output) on all cahnnels. Also I have added gain (volume) controls, wherever it is possible, but they are working for both channels in case of stereo outputs. I tried to name the controls to match with the output's name, so you can easily tell which pin(s) the control will touch. The capture part is not touched, yet. I have not included the changes regarding to the SND_SOC_DAPM_OUTPUT() things. So you have one (actually two) outputs and you can use the mixer interface to select where the audio actually routed. The next step is to make correct dapm mapping for the TWL, but that will take some time. Please comment, if I'm doing something that I should not. --- Peter Ujfalusi (9): ASoC: TWL4030: Change the Master volume control to TLV ASoC: TWL4030: Add CGAIN volume control ASoC: TWL4030: Add helper function for mux handling. ASoC: TWL4030: Add helper function for output gain controls ASoC: TWL4030: Add PreDriv outupt mux and volume controls ASoC: TWL4030: Add Headset outupt mux and volume controls ASoC: TWL4030: Add Carkit outupt mux and volume controls ASoC: TWL4030: Add Hands-free outupt mux control ASoC: TWL4030: Add Earpiece outupt mux and volume controls sound/soc/codecs/twl4030.c | 298 +++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 296 insertions(+), 2 deletions(-)

2 16

[alsa-devel] Infrasonic Quartet
by Vedran Miletić 25 Nov '08

25 Nov '08

Auzen started to distribute one cool new Envy24HT-S card, and it looks really interesting to me. Specs are awesome, and it also has word clock on card! http://www.auzentech.com/site/products/infra_quartet.php http://www.infra-sonic.com/site/p_PCI_quartet.php http://www.newegg.com/Product/Product.aspx?Item=N82E16829156007 According to envy24.svobodno.com, it has: Infrasonic Quartet 0x3030 0x4953 Envy24HT-S - supported by ALSA AK4620B (ADC / DAC) - not sure, probably not supported? AK4112 - not sure either? However it has Xilinx on card in similar way as ESI Juli@ does, so it's probably not a standard Envy24HT-S card. Any ideas if this card could be supported? -- Vedran Miletić

3 7

[alsa-devel] [topic/asoc][RFC 0/1] ASoC: OMAP: Add support for mono link configuration to McBSP DAI
by Jarkko Nikula 25 Nov '08

25 Nov '08

Hi I'm trying to cover also pure mono codecs like bluetooth audio with McBSP DAI driver. Protocol part is trivial like patch here but interfacing with machine driver is somewhat question mark. With this patch, machine driver needs to "hack" with cpu_dai in its init function: cpu_dai->playback.channels_min = 1; cpu_dai->playback.channels_max = 1; cpu_dai->capture.channels_min = 1; cpu_dai->capture.channels_max = 1; Another idea is to introduce new omap_mcbsp_dai_mono[] array in omap-mcbsp.c and initialize those with modified builder macro, e.g: -#define OMAP_MCBSP_DAI_BUILDER(link_id) \ +#define OMAP_MCBSP_DAI_BUILDER(link_id, min_ch, max_ch) \ { \ .name = "omap-mcbsp-dai-"#link_id, \ .id = (link_id), \ .type = SND_SOC_DAI_I2S, \ .playback = { \ - .channels_min = 2, \ - .channels_max = 2, \ + .channels_min = (min_ch), \ + .channels_max = (max_ch), \ ... struct snd_soc_dai omap_mcbsp_dai[] = { - OMAP_MCBSP_DAI_BUILDER(0), - OMAP_MCBSP_DAI_BUILDER(1), + OMAP_MCBSP_DAI_BUILDER(0, 2, 2), + OMAP_MCBSP_DAI_BUILDER(1, 2, 2), #if NUM_LINKS >= 3 ... and +struct snd_soc_dai omap_mcbsp_dai_mono[] = { + OMAP_MCBSP_DAI_BUILDER(0, 1, 1), + OMAP_MCBSP_DAI_BUILDER(1, 1, 1), +#if NUM_LINKS >= 3 ... but this looks a bit overkill if typical configuration has only one such a mono link and might confuse if a machine driver still needs to reference those arrays using sequential array index like: .cpu_dai = &omap_mcbsp_dai[0], and .cpu_dai = &omap_mcbsp_dai_mono[1], sequential since if I like that cpu_dai->id is also sequential for links. Opinnions or better ideas :-) -- Jarkko

2 5

[alsa-devel] [PATCH] ALSA: SoC: Add support for TI SDP3430
by Misael Lopez Cruz 25 Nov '08

25 Nov '08

This patch add ASoC support for TI SDP3430. It's based on Gumstix Overo SoC code by Steve Sakoman. Signed-off-by: Misael Lopez Cruz <mesak82(a)gmail.com> --- sound/soc/omap/Kconfig | 7 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp3430.c | 152 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 161 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/sdp3430.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 0daeee4..e816fa4 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -38,4 +38,11 @@ config SND_OMAP_SOC_OVERO help Say Y if you want to add support for SoC audio on the Gumstix Overo. +config SND_OMAP_SOC_SDP3430 + tristate "SoC Audio support for Texas Instruments SDP3430" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 4bae404..64e560b 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -10,9 +10,11 @@ snd-soc-n810-objs := n810.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o +snd-soc-sdp3430-objs := sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o +obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c new file mode 100644 index 0000000..85fd160 --- /dev/null +++ b/sound/soc/omap/sdp3430.c @@ -0,0 +1,152 @@ +/* + * sdp3430.c -- SoC audio for TI OMAP3430 SDP + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * Based on: + * Author: Steve Sakoman <steve(a)sakoman.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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int sdp3430_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops sdp3430_ops = { + .hw_params = sdp3430_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sdp3430_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &sdp3430_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_sdp3430 = { + .name = "SDP3430", + .dai_link = &sdp3430_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device sdp3430_snd_devdata = { + .machine = &snd_soc_machine_sdp3430, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *sdp3430_snd_device; + +static int __init sdp3430_soc_init(void) +{ + int ret; + + if (!machine_is_omap_3430sdp()) { + pr_debug("Not SDP3430!\n"); + return -ENODEV; + } + printk(KERN_INFO "SDP3430 SoC init\n"); + + sdp3430_snd_device = platform_device_alloc("soc-audio", -1); + if (!sdp3430_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata); + sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev; + *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(sdp3430_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(sdp3430_snd_device); + + return ret; +} +module_init(sdp3430_soc_init); + +static void __exit sdp3430_soc_exit(void) +{ + platform_device_unregister(sdp3430_snd_device); +} +module_exit(sdp3430_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC SDP3430"); +MODULE_LICENSE("GPL"); + -- 1.5.6.3

1 0

[alsa-devel] Medion MD96420. ALC883/ALC268 conflicts
by Nigel Henry 25 Nov '08

25 Nov '08

I've been helping GN out on the user list, to get his sound working. The sound is now working, but only by using the options line below. options snd-hda-intel probe_mask=1. I see on the changelog for 1.0.18 > 1.0.18a that Takashi added a quirk for a Medion MD96630 as below. <quote> - ALSA: hda - Add a quirk for MEDION MD96630 Use model=lenovo-ms7195-dig for MEDION MD96630 laptop (17c0:4085) with ALC888 codec. Reference: Novell bnc#412548 https://bugzilla.novell.com/show_bug.cgi?id=412528 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> <end quote> Some details of his machine, OS, and other output below. Medion MD96420. Model MIM 2300 Notebook PC OS: Fedora 9, with alsadriver 1.0.16 upgraded to 1.0.18a Desktop: KDE 4.1 /sbin/lspci (soundcard part) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Mitac Unknown device 8227 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at fc200000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel [graeme@flintstone ~]$ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc200000 irq 22 [graeme@flintstone ~]$ grep ^Codec /proc/asound/card?/codec* /proc/asound/card0/codec#0:Codec: Realtek ALC883 /proc/asound/card0/codec#1:Codec: LSI ID 1040 /proc/asound/card0/codec#2:Codec: Realtek ALC268 A bit of info from Graeme (aka GN), after asking him to try probe_mask=1. <quote> I tried the different probe-mask=?. Probe-mask=1 works OK but not the others. model=lenovo-ms7195-dig doesn't work. Kmix shows varying channels depending on the 'model' or 'probe-mask' used. probe-mask=1 gives the most channels. alsamixer now loads up OK but still no sound. Shows as ALC883 with 'probe-mask=1' Starting Amarok elicits the xine error dialogue window with "xine was unable to initialize any audio drivers" UPDATE: With 'probe-mask=1 and using Rhythmbox I can get sound out of the speakers and the front headphone jack. Volume not very high out of the speakers but is good from the headphones. Too much system activity or keyboard activity (I'm writing this listening to Tchaikovsky and I can hear interruptions from the keyboard at times) seems to interfere with the audio stream. Also there is a lot of noise from the CD drive coming through the earphones. Don't know how that is happening. Pulseaudio Vol Cntl can be used as can the panel Vol. Cntl. and also alsamixer. <end quote> The last things I asked him to try, were to disable Pulseaudio, by removing the package alsa-plugins-pulseaudio, and to try the latest snapshot for the alsa driver. Disabling pulseaudio has resulted in good sound levels from the speakers, but as he is away for a few days, he hasn't yet tried the latest snapshot of the alsa driver. A few extra details. Without a model option all snd modules were loaded, but no card was detected. Using the option model=medion, resulted in the card being detected, but problems with alsamixer as below. model=mitac gives same results as below. <quote from Graeme's reply on user list> [graeme@flintstone ~]$ amixer ALSA lib simple_none.c:1491:(simple_add1) helem (MIXER,'Headphone Playback Switch',0,2,0) appears twice or more amixer: Mixer default load error: Invalid argument [graeme@flintstone ~]$ alsamixer ALSA lib simple_none.c:1491:(simple_add1) helem (MIXER,'Headphone Playback Switch',0,2,0) appears twice or more alsamixer: function snd_mixer_load failed: Invalid argument <end quote> Then I searched through the changelog, found the bug report about the codec conflicts, and suggested probe_mask=1, which has got the sound working. I think I've gone about as far as I can now, which is why I'm posting this to the devel list. Nigel.

2 3

[alsa-devel] Advice requested for developing C01U support via snd-usb-audio
by Keith A. Milner 24 Nov '08

24 Nov '08

Hi, New to this mailing list so please go easy on me. I've been reading through the Alsa docs, wiki, and still not sure of the best approach, so looking for advice from the experts. I'm looking to develop support for the Samson C01U USB studio microphone. http://www.samsontech.com/products/productpage.cfm?prodID=1810 This currently does work via the snd-usb-audio alsa driver, but not in a way that is particularly usable (to the extent that many users have been reporting it as "not working". I thought I might attempt to remedy this. The problem is that, internally, the device contains two digitally controlled amplifiers. Both of these gain controls are exposed via the USB interface (and, hence, by snd-usb-audio) and you can even access the audio stream which is output from each amp. The problem is the raw gain controls are not meant to be accessed directly in this way. The output of one of the amps is fed into the input of the other. With the Windows and Mac OSX drivers, only one gain control is presented. When this is is altered, the driver adjusts the gain of both amplifiers in order to get the correct output level but with minimum noise. In effect, there is a "virtual gain control" which is used to adjust the real gain controls in a predefined way. By careful "twiddling" of the controls it's possible to get the microphone working properly, but it is, quite frankly, a very poor way to handle the problem and is really beyond what should be expected of most users. I would like to do something about this if I can. My aim is to develop something which presents a single gain control to users which controls the internal amp gains in a sensible way. The question is, what is the best way to approach this? Looking at it, I believe there are three possible approaches. In order of my preference they are: 1) Develop at alsa driver level to get a driver which presents a single mixer control to users 2) Develop a new plugin for alsa which can be patched in via asoundrc to provide a generic "Virtual gain control" 3) Provide a user-space tool, a bit like envy24control, which provides this capability My question to the alsa-devel community is which path should I take? As I said my preference is to do something at the driver level simply because it would be most transparent to users and applications: it would simply work. It may also require less code. If this is the correct path to take, this then presents some sub-options (again in order of preference): a) Patch the existing snd-usb-audio driver with some vendor-specific code for this devices (and others which will be similar, like the C03U) b) Fork the snd-usb-driver into a new vendor-specific version and modify as before I prefer a) again because it is transparent to the users, whilst b) will require modprobe or udev configuration, but there may be reasons why it is undesirable to "bloat" the code of snd-usb-audio with vendor-specific tweaks. Bear in mind at this stage I haven't even looked at any of the code yet. Once I do I'm sure to have more questions. But at the moment my question is simply: which is the best way to approach this to be compatible with the aims and philosophy of Alsa? Many thanks, -- Keith A. Milner

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[alsa-devel] [PATCH] ASoC: TWL4030: Disable soft-volume
by Peter Ujfalusi 24 Nov '08

24 Nov '08

Keep Soft-volume disabled for now, since if it is enabled the FGAIN volume controls are not working in the current configuration: CODEC_MODE:OPT_MODE = 1 OPTION:ARXR2_EN = 1 OPTION:ARXL2_EN = 1 OPTION:ARXR1_EN = 0 OPTION:ARXL1_VRX_EN = 0 RX_PATH_SEL:RXL1_SEL = 0x0 (or 0x1) RX_PATH_SEL:RXR1_SEL = 0x0 (or 0x1) After the patch, FGAIN volume control works. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 3348951..49c7da3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -87,7 +87,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_ALC_SET1 (0x2C) */ 0x00, /* REG_ALC_SET2 (0x2D) */ 0x00, /* REG_BOOST_CTL (0x2E) */ - 0x01, /* REG_SOFTVOL_CTL (0x2F) */ + 0x00, /* REG_SOFTVOL_CTL (0x2F) */ 0x00, /* REG_DTMF_FREQSEL (0x30) */ 0x00, /* REG_DTMF_TONEXT1H (0x31) */ 0x00, /* REG_DTMF_TONEXT1L (0x32) */ -- 1.6.0.3

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[alsa-devel] [PATCH] Remove unused variable from tlv320aic23 codec
by Richard A. Holden III 24 Nov '08

24 Nov '08

Remove codec var from tlv320aic23_set_dai_sysclk, it is assigned to but never read. Signed-off-by: Richard A. Holden III <aciddeath(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/tlv320aic23.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 44308da..45395d0 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -421,8 +421,6 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - switch (freq) { case 12000000: return 0; -- 1.5.6.4

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[alsa-devel] (no subject)
by Nitin Mahajan 24 Nov '08

24 Nov '08

unsubscribe Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/

1 0

[alsa-devel] More constants for alsa-lib
by flameeyes＠gmail.com 24 Nov '08

24 Nov '08

git://git.flameeyes.eu/others/alsa/alsa-lib.git flameeyes Diego E. 'Flameeyes' Pettenò (6): Make seq, rawmidi and control operation structures static const. Make snd_pcm_hw_params_names static to pcm_params.c . Make all the PCM plugins ops structure constant. Make all the remaining ops structure constants. Make string arrays as constant as possible. Mark static tables as constant when possible. (This time, signed off right away ;)) -- Diego "Flameeyes" Pettenò http://blog.flameeyes.eu/

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[alsa-devel] [PATCH] Added Realtek ALC888 model entry for the Acer Aspire 4930G laptop
by Petry Vincent 24 Nov '08

24 Nov '08

Hello, I would like to contribute a patch that adds a model entry for the Acer Aspire 4930G laptop. Its chipset is a Realtek ALC888S-VC. The new model is based on the "acer" model and fixes the following bugs missing features: - https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4223 Reversed stereo (invalid report, please close it) - https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4224 Headphone jack sense - https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4225 Internal mic - channel mode (it was missing from the base acer model) The model name is "acer-aspire-4930g" but could be renamed to something more general, if necessary. See attached patch file. It has been made against alsa-driver-1.0.18a. Thanks, Vincent Petry

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[alsa-devel] Suggestion: Jack nodes detection
by Vincent Petry 24 Nov '08

24 Nov '08

Hi, I have noticed that from one model to another in ALC888 (maybe others as well), the nodes for Int speakers, HP out, are not always the same. But in the current driver model in patch_realtek.c we always hard-code the node numbers, for example 0x14 for int speakers, 0x15 for hp out, etc. It would be good to have it this way: 1. Add a spec attribute that contains the output nids in a given order or a map, like this for my laptop: alc888_acer_aspire_jack_nids = { {0x14, NID_INT_SPEAKER}, {0x15, NID_HP_OUT}, {0x18, NID_MIC_IN}, {0x1a, NID_LINE_IN}, {} }; For another laptop, that has built-in subwoofer: alc888_fujitsu_xa2530_jack_nids = { { {0x14, NID_INT_SPEAKER}, {0x15, NID_INT_SPEAKER}, // subwoofer, also considered as int speaker for the jack sense muting {0x17, NID_LINE_OUT}, // this is also side {0x18, NID_MIC_IN}, {0x1a, NID_LINE_IN}, {} }; This way we could then use a function to access the NID number of each one of them, for example get_jack_nid(..., NID_INT_SPEAKER). 2. Once this is implemented, we could add a detection routine that would build the above map automatically, based on the spec info returned by the codec. 3. Then I think we should add some special "detection" model that has no hard-coded values that anyone could use by default, if no suitable model def exists. How about this idea? Is someone already working on it? Thanks, Vincent Petry

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[alsa-devel] [PATCH 0/7] minor HDMI cleanups and fixes
by Wu Fengguang 24 Nov '08

24 Nov '08

Takashi, Here are another set of HDMI cleanups and fixes. [PATCH 1/7] hda - minor HDMI code cleanups [PATCH 2/7] hda - report selected CA index for Audio InfoFrame [PATCH 3/7] hda - make HDMI messages more user friendly [PATCH 4/7] hda - ELD proc interface write updates [PATCH 5/7] hda - document the ELD proc interface [PATCH 6/7] hda - fix DisplayPort naming [PATCH 7/7] hda - fix build warning when CONFIG_PROC_FS=n Thanks, Fengguang --

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[alsa-devel] Problem: different volumes for different jacks in ALC888
by Vincent Petry 22 Nov '08

22 Nov '08

Hi, We have designed a new model for the following laptop: Amilo Xa3530 from Fujitsu Siemens (see bug report 0004240 for alsa-info). The chipset is Realtek ALC888. When using channel mode with 6ch or 8ch, the mic-in jack (0x18) is converted to an output (PIN_OUT 0x40). The problem is that the CLFE channel (0x0e) on the mic-in jack (0x18) is not loud enough. When reassigning the CLFE (0x0e) to the line-out jack (0x17), the volume is at a normal level. But in the codec there is no volume difference between the connection paths from mic-in and side path. I saw that the Mic-in also has a VREF value, but changing it doesn't affect the volume. AFAIK the VREF value only affects Has someone met this kind of issue before? Thanks, Vincent Petry

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Re: [alsa-devel] Creative X-Fi Xtreme Audio (CA0110 chip)
by Vedran Miletić 21 Nov '08

21 Nov '08

Well, don't know what's going on, but anway, snapshots of sound-unstable-2.6 are located here: http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/alsa-d… Lastest one doesn't compile for me on Debian 2.6.26 (pcsp problems or something), but you can always isolate the exact drivers you need and it's likely that those will compile without problems. 2008/11/21 Dmitri Seletski <drjoms(a)gmail.com>: > Hi Vedran, once again, could I get experimental drivers for Creative XFI pro > gamer drivers?(link sent to manuel seemed to be broken) > Regards > > On Thu, Nov 20, 2008 at 1:51 PM, Vedran Miletić <rivanvx(a)gmail.com> wrote: >> >> CA0110 HDA mode is supported in this tree: >> >> http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-unstable-2.6.git;a=su… >> CA0111 (PCI version) is supported by snd-ca0106, not sure if CA0106 >> mode can be supported that way too. >> >> 2008/11/19 Manuel C. <manuel(a)ekerazha.com>: >> > Hi, >> > I want to bring to your attention the Creative X-Fi Xtreme Audio card >> > with CA0110 chip, it is also bundled with some MSI motherboards. >> > >> > CA0110 web page: http://www.creative.com/oem/products/chips/CA0110.asp >> > >> > >From that web page: >> > "CA0110 is a PCI Express audio chip using the CA0111 audio mode or High >> > Definition Audio mode. >> > >> > [...] >> > >> > CA0106 mode >> > - Same as original CA0111." >> > >> > So it should be a PCI Express version of the CA0111 chip (and CA0111 >> > could be similar to the CA0106 chip). >> > >> > The CA0110 chip isn't even listed in the Creative products matrix on the >> > wiki. Is a CA0110 driver under development? Thank you. >> > >> > _______________________________________________ >> > Alsa-devel mailing list >> > Alsa-devel(a)alsa-project.org >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> > >> >> >> >> -- >> Vedran Miletić >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- Vedran Miletić

1 0

[alsa-devel] [PATCH] ASoC: Add support for OMAP3 EVM
by Anuj Aggarwal 21 Nov '08

21 Nov '08

This patch adds ALSA SoC support for OMAP3 EVM using TWL4030 audio codec. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Kconfig | 7 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3evm.c | 147 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 156 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3evm.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 6c56277..2a2a74e 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -38,5 +38,12 @@ config SND_OMAP_SOC_OMAP2EVM help Say Y if you want to add support for SoC audio on the omap2evm board. +config SND_OMAP_SOC_OMAP3EVM + tristate "SoC Audio support for OMAP3EVM board" + depends on SND_OMAP_SOC && MACH_OMAP3EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the omap3evm board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index f5da3cc..bbb3a77 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -10,9 +10,11 @@ snd-soc-n810-objs := n810.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o +snd-soc-omap3evm-objs := omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c new file mode 100644 index 0000000..570af55 --- /dev/null +++ b/sound/soc/omap/omap3evm.c @@ -0,0 +1,147 @@ +/* + * omap3evm.c -- ALSA SoC support for OMAP3 EVM + * + * Author: Anuj Aggarwal <anuj.aggarwal(a)ti.com> + * + * Based on sound/soc/omap/beagle.c by Steve Sakoman + * + * Copyright (C) 2008 Texas Instruments, Incorporated + * + * 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 version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, + * whether express or implied; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap3evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3evm_ops = { + .hw_params = omap3evm_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3evm_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_omap3evm = { + .name = "omap3evm", + .dai_link = &omap3evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3evm_snd_devdata = { + .machine = &snd_soc_machine_omap3evm, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3evm_snd_device; + +static int __init omap3evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap3evm()) { + pr_debug("Not OMAP3 EVM!\n"); + return -ENODEV; + } + pr_info("OMAP3 EVM SoC init\n"); + + omap3evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata); + omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev; + *(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(omap3evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3evm_snd_device); + + return ret; +} + +static void __exit omap3evm_soc_exit(void) +{ + platform_device_unregister(omap3evm_snd_device); +} + +module_init(omap3evm_soc_init); +module_exit(omap3evm_soc_exit); + +MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); +MODULE_LICENSE("GPL"); -- 1.6.0.2.1172.ga5ed0

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[alsa-devel] Where are dma interrupts handled in ASoC code?
by John Kacur 21 Nov '08

21 Nov '08

Where are dma overflow interrupts handled? For example I don't see this in sound/soc/s3c24xx-i2s.c Thanks

2 3

[alsa-devel] twl4030 asoc kcontrols and widgets
by naveen krishna ch 21 Nov '08

21 Nov '08

Hi Alsa devel, I have been working on TWL4030 codec driver for ALSA SOC. I have taken sound/soc/codec/twl4030.c as reference from main line This Patch adds some kcontrols, widgets and interconnection map for some of the TWL4030 ASOC codec I was building it for a custom board as it was for OVERO Suggestions on the DAPM part of the driver would be helpful Thanks in advance. The patch is as follows. --- twl4030.c 2008-11-19 12:04:32.000000000 +0530 +++ /home/chnaveen/Desktop/twl4030.c 2008-11-21 15:08:06.000000000 +0530 @@ -16,7 +16,9 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA * 02110-1301 USA - * + * Modified by : Naveen Krishna Ch + * Added Kcontrols for OMAP3 WaterlooBoard + * Dated : 28th october 2008 */ #include <linux/module.h> @@ -29,24 +31,27 @@ #include <linux/i2c/twl4030.h> #include <sound/core.h> #include <sound/pcm.h> +#include <sound/jack.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> - +#include <asm/arch/twl4030.h> #include "twl4030.h" +static int twl4030_jack_func; + /* * twl4030 register cache & default register settings */ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* this register not used */ - 0x93, /* REG_CODEC_MODE (0x1) */ + 0x03, /* REG_CODEC_MODE (0x1) */ 0xc3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ - 0x24, /* REG_ANAMICL (0x5) */ - 0x04, /* REG_ANAMICR (0x6) */ + 0xb0, /* REG_ANAMICL (0x5) */ + 0x10, /* REG_ANAMICR (0x6) */ 0x0a, /* REG_AVADC_CTL (0x7) */ 0x00, /* REG_ADCMICSEL (0x8) */ 0x00, /* REG_DIGMIXING (0x9) */ @@ -67,22 +72,22 @@ 0x00, /* REG_ARX2VTXPGA (0x18) */ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ - 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ - 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x2b, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x2b, /* REG_ARXR2_APGA_CTL (0x1C) */ 0x00, /* REG_ATX2ARXPGA (0x1D) */ 0x00, /* REG_BT_IF (0x1E) */ 0x00, /* REG_BTPGA (0x1F) */ 0x00, /* REG_BTSTPGA (0x20) */ - 0x00, /* REG_EAR_CTL (0x21) */ - 0x24, /* REG_HS_SEL (0x22) */ - 0x0a, /* REG_HS_GAIN_SET (0x23) */ - 0x00, /* REG_HS_POPN_SET (0x24) */ - 0x00, /* REG_PREDL_CTL (0x25) */ - 0x00, /* REG_PREDR_CTL (0x26) */ - 0x00, /* REG_PRECKL_CTL (0x27) */ - 0x00, /* REG_PRECKR_CTL (0x28) */ - 0x00, /* REG_HFL_CTL (0x29) */ - 0x00, /* REG_HFR_CTL (0x2A) */ + 0x20, /* REG_EAR_CTL (0x21) */ + 0x00, /* REG_HS_SEL (0x22) */ + 0x05, /* REG_HS_GAIN_SET (0x23) */ + 0x42, /* REG_HS_POPN_SET (0x24) */ + 0x20, /* REG_PREDL_CTL (0x25) */ + 0x20, /* REG_PREDR_CTL (0x26) */ + 0x20, /* REG_PRECKL_CTL (0x27) */ + 0x20, /* REG_PRECKR_CTL (0x28) */ + 0x1f, /* REG_HFL_CTL (0x29) */ + 0x1f, /* REG_HFR_CTL (0x2A) */ 0x00, /* REG_ALC_CTL (0x2B) */ 0x00, /* REG_ALC_SET1 (0x2C) */ 0x00, /* REG_ALC_SET2 (0x2D) */ @@ -112,9 +117,40 @@ 0x00, /* REG_VIBRA_CTL (0x45) */ 0x00, /* REG_VIBRA_SET (0x46) */ 0x00, /* REG_VIBRA_PWM_SET (0x47) */ - 0x00, /* REG_ANAMIC_GAIN (0x48) */ + 0x24, /* REG_ANAMIC_GAIN (0x48) */ 0x00, /* REG_MISC_SET_2 (0x49) */ }; +static void twl4030_ext_control(struct snd_soc_codec *codec) +{ + if (twl4030_jack_func) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + snd_soc_dapm_sync(codec); +} + +static int twl4030_get_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = twl4030_jack_func; + + return 0; +} + +static int twl4030_set_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (twl4030_jack_func == ucontrol->value.integer.value[0]) + return 0; + + twl4030_jack_func = ucontrol->value.integer.value[0]; + twl4030_ext_control(codec); + + return 1; +} /* * read twl4030 register cache @@ -188,14 +224,83 @@ twl4030_write(codec, i, twl4030_reg[i]); } +static const char *jack_function[] = {"Off", "Headphone"}; + +static const struct soc_enum twl4030_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function), +}; + static const struct snd_kcontrol_new twl4030_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", - TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 0, 127, 0), - SOC_DOUBLE_R("Capture Volume", - TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, - 0, 127, 0), + + /* Master Playback Volume Controls */ + SOC_DOUBLE_R("Master PLayback Course Gain ctrl", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 6, 3, 0), + SOC_DOUBLE_R("Master Playback Fine Gain ctrl", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 0, 63, 0), + + /* Playback Speaker volume controls */ + SOC_DOUBLE_R("Speaker R2+L2 Volume ctrls", + TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, + 3, 17, 0), + + /* Capture Gain controls */ + SOC_DOUBLE_R("Master Capture Gain ctrl", + TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, + 0, 31, 0), + /* Loop gain controls*/ + SOC_DOUBLE("Loop Gain ctrl", TWL4030_REG_ATX2ARXPGA, + 3 , 0, 7, 0), + + SOC_DOUBLE("Main +Sub mic capture gain ctrl", + TWL4030_REG_ANAMIC_GAIN, 3 , 0, 5, 0), + + SOC_DOUBLE_R("External Speaker Volume control", + TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL, + 4, 3, 0), + + SOC_DOUBLE_R("Pre Car kit Volume control", + TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL, + 4, 3, 0), + + SOC_SINGLE("DACL2", TWL4030_REG_AVDAC_CTL, 3, 1, 0), + SOC_SINGLE("DACR2", TWL4030_REG_AVDAC_CTL, 2, 1, 0), + + SOC_ENUM_EXT("Jack Function", twl4030_enum[0], + twl4030_get_jack, twl4030_set_jack), +}; + +/* Right PGA Mixer control switches */ +static const struct snd_kcontrol_new twl4030_right_pga_mixer_controls[] = { + SOC_DAPM_SINGLE("HS Mic switch", TWL4030_REG_ANAMICL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux/FM right switch", TWL4030_REG_ANAMICR, 2, 1, 0), + SOC_DAPM_SINGLE("Sub Mic switch", TWL4030_REG_ANAMICR, 0, 1, 0), +}; + +/* Left PGA Mixer control switches */ +static const struct snd_kcontrol_new twl4030_left_pga_mixer_controls[] = { + SOC_DAPM_SINGLE("HS Mic switch", TWL4030_REG_ANAMICL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux/FM left switch", TWL4030_REG_ANAMICL, 2, 1, 0), + SOC_DAPM_SINGLE("Main Mic switch", TWL4030_REG_ANAMICL, 0, 1, 0), +}; + +/* Right DACR2 Mixer */ +static const struct snd_kcontrol_new twl4030_dacr2_mixer_controls[] = { + SOC_DAPM_SINGLE("Headset-R switch", TWL4030_REG_HS_SEL, 5, 1, 0), + SOC_DAPM_SINGLE("Handsfree-R switch", TWL4030_REG_HFR_CTL, 5, 1, 0), + SOC_DAPM_SINGLE("PRECK-R switch", TWL4030_REG_PRECKR_CTL, 2, 1, 0), + SOC_DAPM_DOUBLE("E class-D amp-R switch", TWL4030_REG_PREDR_CTL, 3, 2, 1, 0, 0), +}; + +/* Left DACL2 Mixer */ +static const struct snd_kcontrol_new twl4030_dacl2_mixer_controls[] = { + SOC_DAPM_SINGLE("Headset-L switch", TWL4030_REG_HS_SEL, 2, 1, 0), + SOC_DAPM_SINGLE("Handsfree-L switch", TWL4030_REG_HFL_CTL, 5, 1, 0), + SOC_DAPM_SINGLE("Ear peice switch", TWL4030_REG_EAR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("PRECK-L switch", TWL4030_REG_PRECKL_CTL, 2, 1, 0), + SOC_DAPM_DOUBLE("E class-D amp-L switch", TWL4031_REG_PREDL_CTL, 3, 2, 1, 0, 0), }; /* add non dapm controls */ @@ -215,27 +320,96 @@ } static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("INL"), - SND_SOC_DAPM_INPUT("INR"), - - SND_SOC_DAPM_OUTPUT("OUTL"), - SND_SOC_DAPM_OUTPUT("OUTR"), - - SND_SOC_DAPM_DAC("DACL", "Left Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DACR", "Right Playback", SND_SOC_NOPM, 0, 0), - - SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_HP("Headphone Jack", NULL), + + /* Left ADC for capture */ + SND_SOC_DAPM_ADC("ADCL", "Left Capture ADC", TWL4030_REG_AVADC_CTL, 3, 0), + + /* dapm widget (path domain) for left PGA mixer */ + SND_SOC_DAPM_MIXER("LINEIN-L", SND_SOC_NOPM, 0, 0, + &twl4030_left_pga_mixer_controls[0], + ARRAY_SIZE(twl4030_left_pga_mixer_controls)), + + /* Right ADC for capture*/ + SND_SOC_DAPM_ADC("ADCR", "Right Capture ADC", TWL4030_REG_AVADC_CTL, 1, 0), + + /* dapm widget (path domain) for right PGA mixer */ + SND_SOC_DAPM_MIXER("LINEIN-R", SND_SOC_NOPM, 0, 0, + &twl4030_right_pga_mixer_controls[0], + ARRAY_SIZE(twl4030_right_pga_mixer_controls)), + + /* dapm widget (path domain) for left DACL2 Mixer */ + + SND_SOC_DAPM_MIXER("DACL2 Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dacl2_mixer_controls[0], + ARRAY_SIZE(twl4030_dacl2_mixer_controls)), + + /* dapm widget (path domain) for DACR2 Mixer */ + SND_SOC_DAPM_MIXER("DACR2 Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dacr2_mixer_controls[0], + ARRAY_SIZE(twl4030_dacr2_mixer_controls)), + + /* Inputs Left*/ + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("Aux/fm left"), + SND_SOC_DAPM_INPUT("Main mic"), + + /* Inputs Right*/ + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("Aux/fm right"), + SND_SOC_DAPM_INPUT("Sub mic"), + + /* Outputs Left*/ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("IHF_LEFT"), + SND_SOC_DAPM_OUTPUT("EAR"), + SND_SOC_DAPM_OUTPUT("E class-D L"), + SND_SOC_DAPM_OUTPUT("PRECK-L"); + + /* Outputs Right */ + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("IHF_RIGHT"), + SND_SOC_DAPM_OUTPUT("E class-D R"), + SND_SOC_DAPM_OUTPUT("PRECK-R"); }; static const struct snd_soc_dapm_route intercon[] = { - /* outputs */ - {"OUTL", NULL, "DACL"}, - {"OUTR", NULL, "DACR"}, - - /* inputs */ - {"ADCL", NULL, "INL"}, - {"ADCR", NULL, "INR"}, + + /* ******** Left input ******** */ + {"LINEIN-L", "HS Mic switch", "HSMIC"}, + {"LINEIN-L", "Aux/FM left switch", "Aux/fm left"}, + {"LINEIN-L", "Main Mic switch", "Main mic"}, + + {"ADCL", NULL, "LINEIN-L"}, + + + /* ******** Right Input ******** */ + {"LINEIN-R", "HS Mic switch", "HSMIC"}, + {"LINEIN-R", "Aux/FM right switch", "Aux/fm right"}, + {"LINEIN-R", "Sub Mic switch", "Sub mic"}, + + {"ADCR", NULL, "LINEIN-R"}, + + /* ******** Left Output ******** */ + //{"DACL2", NULL, "DACL2 Mixer"}, + + {"DACL2 Mixer", "Headset-L switch", "HSOL"}, + {"DACL2 Mixer", "Handsfree-L switch", "IHF_LEFT"}, + {"DACL2 Mixer", "Ear peice switch", "EAR"}, + {"DACL2 Mixer", "PRECK-L switch", "PRECK-L"}, + {"DACL2 Mixer","E class-D amp-L switch","E class-D L"}, + + {"Headphone Jack",NULL, "HSOL"}, + /* ******** Right Output ******** */ + //{"DACR2", NULL, "DACR2 Mixer"}, + + {"DACR2 Mixer", "Headset-R switch","HSOR"}, + {"DACR2 Mixer", "Handsfree-R switch","IHF_RIGHT"}, + {"DACR2 Mixer", "PRECK-R switch", "PRECK-R"}, + {"DACR2 Mixer","E class-D amp-R switch","E class-D R"}, + + {"Headphone Jack",NULL, "HSOR"}, }; static int twl4030_add_widgets(struct snd_soc_codec *codec) @@ -280,7 +454,7 @@ /* toggle CODECPDZ as per TRM */ twl4030_clear_codecpdz(codec); twl4030_set_codecpdz(codec); - + /* program anti-pop with bias ramp delay */ popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); popn &= TWL4030_RAMP_DELAY; @@ -384,6 +558,9 @@ case 48000: mode |= TWL4030_APLL_RATE_48000; break; + case 96000: + mode |= TWL4030_APLL_RATE_96000; + break; default: printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", params_rate(params)); @@ -434,6 +611,7 @@ u8 infreq; switch (freq) { + case 19200000: infreq = TWL4030_APLL_INFREQ_19200KHZ; break; @@ -504,20 +682,24 @@ return 0; } -#define TWL4030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define TWL4030_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) + #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { .stream_name = "Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .capture = { .stream_name = "Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, @@ -616,13 +798,14 @@ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) - return -ENOMEM; - + return -ENOMEM; + socdev->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); - + + twl4030_socdev = socdev; twl4030_init(socdev); Patch ends.. I need suggestions on the DAPM part of the driver. Thanks and Regards, (: Naveen Krishna Ch :)

4 7

[alsa-devel] [PATCH 1/1] ASoC: Fix typo in snd_soc_card update documentation
by Mark Brown 21 Nov '08

21 Nov '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Documentation/sound/alsa/soc/machine.txt | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index 4a9f51e..bab7711 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -9,7 +9,7 @@ the audio subsystem with the kernel as a platform device and is represented by the following struct:- /* SoC machine */ -struct snd_soc_card {_ +struct snd_soc_card { char *name; int (*probe)(struct platform_device *pdev); -- 1.5.6.5

2 1

[alsa-devel] [PATCH 0/10] ASoC updates
by Mark Brown 21 Nov '08

21 Nov '08

The following changes since commit 9b0db7e7fd20d5a38844e9435f7d4246ea44978a: Mark Brown (1): ASoC: Convert blackfin machines to use DAI accessor functions are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Jarkko Nikula (1): ASoC: OMAP: Fix preprocessor filled DAI name in McBSP DAI Karl Beldan (2): ASoC: ssm2602: Fix priv substreams refs ASoC: ssm2602: Update supported stream formats Mark Brown (7): ASoC: Rename snd_soc_card to snd_soc_machine ASoC: Add abbreviation to maintainers for searchability ASoC: Move DAI structure definitions into new soc-dai.h ASoC: Merge snd_soc_ops into snd_soc_dai_ops ASoC: Add Marvell Zylonite machine support ASoC: Staticise pxa2xx_pcm_ops ASoC: Improve error reporting for AC97 reset failures Documentation/sound/alsa/soc/machine.txt | 8 +- MAINTAINERS | 2 +- include/sound/soc-dai.h | 225 ++++++++++++++++++++++++++++++ include/sound/soc.h | 156 +-------------------- sound/soc/atmel/atmel_ssc_dai.c | 21 ++-- sound/soc/atmel/playpaq_wm8510.c | 4 +- sound/soc/atmel/sam9g20_wm8731.c | 4 +- sound/soc/au1x/psc-ac97.c | 5 +- sound/soc/au1x/psc-i2s.c | 8 +- sound/soc/au1x/sample-ac97.c | 4 +- sound/soc/blackfin/bf5xx-ad1980.c | 6 +- sound/soc/blackfin/bf5xx-ad73311.c | 6 +- sound/soc/blackfin/bf5xx-i2s.c | 12 +- sound/soc/blackfin/bf5xx-ssm2602.c | 6 +- sound/soc/codecs/ac97.c | 3 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ak4535.c | 5 +- sound/soc/codecs/cs4270.c | 11 +- sound/soc/codecs/ssm2602.c | 39 ++++- sound/soc/codecs/tlv320aic23.c | 19 ++-- sound/soc/codecs/tlv320aic26.c | 5 +- sound/soc/codecs/tlv320aic3x.c | 5 +- sound/soc/codecs/twl4030.c | 5 +- sound/soc/codecs/uda134x.c | 12 +- sound/soc/codecs/uda1380.c | 15 +-- sound/soc/codecs/wm8510.c | 5 +- sound/soc/codecs/wm8580.c | 12 +- sound/soc/codecs/wm8728.c | 5 +- sound/soc/codecs/wm8731.c | 11 +- sound/soc/codecs/wm8750.c | 5 +- sound/soc/codecs/wm8753.c | 25 ++-- sound/soc/codecs/wm8900.c | 5 +- sound/soc/codecs/wm8903.c | 11 +- sound/soc/codecs/wm8971.c | 5 +- sound/soc/codecs/wm8990.c | 6 +- sound/soc/codecs/wm9712.c | 8 +- sound/soc/codecs/wm9713.c | 23 ++-- sound/soc/davinci/davinci-evm.c | 4 +- sound/soc/davinci/davinci-i2s.c | 20 ++-- sound/soc/davinci/davinci-sffsdr.c | 7 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_ssi.c | 14 +- sound/soc/fsl/mpc5200_psc_i2s.c | 17 ++- sound/soc/fsl/mpc8610_hpcd.c | 6 +- sound/soc/fsl/soc-of-simple.c | 10 +- sound/soc/omap/n810.c | 4 +- sound/soc/omap/omap-mcbsp.c | 16 ++- sound/soc/omap/omap2evm.c | 7 +- sound/soc/omap/omap3beagle.c | 4 +- sound/soc/omap/osk5912.c | 4 +- sound/soc/omap/overo.c | 4 +- sound/soc/pxa/Makefile | 2 + sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/e800_wm9712.c | 6 +- sound/soc/pxa/em-x270.c | 4 +- sound/soc/pxa/palm27x.c | 4 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa-ssp.c | 20 +-- sound/soc/pxa/pxa2xx-ac97.c | 9 +- sound/soc/pxa/pxa2xx-i2s.c | 15 ++- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/tosa.c | 6 +- sound/soc/pxa/zylonite.c | 219 +++++++++++++++++++++++++++++ sound/soc/s3c24xx/ln2440sbc_alc650.c | 6 +- sound/soc/s3c24xx/neo1973_wm8753.c | 6 +- sound/soc/s3c24xx/s3c2412-i2s.c | 8 +- sound/soc/s3c24xx/s3c2443-ac97.c | 8 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 9 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 4 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 6 +- sound/soc/sh/hac.c | 3 +- sound/soc/sh/sh7760-ac97.c | 4 +- sound/soc/sh/ssi.c | 16 +- sound/soc/soc-core.c | 164 +++++++++++----------- sound/soc/soc-dapm.c | 6 +- 76 files changed, 851 insertions(+), 516 deletions(-) create mode 100644 include/sound/soc-dai.h create mode 100644 sound/soc/pxa/zylonite.c

3 14

[alsa-devel] [PATCH 0/1] ASoC: OMAP: Fix preprocessor filled DAI name in McBSP DAI
by Jarkko Nikula 21 Nov '08

21 Nov '08

IMO, this is a minor fix, so generated on top of topic/asoc. -- Jarkko

2 2

[alsa-devel] Fixes and misc improvements for alsa-lib, alsa-plugins, alsa-utils
by flameeyes＠gmail.com 21 Nov '08

21 Nov '08

I've taken some time to fix some autotools nitpicks and some generic optimisations, I hope they can be useful: git://git.flameeyes.eu/others/alsa/alsa-lib.git flameeyes Diego E. 'Flameeyes' Pettenò (5): Don't use AC_CANONICAL_SYSTEM, only use AC_CANONICAL_HOST. Add the attributes.m4 macro file from xine/lscube. Check for --no-undefined linker flag and use it. Make sure that python libraries are passed through LIBADD. Make some static tables and strings constants. git://git.flameeyes.eu/others/alsa/alsa-plugins.git flameeyes Diego E. 'Flameeyes' Pettenò (6): Add the attributes.m4 macro file from xine/lscube. Check for --no-undefined linker flag and use it. Cleanup flags in maemo/Makefile.am [RFC] Don't use pow() for calculating a power of 2, use shift instead. Make some static tables and strings constants. Mark as static the functions not used outside their unit. git://git.flameeyes.eu/others/alsa/alsa-utils.git flameeyes Diego E. 'Flameeyes' Pettenò (4): Remove some dead code (comparisons between 0 and unsigned integers). Remove some unused variables. Mark static the functions not used outside their unit. Make some static tables and strings constants. -- Diego "Flameeyes" Pettenò http://blog.flameeyes.eu/

2 5

[alsa-devel] [PATCH] ARM DaVinci: Add PCM3008 ALSA SoC driver for the Lyrtech SFFSDR board
by Hugo Villeneuve 21 Nov '08

21 Nov '08

The PCM3008 is used on the Lyrtech SFFSDR board, in conjunction with an FPGA that generates the bit clock and the master clock Signed-off-by: Hugo Villeneuve <hugo.villeneuve(a)lyrtech.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm3008.c | 223 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/pcm3008.h | 25 ++++ sound/soc/davinci/Kconfig | 9 ++ sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-sffsdr.c | 145 +++++++++++++++++++++++ 7 files changed, 410 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/pcm3008.c create mode 100644 sound/soc/codecs/pcm3008.h create mode 100644 sound/soc/davinci/davinci-sffsdr.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5df7402..dc58ce2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -80,6 +80,10 @@ config SND_SOC_TWL4030 tristate depends on TWL4030_CORE +config SND_SOC_PCM3008 + tristate + depends on SFFSDR_FPGA + config SND_SOC_UDA1380 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3b9b58a..7af88e7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o +snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -26,6 +27,7 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c new file mode 100644 index 0000000..20d8b94 --- /dev/null +++ b/sound/soc/codecs/pcm3008.c @@ -0,0 +1,223 @@ +/* + * ALSA Soc PCM3008 codec support + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * Based on AC97 Soc codec, original copyright follow: + * + * Copyright 2005 Wolfson Microelectronics PLC. + * + * 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. + * + * Generic PCM3008 support. + */ + +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/gpio.h> +#include <asm/plat-sffsdr/sffsdr-fpga.h> + +#include "pcm3008.h" + +#define PCM3008_VERSION "0.1" + +#define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +static int pcm3008_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int fs; + + /* Fsref can be 32000, 44100 or 48000. */ + fs = params_rate(params); + + printk(KERN_INFO "pcm3008_hw_params: rate = %d Hz\n", fs); + + return sffsdr_fpga_set_codec_fs(fs); +} + +struct snd_soc_dai pcm3008_dai = { + .name = "PCM3008 HiFi", + .type = SND_SOC_DAI_I2S, + .playback = { + .stream_name = "PCM3008 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = PCM3008_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "PCM3008 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = PCM3008_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .hw_params = pcm3008_hw_params, + }, +}; +EXPORT_SYMBOL_GPL(pcm3008_dai); + +static void pcm3008_gpio_free(struct pcm3008_setup_data *setup) +{ + gpio_free(setup->dem0_pin); + gpio_free(setup->dem1_pin); + gpio_free(setup->pdad_pin); + gpio_free(setup->pdda_pin); +} + +static int pcm3008_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct pcm3008_setup_data *setup = socdev->codec_data; + int ret = 0; + + printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (!socdev->codec) + return -ENOMEM; + + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->name = "PCM3008"; + codec->owner = THIS_MODULE; + codec->dai = &pcm3008_dai; + codec->num_dai = 1; + codec->write = NULL; + codec->read = NULL; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* Register PCMs. */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "pcm3008: failed to create pcms\n"); + goto pcm_err; + } + + /* Register Card. */ + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "pcm3008: failed to register card\n"); + goto card_err; + } + + /* DEM1 DEM0 DE-EMPHASIS_MODE + * Low Low De-emphasis 44.1 kHz ON + * Low High De-emphasis OFF + * High Low De-emphasis 48 kHz ON + * High High De-emphasis 32 kHz ON + */ + + /* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */ + ret = gpio_request(setup->dem0_pin, "codec_dem0"); + if (ret == 0) + ret = gpio_direction_output(setup->dem0_pin, 1); + if (ret != 0) + goto gpio_err; + + /* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */ + ret = gpio_request(setup->dem1_pin, "codec_dem1"); + if (ret == 0) + ret = gpio_direction_output(setup->dem1_pin, 0); + if (ret != 0) + goto gpio_err; + + /* Configure PDAD GPIO. */ + ret = gpio_request(setup->pdad_pin, "codec_pdad"); + if (ret == 0) + ret = gpio_direction_output(setup->pdad_pin, 1); + if (ret != 0) + goto gpio_err; + + /* Configure PDDA GPIO. */ + ret = gpio_request(setup->pdda_pin, "codec_pdda"); + if (ret == 0) + ret = gpio_direction_output(setup->pdda_pin, 1); + if (ret != 0) + goto gpio_err; + + return ret; + +gpio_err: + pcm3008_gpio_free(setup); +card_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->codec); + + return ret; +} + +static int pcm3008_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct pcm3008_setup_data *setup = socdev->codec_data; + + if (!codec) + return 0; + + pcm3008_gpio_free(setup); + snd_soc_free_pcms(socdev); + kfree(socdev->codec); + + return 0; +} + +#ifdef CONFIG_PM +static int pcm3008_soc_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct pcm3008_setup_data *setup = socdev->codec_data; + + printk(KERN_INFO "pcm3008_soc_suspend(): TODO\n"); + + gpio_set_value(setup->pdad_pin, 0); + gpio_set_value(setup->pdda_pin, 0); + + return 0; +} + +static int pcm3008_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct pcm3008_setup_data *setup = socdev->codec_data; + + printk(KERN_INFO "pcm3008_soc_resume(): TODO\n"); + + gpio_set_value(setup->pdad_pin, 1); + gpio_set_value(setup->pdda_pin, 1); + + return 0; +} +#else +#define pcm3008_soc_suspend NULL +#define pcm3008_soc_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_pcm3008 = { + .probe = pcm3008_soc_probe, + .remove = pcm3008_soc_remove, + .suspend = pcm3008_soc_suspend, + .resume = pcm3008_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008); + +MODULE_DESCRIPTION("Soc PCM3008 driver"); +MODULE_AUTHOR("Hugo Villeneuve"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h new file mode 100644 index 0000000..81c82c9 --- /dev/null +++ b/sound/soc/codecs/pcm3008.h @@ -0,0 +1,25 @@ +/* + * PCM3008 ALSA SoC Layer + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * 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. + */ + +#ifndef __LINUX_SND_SOC_PCM3008_H +#define __LINUX_SND_SOC_PCM3008_H + +struct pcm3008_setup_data { + u8 dem0_pin; + u8 dem1_pin; + u8 pdad_pin; + u8 pdda_pin; +}; + +extern struct snd_soc_codec_device soc_codec_dev_pcm3008; +extern struct snd_soc_dai pcm3008_dai; + +#endif diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 8f7e338..7978b81 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -17,3 +17,12 @@ config SND_DAVINCI_SOC_EVM help Say Y if you want to add support for SoC audio on TI DaVinci EVM platform. + +config SND_DAVINCI_SOC_SFFSDR + tristate "SoC Audio support for SFFSDR" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR + select SND_DAVINCI_SOC_I2S + select SND_SOC_PCM3008 + help + Say Y if you want to add support for SoC audio on + Lyrtech SFFSDR board. diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca772e5..ca8bae1 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -7,5 +7,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o +snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c new file mode 100644 index 0000000..b232388 --- /dev/null +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -0,0 +1,145 @@ +/* + * ASoC driver for Lyrtech SFFSDR board. + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/gpio.h> +#include <asm/dma.h> +#include <mach/mcbsp.h> +#include <mach/edma.h> + +#include "../codecs/pcm3008.h" +#include "davinci-pcm.h" +#include "davinci-i2s.h" + +static int sffsdr_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + /* Set cpu DAI configuration: + * CLKX and CLKR are the inputs for the Sample Rate Generator. + * FSX and FSR are outputs, driven by the sample Rate Generator. */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_RIGHT_J | + SND_SOC_DAIFMT_CBM_CFS | + SND_SOC_DAIFMT_IB_NF); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops sffsdr_ops = { + .hw_params = sffsdr_hw_params, +}; + +/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sffsdr_dai = { + .name = "PCM3008", /* Codec name */ + .stream_name = "PCM3008 HiFi", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &pcm3008_dai, + .ops = &sffsdr_ops, +}; + +/* davinci-sffsdr audio machine driver */ +static struct snd_soc_machine snd_soc_machine_sffsdr = { + .name = "DaVinci SFFSDR", + .dai_link = &sffsdr_dai, + .num_links = 1, +}; + +/* sffsdr audio private data */ +static struct pcm3008_setup_data sffsdr_pcm3008_setup = { + .dem0_pin = GPIO(45), + .dem1_pin = GPIO(46), + .pdad_pin = GPIO(47), + .pdda_pin = GPIO(38), +}; + +/* sffsdr audio subsystem */ +static struct snd_soc_device sffsdr_snd_devdata = { + .machine = &snd_soc_machine_sffsdr, + .platform = &davinci_soc_platform, + .codec_dev = &soc_codec_dev_pcm3008, + .codec_data = &sffsdr_pcm3008_setup, +}; + +static struct resource sffsdr_snd_resources[] = { + { + .start = DAVINCI_MCBSP_BASE, + .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data sffsdr_snd_data = { + .tx_dma_ch = DAVINCI_DMA_MCBSP_TX, + .rx_dma_ch = DAVINCI_DMA_MCBSP_RX, +}; + +static struct platform_device *sffsdr_snd_device; + +static int __init sffsdr_init(void) +{ + int ret; + + sffsdr_snd_device = platform_device_alloc("soc-audio", 0); + if (!sffsdr_snd_device) { + printk(KERN_ERR "platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata); + sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev; + sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data; + + ret = platform_device_add_resources(sffsdr_snd_device, + sffsdr_snd_resources, + ARRAY_SIZE(sffsdr_snd_resources)); + if (ret) { + printk(KERN_ERR "platform device add ressources failed\n"); + goto error; + } + + ret = platform_device_add(sffsdr_snd_device); + if (ret) + goto error; + + return ret; + +error: + platform_device_put(sffsdr_snd_device); + return ret; +} + +static void __exit sffsdr_exit(void) +{ + platform_device_unregister(sffsdr_snd_device); +} + +module_init(sffsdr_init); +module_exit(sffsdr_exit); + +MODULE_AUTHOR("Hugo Villeneuve"); +MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver"); +MODULE_LICENSE("GPL"); -- 1.5.5

3 3

[alsa-devel] Alsa "short read" questions
by Pierre Vanhoucke 21 Nov '08

21 Nov '08

Hi All, Is there anybody out there who can help me to answer the following questions: I have converted an application called Linrad ( http://www.nitehawk.com/sm5bsz/linuxdsp/linrad.htm ) from OSS to ALSA. Linrad is a remarkable DSP application used worldwide by many radio-hams. I am facing hereby a problem with a "short read" : An ALSA device is opened in blocking mode: ********************************************* err=snd_pcm_open(&rx_ad_handle, rx_sndin_plughw_pcm_name, SND_PCM_STREAM_CAPTURE, 0); ********************************************* When I subsequently read the device with following statement I sometimes get a "short read": ******************************************** err=snd_pcm_readi(rx_ad_handle,rxin_isho, rxad.block_frames); if((unsigned int)err != rxad.block_frames) { if(err == -EPIPE) // EPIPE, try recovery { err= snd_pcm_recover(rx_ad_handle, err, 0); if(err < 0)lirerr(1108); // recovery failed } else { if(err < 0) { lirerr(1113); // catch all } else { lirerr(1116); // short read } } } } ************************************************** Here are my questions: 1. How is it possible to get a "short read" in blocking mode ? I would think that in blocking mode the control is only returned after the read is completed. 2. What are the possible causes for a "short read" ? 3. What can be done to avoid a "short read" ? 4. How am I supposed to recover from a "short read"? Best regards, Pierre Vanhoucke/ ON5GN

2 1

Re: [alsa-devel] [Intel-gfx] [RFC][PATCH] ELD routines and proc interface
by Wu Fengguang 21 Nov '08

21 Nov '08

On Thu, Nov 20, 2008 at 05:46:49PM -0800, Shane W wrote: > On Fri, Nov 21, 2008 at 09:36:55AM +0800, Wu Fengguang wrote: > > Yes it's unexpected. Do you know its real speaker numbers and > > allocations? Does it provide some number of line-out ports? > > There are six speakers connected, fl, ct, fr, rl, rr and lfe. Hi Shane, Apply this patch and run "echo speakers f > /proc/asound/card0/eld\#3". That will make the ELD "speakers" field right. Thanks, Fengguang --- support writing to the ELD proc file Allow users to fix quicks of ELD ROMs by writing new values to the ELD proc interface. The format is one or more lines of "name hex_value". Users can add/remove/modify up to 32 SAD(Short Audio Descriptor) entries. Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/hda/hda_eld.c | 56 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) --- sound-2.6.orig/sound/pci/hda/hda_eld.c +++ sound-2.6/sound/pci/hda/hda_eld.c @@ -500,6 +500,59 @@ static void hdmi_print_eld_info(struct s hdmi_print_sad_info(i, e->sad + i, buffer); } +static void hdmi_write_eld_item(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct hdmi_eld *e = entry->private_data; + char line[64]; + char name[64]; + char *sname; + long long val; + int n; + + while (!snd_info_get_line(buffer, line, sizeof(line))) { + if (sscanf(line, "%s %llx", name, &val) != 2) + continue; + if (!strcmp(name, "connection_type")) + e->conn_type = val; + else if (!strcmp(name, "port_id")) + e->port_id = val; + else if (!strcmp(name, "support_hdcp")) + e->support_hdcp = val; + else if (!strcmp(name, "support_ai")) + e->support_ai = val; + else if (!strcmp(name, "audio_sync_delay")) + e->aud_synch_delay = val; + else if (!strcmp(name, "speakers")) + e->spk_alloc = val; + else if (!strcmp(name, "sad_count")) + e->sad_count = val; + else if (!strncmp(name, "sad", 3)) { + sname = name + 4; + n = name[3] - '0'; + if (name[4] >= '0' && name[4] <= '9') { + sname++; + n = 10 * n + name[4] - '0'; + } + if (n < 0 || n > 31) /* double the CEA limit */ + continue; + if (!strcmp(sname, "_coding_type")) + e->sad[n].format = val; + else if (!strcmp(sname, "_channels")) + e->sad[n].channels = val; + else if (!strcmp(sname, "_rates")) + e->sad[n].rates = val; + else if (!strcmp(sname, "_bits")) + e->sad[n].sample_bits = val; + else if (!strcmp(sname, "_max_bitrate")) + e->sad[n].max_bitrate = val; + if (n >= e->sad_count) + e->sad_count = n + 1; + } + } +} + + int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld) { char name[32]; @@ -512,6 +565,9 @@ int snd_hda_eld_proc_new(struct hda_code return err; snd_info_set_text_ops(entry, eld, hdmi_print_eld_info); + entry->c.text.write = hdmi_write_eld_item; + entry->mode |= S_IWUSR; + return 0; }

2 2

[alsa-devel] PATCH] hda: Add STAC_DELL_M4_3 quirk
by Matthew Ranostay 21 Nov '08

21 Nov '08

Added STAC_DELL_M4_3 quirk for Dell systems, also reorganized the board config switch to assign number of digital muxes, microphones, and SPDIF muxes via the PCI quirk defined. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/Documentation/ALSA-Configuration.txt b/Documentation/ALSA-Configuration.txt index 3ab5fb1..8b99a22 100644 --- a/Documentation/ALSA-Configuration.txt +++ b/Documentation/ALSA-Configuration.txt @@ -1076,6 +1076,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ref Reference board dell-m4-1 Dell desktops dell-m4-2 Dell desktops + dell-m4-3 Dell desktops STAC92HD73* ref Reference board diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index e0298df..f1a0f13 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -83,6 +83,7 @@ enum { STAC_92HD71BXX_REF, STAC_DELL_M4_1, STAC_DELL_M4_2, + STAC_DELL_M4_3, STAC_HP_M4, STAC_92HD71BXX_MODELS }; @@ -1689,10 +1690,17 @@ static unsigned int dell_m4_2_pin_configs[11] = { 0x40f000f0, 0x044413b0, 0x044413b0, }; +static unsigned int dell_m4_3_pin_configs[11] = { + 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, + 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0, + 0x40f000f0, 0x044413b0, 0x044413b0, +}; + static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, [STAC_DELL_M4_1] = dell_m4_1_pin_configs, [STAC_DELL_M4_2] = dell_m4_2_pin_configs, + [STAC_DELL_M4_3] = dell_m4_3_pin_configs, [STAC_HP_M4] = NULL, }; @@ -1700,6 +1708,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", [STAC_DELL_M4_2] = "dell-m4-2", + [STAC_DELL_M4_3] = "dell-m4-3", [STAC_HP_M4] = "hp-m4", }; @@ -1731,6 +1740,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "unknown Dell", STAC_DELL_M4_2), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0264, "unknown Dell", STAC_DELL_M4_2), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02aa, + "unknown Dell", STAC_DELL_M4_3), {} /* terminator */ }; @@ -4650,14 +4661,21 @@ again: switch (spec->board_config) { case STAC_HP_M4: - spec->num_dmics = 0; - spec->num_smuxes = 0; - spec->num_dmuxes = 0; - /* enable internal microphone */ stac92xx_set_config_reg(codec, 0x0e, 0x01813040); stac92xx_auto_set_pinctl(codec, 0x0e, AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); + /* fallthru */ + case STAC_DELL_M4_2: + spec->num_dmics = 0; + spec->num_smuxes = 0; + spec->num_dmuxes = 0; + break; + case STAC_DELL_M4_1: + case STAC_DELL_M4_3: + spec->num_dmics = 1; + spec->num_smuxes = 0; + spec->num_dmuxes = 0; break; default: spec->num_dmics = STAC92HD71BXX_NUM_DMICS;

2 1

[alsa-devel] [PATCH] hda: modify monitor name to be consistent with other ELD proc items
by Wu Fengguang 21 Nov '08

21 Nov '08

Rename "monitor name" to "monitor_name" to conform with the keyword style. Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/hda/hda_eld.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- sound-2.6.orig/sound/pci/hda/hda_eld.c +++ sound-2.6/sound/pci/hda/hda_eld.c @@ -477,7 +477,7 @@ static void hdmi_print_eld_info(struct s char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; int i; - snd_iprintf(buffer, "monitor name\t\t%s\n", e->monitor_name); + snd_iprintf(buffer, "monitor_name\t\t%s\n", e->monitor_name); snd_iprintf(buffer, "connection_type\t\t%s\n", eld_connection_type_names[e->conn_type]); snd_iprintf(buffer, "eld_version\t\t[0x%x] %s\n", e->eld_ver,

2 1

Re: [alsa-devel] [RFC][PATCH] ELD routines and proc interface
by Wu Fengguang 21 Nov '08

21 Nov '08

Hi Shane, On Thu, Nov 20, 2008 at 12:06:06PM -0800, Shane W wrote: > Hi, > > Ok, it appears to be the kernel I was using. Going to a > 2.6.27 kernel rather than 2.6.28-rc fixed the segfault. So > now I get ELD data though 5.1 PCM isn't quite there yet. > Here's the ELD info. Thanks. > HDMI intrinsic event: PD=1 ELDV=1 > detected monitor Haier TV*** at connection type HDMI > supports coding type LPCM: channels = 2, rates = 44100 > 48000 88200 176400 192000 384000, bits = > supports coding type LPCM: channels = 8, rates = 44100 > 48000 88200 176400 192000 384000, bits = It reveals bugs in showing the bits field. Will fix it in a following email. > supports coding type AC-3: channels = 6, rates = 44100 > 48000 88200, max bitrate = 640000 > supports coding type DTS: channels = 7, rates = 44100 48000 > 88200 176400 192000, max bitrate = 1536000 > supports coding type DSD (One Bit Audio): channels = 6, > rates = 48000 It's weird that DTS supports 7 channels while DSD supports 6. DTS is simple the compressed form of DSD. > set status page addr 0x08360000 > [drm:i915_getparam] *ERROR* Unknown parameter 5 > > shane@chintoka:~$ cat /proc/asound/Intel/eld#3 > monitor name Haier TV*** > connection_type HDMI > eld_version [0x2] CEA-861D or below > edid_version [0x1] CEA-861 > manufacture_id 0x4148 > product_id 0xb01 > port_id 0x0 > support_hdcp 0 > support_ai 0 > audio_sync_delay 0 > speakers [0x0] > sad_count 8 > sad0_coding_type [0x1] LPCM > sad0_channels 2 > sad0_rates [0x1ee0] 44100 48000 88200 176400 > 192000 384000 > sad0_bits [0xe0000] > sad1_coding_type [0x1] LPCM > sad1_channels 8 > sad1_rates [0x1ee0] 44100 48000 88200 176400 > 192000 384000 > sad1_bits [0xe0000] > sad2_coding_type [0x2] AC-3 > sad2_channels 6 > sad2_rates [0xe0] 44100 48000 88200 > sad2_max_bitrate 640000 > sad3_coding_type [0x7] DTS > sad3_channels 7 > sad3_rates [0x6e0] 44100 48000 88200 176400 > 192000 > sad3_max_bitrate 1536000 > sad4_coding_type [0x9] DSD (One Bit Audio) > sad4_channels 6 > sad4_rates [0x40] 48000 > sad5_coding_type [0x0] undefined > sad5_channels 1 > sad5_rates [0x0] > sad6_coding_type [0x0] undefined > sad6_channels 1 > sad6_rates [0x0] > sad7_coding_type [0x0] undefined > sad7_channels 1 > sad7_rates [0x0] I wonder why the monitor report these zero valued SADs? > The speakers 0 line is a bit confusing, not sure if that's Yes it's unexpected. Do you know its real speaker numbers and allocations? Does it provide some number of line-out ports? > what's doing it. I am using: > aplay 51test.wav > > which I have put here: > http://www.csy.ca/~shane/51test.wav I hear only "front left" and "front right" in my T61 :-) Thanks, Fengguang --- > > Shane > On Thu, Nov 20, 2008 at 09:02:04AM +0800, Wu Fengguang wrote: > > Thanks Shane! > > > > Ma Ling: do you have a quick fix for it? Or shall I look into it? > > > > Thank you, > > Fengguang > > > > On Wed, Nov 19, 2008 at 12:02:01PM -0800, Shane W wrote: > > > On Wed, Nov 19, 2008 at 05:39:40PM +0800, Wu Fengguang wrote: > > > > I managed to update the last working Xorg ELD patches to the latest > > > > git tree, and they compile flawlessly. The attached 2 patches are for > > > > xf86-video-intel/ and xserver/ respectively. > > > > > > They do compile but I can't get them to run. Here's the > > > Xorg backtrace when starting. I'm having trouble > > > generating a core but I'll work on getting gdb output if > > > needed. > > > > > > Backtrace: > > > 0: /opt/gfx-test/bin/Xorg(xorg_backtrace+0x26) [0x4ebbb6] > > > 1: /opt/gfx-test/bin/Xorg(xf86SigHandler+0x39) [0x495b49] > > > 2: /lib/libc.so.6 [0x7f2d1648bf60] > > > 3: > > > /opt/gfx-test/lib/xorg/modules/drivers//intel_drv.so(gen4_render_state_cleanup+0x1c) > > > [0x7f2d14ebab9c] > > > 4: /opt/gfx-test/lib/xorg/modules/drivers//intel_drv.so > > > [0x7f2d14e92dfd] > > > 5: /opt/gfx-test/bin/Xorg(AbortDDX+0x8d) [0x4690fd] > > > 6: /opt/gfx-test/bin/Xorg(AbortServer+0x1d) [0x4f505d] > > > 7: /opt/gfx-test/bin/Xorg(LogVMessageVerb+0) [0x4f5700] > > > 8: > > > /opt/gfx-test/lib/xorg/modules/drivers//intel_drv.so(i830_bind_all_memory+0x102) > > > [0x7f2d14e99592] > > > 9: /opt/gfx-test/lib/xorg/modules/drivers//intel_drv.so > > > [0x7f2d14e93ad0] > > > 10: /opt/gfx-test/lib/xorg/modules/drivers//intel_drv.so > > > [0x7f2d14e950f8] > > > 11: /opt/gfx-test/bin/Xorg(AddScreen+0x1c6) [0x431086] > > > 12: /opt/gfx-test/bin/Xorg(InitOutput+0x241) [0x469ae1] > > > 13: /opt/gfx-test/bin/Xorg(main+0x205) [0x431795] > > > 14: /lib/libc.so.6(__libc_start_main+0xe6) [0x7f2d164781a6] > > > 15: /opt/gfx-test/bin/Xorg [0x430dd9] > > > > > > It seems to be the xserver patch that's doing it. When they > > > are both unapplied, X starts fine, when the Xserver patch > > > is applied, we segfault, when the intel is applied alone, > > > it starts fine and when both are applied the segfault > > > returns. > > > > > > Shane

2 2

[alsa-devel] [PATCH 1/1] [ALSA] ASoc: TLV320AIC23B Cleanup
by Troy Kisky 21 Nov '08

21 Nov '08

Use #ifdef DEBUG to print actual sample rates. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index a95b538..8a9ef21 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -331,10 +331,11 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) adc_h = need_adc + (need_adc >> 5); dac_l = need_dac - (need_dac >> 5); dac_h = need_dac + (need_dac >> 5); - for (i = 0; i < 4; i++) { + for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) { int base = mclk / bosr_usb_divisor_table[i]; int mask = sr_valid_mask[i]; - for (j = 0; j < 16; j++, mask >>= 1) { + for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table); + j++, mask >>= 1) { int adc; int dac; int score; @@ -364,21 +365,22 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT); } -static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, +#ifdef DEBUG +static void get_current_sample_rates(int sample_rate_control, int mclk, u32 *sample_rate_adc, u32 *sample_rate_dac) { - int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); - int sr = (src >> 2) & 0x0f; - int val = (mclk / bosr_usb_divisor_table[src & 3]); + int sr = (sample_rate_control >> 2) & 0x0f; + int val = (mclk / bosr_usb_divisor_table[sample_rate_control & 3]); int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; int dac = (val * sr_dac_mult_table[sr]) / SR_MULT; - if (src & TLV320AIC23_CLKIN_HALF) { + if (sample_rate_control & TLV320AIC23_CLKIN_HALF) { adc >>= 1; dac >>= 1; } *sample_rate_adc = adc; *sample_rate_dac = dac; } +#endif static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, u32 sample_rate_adc, u32 sample_rate_dac) @@ -391,12 +393,14 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, return -EINVAL; } tlv320aic23_write(codec, TLV320AIC23_SRATE, data); - if (1) { +#ifdef DEBUG + { int adc, dac; - get_current_sample_rates(codec, mclk, &adc, &dac); + get_current_sample_rates(data, mclk, &adc, &dac); printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n", adc, dac, data); } +#endif return 0; } -- 1.5.4.3

2 2

[alsa-devel] [PATCH] hda: Add STAC_DELL_M4_3 quirk
by Matthew Ranostay 21 Nov '08

21 Nov '08

Add PCI quirk STAC_DELL_M4_3 for 92hd71bxx based laptops. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index e0298df..dab6013 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -83,6 +83,7 @@ enum { STAC_92HD71BXX_REF, STAC_DELL_M4_1, STAC_DELL_M4_2, + STAC_DELL_M4_3, STAC_HP_M4, STAC_92HD71BXX_MODELS }; @@ -1689,10 +1690,17 @@ static unsigned int dell_m4_2_pin_configs[11] = { 0x40f000f0, 0x044413b0, 0x044413b0, }; +static unsigned int dell_m4_3_pin_configs[11] = { + 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, + 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0, + 0x40f000f0, 0x044413b0, 0x044413b0, +}; + static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, [STAC_DELL_M4_1] = dell_m4_1_pin_configs, [STAC_DELL_M4_2] = dell_m4_2_pin_configs, + [STAC_DELL_M4_3] = dell_m4_3_pin_configs, [STAC_HP_M4] = NULL, }; @@ -1700,6 +1708,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", [STAC_DELL_M4_2] = "dell-m4-2", + [STAC_DELL_M4_3] = "dell-m4-3", [STAC_HP_M4] = "hp-m4", }; @@ -1731,6 +1740,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "unknown Dell", STAC_DELL_M4_2), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0264, "unknown Dell", STAC_DELL_M4_2), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02aa, + "unknown Dell", STAC_DELL_M4_3), {} /* terminator */ }; @@ -4650,14 +4661,20 @@ again: switch (spec->board_config) { case STAC_HP_M4: - spec->num_dmics = 0; - spec->num_smuxes = 0; - spec->num_dmuxes = 0; - /* enable internal microphone */ stac92xx_set_config_reg(codec, 0x0e, 0x01813040); stac92xx_auto_set_pinctl(codec, 0x0e, AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); + case STAC_DELL_M4_2: + spec->num_dmics = 0; + spec->num_smuxes = 0; + spec->num_dmuxes = 0; + break; + case STAC_DELL_M4_1: + case STAC_DELL_M4_3: + spec->num_dmics = 1; + spec->num_smuxes = 0; + spec->num_dmuxes = 0; break; default: spec->num_dmics = STAC92HD71BXX_NUM_DMICS;

2 4

Re: [alsa-devel] [Intel-gfx] [RFC][PATCH] ELD routines and proc interface
by Wu Fengguang 21 Nov '08

21 Nov '08

On Thu, Nov 20, 2008 at 05:46:49PM -0800, Shane W wrote: > On Fri, Nov 21, 2008 at 09:36:55AM +0800, Wu Fengguang wrote: > > > > > supports coding type AC-3: channels = 6, rates = 44100 > > > 48000 88200, max bitrate = 640000 > > > supports coding type DTS: channels = 7, rates = 44100 48000 > > > 88200 176400 192000, max bitrate = 1536000 > > > supports coding type DSD (One Bit Audio): channels = 6, > > > rates = 48000 > > > > It's weird that DTS supports 7 channels while DSD supports 6. > > DTS is simple the compressed form of DSD. > > Hmm, I thought DTS was just another way of compressing PCM > whereas DSD is the format used on SACD disks, IE 6 channel. > http://en.wikipedia.org/wiki/Direct_Stream_Digital Well my source is a bit different ;-) http://en.wikipedia.org/wiki/Super_Audio_CD DST To reduce the space and bandwidth requirements of DSD (2.8 Mbit/s per channel), a lossless data compression method called Direct Stream Transfer (DST) is used — DST compression is compulsory for multi-channel regions and optional for stereo regions. This typically compresses by a factor of between two and three, allowing a disc to contain 80 minutes of both 2-channel and 5.1-channel sound. > > > The speakers 0 line is a bit confusing, not sure if that's > > > > Yes it's unexpected. Do you know its real speaker numbers and > > allocations? Does it provide some number of line-out ports? > > There are six speakers connected, fl, ct, fr, rl, rr and > lfe. Takashi, I'd suggest to support writing to the ELD proc interface to alter the internal ELD struct. This could let users debug/fix quicks conveniently. But sure these fixes should eventually be incorporated into the kernel so that it just works. > > > what's doing it. I am using: > > > aplay 51test.wav > > > > > > which I have put here: > > > http://www.csy.ca/~shane/51test.wav > > > > I hear only "front left" and "front right" in my T61 :-) > > Yeah, you should get: > Front left > Center > Rear left > Rear right > and a little boom from the sub > > If you encode 51test.wav to ac3 and stream through hdmi > directly, it does this properly. OK, so 5.1 AC3 audio plays properly on HDMI? Thanks, Fengguang

1 0

[alsa-devel] ASoC, TWL4030 Register defaults
by naveen krishna ch 20 Nov '08

20 Nov '08

Hi alsa devels, Thanks for your suggestions. Here is the patch for the register default values. The defaults are chosen for minimal gain settings of the input and output devices of the CODEC. If asked for i can send the details of each register. I will format and send the other patches soon. The patch is as follows: [PATCH] Register default values for gain, path selection for various input and output devices From: chnaveenkrishna <chnaveen(a)chnaveen.mistral.in> To: Date: Today 04:49:50 pm --- sound/soc/codecs/twl4030.c | 28 ++++++++++++++-------------- 1 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ee2f0d3..3bc42b3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -45,8 +45,8 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0xc3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ - 0x24, /* REG_ANAMICL (0x5) */ - 0x04, /* REG_ANAMICR (0x6) */ + 0xb0, /* REG_ANAMICL (0x5) */ + 0x10, /* REG_ANAMICR (0x6) */ 0x0a, /* REG_AVADC_CTL (0x7) */ 0x00, /* REG_ADCMICSEL (0x8) */ 0x00, /* REG_DIGMIXING (0x9) */ @@ -67,22 +67,22 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_ARX2VTXPGA (0x18) */ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ - 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ - 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x2b, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x2b, /* REG_ARXR2_APGA_CTL (0x1C) */ 0x00, /* REG_ATX2ARXPGA (0x1D) */ 0x00, /* REG_BT_IF (0x1E) */ 0x00, /* REG_BTPGA (0x1F) */ 0x00, /* REG_BTSTPGA (0x20) */ - 0x00, /* REG_EAR_CTL (0x21) */ - 0x24, /* REG_HS_SEL (0x22) */ - 0x0a, /* REG_HS_GAIN_SET (0x23) */ + 0x20, /* REG_EAR_CTL (0x21) */ + 0x00, /* REG_HS_SEL (0x22) */ + 0x05, /* REG_HS_GAIN_SET (0x23) */ 0x00, /* REG_HS_POPN_SET (0x24) */ - 0x00, /* REG_PREDL_CTL (0x25) */ - 0x00, /* REG_PREDR_CTL (0x26) */ - 0x00, /* REG_PRECKL_CTL (0x27) */ - 0x00, /* REG_PRECKR_CTL (0x28) */ - 0x00, /* REG_HFL_CTL (0x29) */ - 0x00, /* REG_HFR_CTL (0x2A) */ + 0x20, /* REG_PREDL_CTL (0x25) */ + 0x20, /* REG_PREDR_CTL (0x26) */ + 0x20, /* REG_PRECKL_CTL (0x27) */ + 0x20, /* REG_PRECKR_CTL (0x28) */ + 0x1f, /* REG_HFL_CTL (0x29) */ + 0x1f, /* REG_HFR_CTL (0x2A) */ 0x00, /* REG_ALC_CTL (0x2B) */ 0x00, /* REG_ALC_SET1 (0x2C) */ 0x00, /* REG_ALC_SET2 (0x2D) */ @@ -112,7 +112,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VIBRA_CTL (0x45) */ 0x00, /* REG_VIBRA_SET (0x46) */ 0x00, /* REG_VIBRA_PWM_SET (0x47) */ - 0x00, /* REG_ANAMIC_GAIN (0x48) */ + 0x24, /* REG_ANAMIC_GAIN (0x48) */ 0x00, /* REG_MISC_SET_2 (0x49) */ }; -- 1.5.3.4 -- Thanks, (: Naveen Krishna Ch :)

4 7

[alsa-devel] [PATCH 1/2] ALSA: ASoC: ssm2602, wm8903: Fix priv substreams refs
by Karl Beldan 20 Nov '08

20 Nov '08

ssm2602.c and wm8903.c read: "The DAI has shared clocks so if we already have a playback or capture going then constrain this substream to match it." Let ssm2602.c handle this the same way as wm8903.c and highlight possible race pitfall in both files. This fixes subsequent failures of snd_pcm_hw_constraints_complete after closure of a stream. NOTE: - The ssm2602 allows pairs of non-matching PB/REC rates. - This is a fix for less evil: The logic is flawed (e.g. the slave might startup before the master's rate and sample_bits are set). Signed-off-by: Karl Beldan <karl.beldan(a)gmail.com> --- sound/soc/codecs/ssm2602.c | 20 ++++++++++++++++++++ sound/soc/codecs/wm8903.c | 2 ++ 2 files changed, 22 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 44ef0da..abc804e 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -292,9 +292,15 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; int i = get_coeff(ssm2602->sysclk, params_rate(params)); + if (substream == ssm2602->slave_substream) { + dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); + return 0; + } + /*no match is found*/ if (i == ARRAY_SIZE(coeff_div)) return -EINVAL; @@ -330,13 +336,21 @@ static int ssm2602_startup(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; /* The DAI has shared clocks so if we already have a playback or * capture going then constrain this substream to match it. + * TODO: the ssm2602 allows pairs of non-matching PB/REC rates + * FIXME: this is racy, e.g. the slave might startup before the + * master's rate and sample_bits are set. */ if (ssm2602->master_substream) { master_runtime = ssm2602->master_substream->runtime; + dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + master_runtime->sample_bits, + master_runtime->rate); + snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, master_runtime->rate, @@ -370,9 +384,15 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; /* deactivate */ if (!codec->active) ssm2602_write(codec, SSM2602_ACTIVE, 0); + + if (ssm2602->master_substream == substream) + ssm2602->master_substream = ssm2602->slave_substream; + + ssm2602->slave_substream = NULL; } static int ssm2602_mute(struct snd_soc_dai *dai, int mute) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ce40d78..ebe0c84 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1273,6 +1273,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream) /* The DAI has shared clocks so if we already have a playback or * capture going then constrain this substream to match it. + * FIXME: this is racy, e.g. the slave might startup before the + * master's rate and sample_bits are set. */ if (wm8903->master_substream) { master_runtime = wm8903->master_substream->runtime; -- 1.6.0.3.523.g304d0

2 1

[alsa-devel] [PATCH 2/2] ALSA: ASoC: ssm2602: Update supported stream formats
by Karl Beldan 20 Nov '08

20 Nov '08

Signed-off-by: Karl Beldan <karl.beldan(a)gmail.com> --- sound/soc/codecs/ssm2602.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index abc804e..714187a 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -516,6 +516,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ SNDRV_PCM_RATE_96000) +#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + struct snd_soc_dai ssm2602_dai = { .name = "SSM2602", .playback = { @@ -523,13 +526,13 @@ struct snd_soc_dai ssm2602_dai = { .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .ops = { .startup = ssm2602_startup, .prepare = ssm2602_pcm_prepare, -- 1.6.0.3.523.g304d0

1 0

[alsa-devel] Kcontrols for ASoC TWL4030 CODEC
by naveen krishna ch 20 Nov '08

20 Nov '08

Here i submit the patch for the kcontrols for ASoC TWL4030 CODEC. This patch comes next to my first patch. There seems some MUA issue so i attached the patch.

2 3

[alsa-devel] Creative X-Fi Xtreme Audio (CA0110 chip)
by Manuel C. 20 Nov '08

20 Nov '08

Hi, I want to bring to your attention the Creative X-Fi Xtreme Audio card with CA0110 chip, it is also bundled with some MSI motherboards. CA0110 web page: http://www.creative.com/oem/products/chips/CA0110.asp >From that web page: "CA0110 is a PCI Express audio chip using the CA0111 audio mode or High Definition Audio mode. [...] CA0106 mode - Same as original CA0111." So it should be a PCI Express version of the CA0111 chip (and CA0111 could be similar to the CA0106 chip). The CA0110 chip isn't even listed in the Creative products matrix on the wiki. Is a CA0110 driver under development? Thank you.

2 1

[alsa-devel] (no subject)
by Nitin Mahajan 20 Nov '08

20 Nov '08

unsubscribe alsa-devel New Email names for you! Get the Email name you've always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/

1 0

[PATCH] Kcontrols for gain selection and enabling, disabling the
by chnaveenkrishna 20 Nov '08

20 Nov '08

ADC, DAC-2 --- sound/soc/codecs/twl4030.c | 61 +++++++++++++++++++++++++++++++++++++++---- 1 files changed, 55 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 3bc42b3..f258ea2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -190,12 +190,61 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } static const struct snd_kcontrol_new twl4030_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", - TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 0, 127, 0), - SOC_DOUBLE_R("Capture Volume", - TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, - 0, 127, 0), + + /* Master Playback Volume Controls */ + SOC_DOUBLE_R("Master PLayback Course Gain", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 6, 3, 0), + + SOC_DOUBLE_R("Master Playback Fine Gain", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 0, 63, 0), + + /* Master Capture Gain controls */ + SOC_DOUBLE_R("Master Capture Gain", + TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, + 0, 31, 0), + + /* Headset volume controls */ + SOC_DOUBLE("Headset Volume", + TWL4030_REG_HS_GAIN_SET, 2, 0, 3, 0), + + /* Ear Piece volume controls */ + SOC_SINGLE("Earpiece Volume", + TWL4030_REG_EAR_CTL, 4, 3, 0), + + /* Internal Class D Speaker volume controls */ + SOC_DOUBLE_R("Internal Speaker Volume", + TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, + 3, 17, 0), + + /* External Class D Speaker volume controls */ + SOC_DOUBLE_R("External Speaker Volume", + TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL, + 4, 3, 0), + + /* Predriver Carkit volume controls */ + SOC_DOUBLE_R("Pre Car-Kit Volume", + TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL, + 4, 3, 0), + + SOC_DOUBLE("Main + Sub Mic Capture Gain", + TWL4030_REG_ANAMIC_GAIN, 3 , 0, 5, 0), + + /* Loop gain controls*/ + SOC_DOUBLE("Loop Gain", TWL4030_REG_ATX2ARXPGA, + 3 , 0, 7, 0), + + /* Left and Right ADC Switches */ + SOC_SINGLE("ADCL", TWL4030_REG_AVADC_CTL, 3, 1, 0), + + SOC_SINGLE("ADCR", TWL4030_REG_AVADC_CTL, 1, 1, 0), + + /* Left and Right DAC-2 Switches */ + SOC_SINGLE("DACL2", TWL4030_REG_AVDAC_CTL, 3, 1, 0), + + SOC_SINGLE("DACR2", TWL4030_REG_AVDAC_CTL, 2, 1, 0), + }; 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1 0

[alsa-devel] Alsa "short read"
by Pierre Vanhoucke 20 Nov '08

20 Nov '08

Hi all, Is there anybody out there who can help me to answer the following questions: I have converted an application called Linrad ( http://www.nitehawk.com/sm5bsz/linuxdsp/linrad.htm ) from OSS to ALSA. Linrad is a remarkable DSP application used worldwide by many radio-hams. I am facing hereby a problem with a "short read" : An ALSA device is opened in blocking mode: ********************************************* err=snd_pcm_open(&rx_ad_handle, rx_sndin_plughw_pcm_name, SND_PCM_STREAM_CAPTURE, 0); ********************************************* When I subsequently read the device with following statement I sometimes get a "short read": ******************************************** err=snd_pcm_readi(rx_ad_handle,rxin_isho, rxad.block_frames); if((unsigned int)err != rxad.block_frames) { if(err == -EPIPE) // EPIPE, try recovery { err= snd_pcm_recover(rx_ad_handle, err, 0); if(err < 0)lirerr(1108); // recovery failed } else { if(err < 0) { lirerr(1113); // catch all } else { lirerr(1116); // short read } } } } ************************************************** Here are my questions: 1. How is it possible to get a "short read" in blocking mode ? I would think that in blocking mode the control is only returned after the read is completed. 2. What are the possible causes for a "short read" ? 3. What can be done to avoid a "short read" ? 4. How am I supposed to recover from a "short read"? Best regards, Pierre Vanhoucke/ ON5GN

1 0

[alsa-devel] [PATCH] azx_probe() cleanup
by Wu Fengguang 20 Nov '08

20 Nov '08

Replace 5 free-and-return-err blocks with goto-out-free ones. This makes the main logic more outstanding. Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 33 +++++++++++++-------------------- 1 file changed, 13 insertions(+), 20 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_intel.c +++ sound-2.6/sound/pci/hda/hda_intel.c @@ -2317,40 +2317,30 @@ static int __devinit azx_probe(struct pc } err = azx_create(card, pci, dev, pci_id->driver_data, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; card->private_data = chip; /* create codec instances */ err = azx_codec_create(chip, model[dev], probe_mask[dev]); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; /* create PCM streams */ err = snd_hda_build_pcms(chip->bus); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; /* create mixer controls */ err = azx_mixer_create(chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; snd_card_set_dev(card, &pci->dev); err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; pci_set_drvdata(pci, card); chip->running = 1; @@ -2359,6 +2349,9 @@ static int __devinit azx_probe(struct pc dev++; return err; +out_free: + snd_card_free(card); + return err; } static void __devexit azx_remove(struct pci_dev *pci)

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: Fix audio stall when doing full duplex
by naresh＠ti.com 20 Nov '08

20 Nov '08

From: Naresh Medisetty <naresh(a)ti.com> Fix concurrent capture/playback issue. The issue is caused by re-initialization of control registers used specifically for capture or playback in both capture and playback operations. Signed-off-by: Steve Chen <schen(a)mvista.com> Naresh Medisetty <naresh(a)ti.com> --- sound/soc/davinci/davinci-i2s.c | 31 +++++++++++++++++++------------ 1 files changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 8c1bf87..11c20d0 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -295,10 +295,14 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, u32 w; /* general line settings */ - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, - DAVINCI_MCBSP_SPCR_RINTM(3) | - DAVINCI_MCBSP_SPCR_XINTM(3) | - DAVINCI_MCBSP_SPCR_FREE); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + w |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } else { + w |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); @@ -329,16 +333,19 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + } else { + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + } return 0; } -- 1.5.4.1

3 2

[alsa-devel] PCM3008 ALSA Soc driver and Lyrtech SFFSDR board audio support (v2)
by Hugo Villeneuve 19 Nov '08

19 Nov '08

Reworked and splitted patch based on Mark Brown comments. Mark, thank-you for your comments. I implemented almost all of your suggestions, except for the de-emphasis control because I don't really know how I can export (and test) that interface. Like you said, maybe we or someone else can do it later. Hugo Villeneuve

2 5

[alsa-devel] [PATCH 0/6] ASoC updates
by Mark Brown 19 Nov '08

19 Nov '08

The following changes since commit 72f2b894455775b980a5ac7ae70ab560b3d3d247: Mark Brown (1): ASoC: Move uda134x_codec.h to uda134x.h are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Arun KS (1): ASoC: Add support for omap2evm board Hugo Villeneuve (2): ASoC: Add PCM3008 ALSA SoC driver ASoC: Add driver for the Lyrtech SFFSDR board Mark Brown (3): ASoC: Remove unused snd_soc_machine_config declaration ASoC: s3c24xx_uda134x DAI accessor functions and static cleanup ASoC: Convert blackfin machines to use DAI accessor functions include/sound/soc.h | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 6 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm3008.c | 201 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/pcm3008.h | 25 +++++ sound/soc/davinci/Kconfig | 10 ++ sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-sffsdr.c | 156 +++++++++++++++++++++++++++ sound/soc/omap/Kconfig | 9 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap2evm.c | 150 ++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c24xx_uda134x.c | 27 +++--- 14 files changed, 578 insertions(+), 19 deletions(-) create mode 100644 sound/soc/codecs/pcm3008.c create mode 100644 sound/soc/codecs/pcm3008.h create mode 100644 sound/soc/davinci/davinci-sffsdr.c create mode 100644 sound/soc/omap/omap2evm.c

2 7

[alsa-devel] omap3evm asoc
by naveen krishna ch 19 Nov '08

19 Nov '08

Hello Alsa devels, Is there any previous work on audio ASOC for OMAP3EVM. I have started working on OMAP3EVM + TWL4030 combination and audio is working. -- Thanks, (: Naveen Krishna Ch :)

3 2

[alsa-devel] [PATCH 1/1] ASoC: Add support for omap2evm board
by Arun KS 19 Nov '08

19 Nov '08

This patch adds twl4030 audio support on omap2evm Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/Kconfig | 9 +++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap2evm.c | 150 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 161 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap2evm.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index cf40e42..6c56277 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -30,4 +30,13 @@ config SND_OMAP_SOC_OVERO help Say Y if you want to add support for SoC audio on the Gumstix Overo. +config SND_OMAP_SOC_OMAP2EVM + tristate "SoC Audio support for OMAP2EVM board" + depends on SND_OMAP_SOC && MACH_OMAP2EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the omap2evm board. + + diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index fefc9be..f5da3cc 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -9,8 +9,10 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o snd-soc-n810-objs := n810.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o +snd-soc-omap2evm-objs := omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o +obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c new file mode 100644 index 0000000..c376213 --- /dev/null +++ b/sound/soc/omap/omap2evm.c @@ -0,0 +1,150 @@ +/* + * omap2evm.c -- SoC audio machine driver for omap2evm board + * + * Author: Arun KS <arunks(a)mistralsolutions.com> + * + * Based on sound/soc/omap/overo.c by Steve Sakoman + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap2evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap2evm_ops = { + .hw_params = omap2evm_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap2evm_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap2evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_omap2evm = { + .name = "omap2evm", + .dai_link = &omap2evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap2evm_snd_devdata = { + .machine = &snd_soc_machine_omap2evm, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap2evm_snd_device; + +static int __init omap2evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap2evm()) { + pr_debug("Not omap2evm!\n"); + return -ENODEV; + } + printk(KERN_INFO "omap2evm SoC init\n"); + + omap2evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap2evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap2evm_snd_device, &omap2evm_snd_devdata); + omap2evm_snd_devdata.dev = &omap2evm_snd_device->dev; + *(unsigned int *)omap2evm_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(omap2evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap2evm_snd_device); + + return ret; +} +module_init(omap2evm_soc_init); + +static void __exit omap2evm_soc_exit(void) +{ + platform_device_unregister(omap2evm_snd_device); +} +module_exit(omap2evm_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC omap2evm"); +MODULE_LICENSE("GPL"); -- 1.5.6.3

2 1

[alsa-devel] Old OMAP mailing list
by Mark Brown 19 Nov '08

19 Nov '08

Please don't CC any ASoC patches to the linux-omap-open-source list - my understanding is that it should no longer be used, having been replaced by linux-omap(a)vger.kernel.org. Any posts CCed to it result in backscatter like this: On Tue, Nov 18, 2008 at 07:57:55AM -0600, linux-omap-open-source-owner(a)linux.omap.com wrote: > You are not allowed to post to this mailing list, and your message has > been automatically rejected. If you think that your messages are > being rejected in error, contact the mailing list owner at > linux-omap-open-source-owner(a)linux.omap.com. which seems a bit hostile. It'd be nice if this bounce mentioned the new list, FWIW.

2 2

Re: [alsa-devel] [RFC][PATCH] ELD routines and proc interface
by Wu Fengguang 19 Nov '08

19 Nov '08

On Tue, Nov 18, 2008 at 11:55:45PM -0800, Shane W wrote: > On Wed, Nov 19, 2008 at 03:17:57PM +0800, Wu Fengguang wrote: > > On Tue, Nov 18, 2008 at 11:11:35PM -0800, Shane W wrote: > > > On Thu, Nov 13, 2008 at 10:21:53AM +0800, Wu Fengguang wrote: > > > > Create hda_eld.c for ELD routines and proc interface. > > > > > Just testing this and your multichannel HDMI code over here > > > but it doesn't seem to get ELD data for my setup. Intel > > > g45-id board with an Yamaha RX-v1800 receiver which is HDMI > > > 1.3 capable. > > > > Do you run the latest xorg intel driver? Also dmesg should tell us > > something if anything goes wrong with ELD retrieving/parsing. > > Yes though Xorg wasn't running when I was doing the audio > tests. Pure console at that point but I fired up Xorg > which is latest git as of today but still no ELD Yeah sorry I missed one note. The xorg git tree only contains the audio output enabling patches, but not yet ELD transferring patches. (the ELD data flow is monitor => video driver => audio driver) I called for the Xorg ELD enabling patches from Ma Ling just after replying to your email. He should send the patches soon. > population. When Xorg starts, I get this in dmesg: > [drm] Initialized i915 1.6.0 20080730 on minor 0 > [drm:i915_gem_object_bind_to_gtt] *ERROR* GTT full, but LRU > list empty > [drm:i915_gem_object_pin] *ERROR* Failure to bind: > -12<6>HDMI intrinsic event: PD=1 ELDV=1 This last dmesg line is funny. The ELDV(i.e. ELD Valid) is true, but the video driver obviously has no ELD data prepared for us. Thanks, Fengguang

1 1

[alsa-devel] [PATCH 0/5] assorted HDMI cleanups
by Wu Fengguang 19 Nov '08

19 Nov '08

Hi Takashi, Here are 5 HDMI cleanup patches, some renames the code and others renames the user visible output messages. The new output goes like: dmesg [ 5742.169578] detected monitor <9d><80>t connection type Display Port [ 5742.169580] available speakers: FL/FR LFE FLC/FRC [ 5742.169582] supports coding type MLP (Dolby TrueHD): channels = 7, rates = 48000 176400 [ 5742.169585] supports coding type AC-3: channels = 1, rates = 48000 88200 192000, max bitrate = 1208000 [ 5742.169588] supports coding type AC-3: channels = 2, rates = 48000 384000, max bitrate = 960000 [ 5742.169591] supports coding type ATRAC: channels = 5, rates = 48000 88200 384000, max bitrate = 224000 /proc/asound/card0/eld\#3 monitor name �� connection_type Display Port eld_version [0xb] reserved edid_version [0x3] CEA-861-B, C or D manufacture_id 0xee5a product_id 0x923c port_id 0x18b263a03e17d1b5 support_hdcp 1 support_ai 1 audio_sync_delay 216 speakers [0x23] FL/FR LFE FLC/FRC sad_count 4 sad0_coding_type [0xc] MLP (Dolby TrueHD) sad0_channels 7 sad0_rates [0x240] 48000 176400 sad1_coding_type [0x2] AC-3 sad1_channels 1 sad1_rates [0x4c0] 48000 88200 192000 sad1_max_bitrate 1208000 sad2_coding_type [0x2] AC-3 sad2_channels 2 sad2_rates [0x1840] 48000 384000 sad2_max_bitrate 960000 sad3_coding_type [0x8] ATRAC sad3_channels 5 sad3_rates [0x8c0] 48000 88200 384000 sad3_max_bitrate 224000 Thanks, Fengguang --

2 6

[alsa-devel] Listing up to 8 PCM devices
by Gregoire ETIENNE 19 Nov '08

19 Nov '08

Hi, I'm currently writting a driver with up to 8 pcm devices, so I turned on "Dynamic device file minor numbers" option. But when I try to list devices through aplay or a record (which are calling ioctl with cmd SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE) I can only list devices from 0 to 7. This is due to this code : linux/sound/core/pcm.c l.64 ------------------------------ case SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE: { int device; if (get_user(device, (int __user *)arg)) return -EFAULT; mutex_lock(&register_mutex); device = device < 0 ? 0 : device + 1; while (device < SNDRV_PCM_DEVICES) { if (snd_pcm_search(card, device)) break; device++; } if (device == SNDRV_PCM_DEVICES) device = -1; mutex_unlock(&register_mutex); if (put_user(device, (int __user *)arg)) return -EFAULT; return 0; } ------------------------------ because SNDRV_PCM_DEVICES is defined to 8 and do not change even if I selected "Dynamic device file minor numbers" option. Is there anything I missed ?

2 1

[alsa-devel] Invitation to connect on LinkedIn
by Arun KS 19 Nov '08

19 Nov '08

LinkedIn ------------ ALSA, I'd like to add you to my professional network on LinkedIn. - Arun Learn more: https://www.linkedin.com/e/isd/409983693/itycjb3I/ ------------------------------------------ What is LinkedIn and why should you join? http://learn.linkedin.com/what-is-linkedin/ ------ (c) 2008, LinkedIn Corporation

1 1

Re: [alsa-devel] [RFC][PATCH] ELD routines and proc interface
by Wu Fengguang 19 Nov '08

19 Nov '08

On Tue, Nov 18, 2008 at 11:11:35PM -0800, Shane W wrote: > On Thu, Nov 13, 2008 at 10:21:53AM +0800, Wu Fengguang wrote: > > Create hda_eld.c for ELD routines and proc interface. > > Hi, Hi Shane, > Just testing this and your multichannel HDMI code over here > but it doesn't seem to get ELD data for my setup. Intel > g45-id board with an Yamaha RX-v1800 receiver which is HDMI > 1.3 capable. Do you run the latest xorg intel driver? Also dmesg should tell us something if anything goes wrong with ELD retrieving/parsing. > shane@chintoka:~$ cat /proc/asound/Intel/eld#3 > monitor name > connection_type HDMI > eld_version [0x0] reserved > edid_version [0x0] no CEA EDID Timing Extension > block present > manufacture_id 0x0 > product_id 0x0 > port_id 0x0 > support_hdcp 0 > support_ai 0 > audio_sync_delay 0 > speakers [0x0] > sad_count 0 > > 2ch audio is coming through just fine over HDMI though so > the connection itself is there. Yes. Obviously the ELD stuct has not been populated yet. Thanks, Fengguang

1 0

[alsa-devel] [PATCH 0/4] multi-channel HDMI audio support
by Wu Fengguang 19 Nov '08

19 Nov '08

Hi Takashi, This patchset attempts to support multi-channel HDMI audio. It contains two basic operations: - setup the CA(channel-allocation) byte in the audio infoframe - setup the converter channel to HDMI slot mapping Please read patch 3 for more detailed descriptions: [PATCH 1/4] hda: make standalone hdmi_fill_audio_infoframe() [PATCH 2/4] hda: make global snd_print_channel_allocation() [PATCH 3/4] hda: HDMI channel allocations for audio infoframe [PATCH 4/4] hda: HDMI channel mapping cleanups The current implementation only does minimal works that ensure multi-channel audio playing out. However the wrong speakers may be selected. We could elaborate the framework/policy if it proves to be a problem. Thank you, Fengguang --

2 6

[alsa-devel] [PATCH] hda: EAPD mute on suspend
by Matthew Ranostay 19 Nov '08

19 Nov '08

Moved support for EAPD mute on suspend from stac92hd71xx_suspend to the generic stac92xx_suspend function. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index e0298df..900d99f 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -4077,7 +4077,17 @@ static int stac92xx_resume(struct hda_codec *codec) codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); return 0; } -#endif + +static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct sigmatel_spec *spec = codec->spec; + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); + return 0; +} +#endif static struct hda_codec_ops stac92xx_patch_ops = { .build_controls = stac92xx_build_controls, @@ -4087,6 +4097,7 @@ static struct hda_codec_ops stac92xx_patch_ops = { .unsol_event = stac92xx_unsol_event, #ifdef SND_HDA_NEEDS_RESUME .resume = stac92xx_resume, + .suspend = stac92xx_suspend, #endif }; @@ -4498,14 +4509,8 @@ static int stac92hd71xx_resume(struct hda_codec *codec) static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state) { - struct sigmatel_spec *spec = codec->spec; - - stac92hd71xx_set_power_state(codec, AC_PWRST_D3); - if (spec->eapd_mask) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data & - ~spec->eapd_mask); - return 0; + stac92hd71xx_set_power_state(codec, AC_PWRST_D3); + return stac92xx_suspend(codec, state); }; #endif

2 1

[alsa-devel] HDA-emu v0.0.3
by Takashi Iwai 18 Nov '08

18 Nov '08

Hi, right now I updated hda-emu to version 0.0.3. hda-emu is a HD-audio emulation program that is hopefully useful for testing and debugging HD-audio drivers. This version contains a simple PCM test in addition to handling of control elements. The latest package is found at ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/hda-emu-0.0.3.tar.gz The git tree is git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git Any fixes, improvements and feedbacks are appreciated thanks, Takashi

1 0

[alsa-devel] [PATCH 0/17] ASoC updates
by Mark Brown 18 Nov '08

18 Nov '08

The following changes since commit 71cfc9028d762419ce4dea62b4afc9c32c4b4820: Mark Brown (1): ASoC: Add WM8728 codec driver are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Bryan Wu (1): ASoC: Blackfin: Simplify the MMAP_SUPPORT macros protected code Christian Pellegrin (2): ASoC: UDA134x codec driver ASoC: Machine driver for for s3c24xx with uda134x Cliff Cai (3): ASoC: AD1980 codec: add multi-channel function support ASoC: Blackfin: add multi-channel function support ASoC: Blackfin: Fix AD1980/1 build with MMAP support disabled Mark Brown (5): ASoC: Allow writes to uncached registers in WM8990 ASoC: Enable WM8990 ADC clocking workaround ASoC: Manage VMID mode for WM8990 ASoC: Build tlv320aic23 cleanly ASoC: Move uda134x_codec.h to uda134x.h Michael Hennerich (1): ASoC: Fix Blackfin AC97 DAI probe function return code Mike Frysinger (3): ASoC: Blackfin: updates Kconfig for SPORT ASoC: Blackfin: do not force TWI bus for ssm2602 codec ASoC: Blackfin: always set a default value for that GPIO range Naresh Medisetty (1): ASoC: DaVinci: Fix audio stall when doing full duplex Peter Ujfalusi (1): ASoC: Fix for master playback/capture volume range for TWL4030 codec include/sound/l3.h | 18 + include/sound/s3c24xx_uda134x.h | 14 + include/sound/uda134x.h | 26 ++ sound/soc/blackfin/Kconfig | 18 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 101 +++--- sound/soc/blackfin/bf5xx-ac97.c | 158 +++++---- sound/soc/blackfin/bf5xx-ac97.h | 35 ++- sound/soc/blackfin/bf5xx-sport.h | 2 +- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/ad1980.c | 19 +- sound/soc/codecs/l3.c | 91 +++++ sound/soc/codecs/tlv320aic23.c | 13 +- sound/soc/codecs/twl4030.c | 4 +- sound/soc/codecs/uda134x.c | 656 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/uda134x.h | 36 ++ sound/soc/codecs/wm8990.c | 23 +- sound/soc/codecs/wm8990.h | 4 +- sound/soc/davinci/davinci-i2s.c | 31 +- sound/soc/s3c24xx/Kconfig | 5 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c24xx_uda134x.c | 374 ++++++++++++++++++++ 22 files changed, 1488 insertions(+), 154 deletions(-) create mode 100644 include/sound/l3.h create mode 100644 include/sound/s3c24xx_uda134x.h create mode 100644 include/sound/uda134x.h create mode 100644 sound/soc/codecs/l3.c create mode 100644 sound/soc/codecs/uda134x.c create mode 100644 sound/soc/codecs/uda134x.h create mode 100644 sound/soc/s3c24xx/s3c24xx_uda134x.c

2 1

[alsa-devel] [PATCH 01/17] ASoC: Fix for master playback/capture volume range for TWL4030 codec
by Mark Brown 18 Nov '08

18 Nov '08

From: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> FGAIN for playback is in range of 0-0x3f, while for capture GAIN it is in the range of 0-0x1f. The original value of 128 (0x7f) would modify the CGAIN also for playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Acked-by: Steve Sakoman <steve(a)sakoman.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl4030.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c1893d2..c778eb4 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -192,10 +192,10 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 0, 127, 0), + 0, 0x3f, 0), SOC_DOUBLE_R("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, - 0, 127, 0), + 0, 0x1f, 0), }; /* add non dapm controls */ -- 1.5.6.5

1 16

[alsa-devel] (no subject)
by Bryan Wu 18 Nov '08

18 Nov '08

3 12

[alsa-devel] [PATCH 0/4] [RFC] multi-channel HDMI audio support
by Wu Fengguang 18 Nov '08

18 Nov '08

Hi all, This patchset attempts to support multi-channel HDMI audio. It contains two basic operations: - setup the CA(channel-allcation) byte in the audio infoframe - setup the converter channel to HDMI slot mapping Please check the last two patches for the detailed descriptions: [PATCH 1/4] hda: make standalone hdmi_fill_audio_infoframe() [PATCH 2/4] hda: make global snd_print_channel_allocation() [PATCH 3/4] hda: HDMI channel allocations for audio infoframe [PATCH 4/4] hda: HDMI channel mapping cleanups The current implementation only does minimal works that can ensure the multi-channel audio playing out. However the wrong speakers may be choosed, and I really need more information/knowledge/instructions from you to get it right. Thank you, Fengguang --

2 9

[alsa-devel] 5d5d5f43f (Samsung hda-intel auto-sensing) causes input regression
by Daniel Drake 18 Nov '08

18 Nov '08

Hi, Just wanted to bring to your attention: Commit 5d5d5f43f1b835c375de9bd270cce030d16e2871 causes a regression on some laptops: input no longer works for some users. > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Tue Feb 12 12:11:36 2008 +0100 > > [ALSA] hda-codec - Implement auto-mic jack sensing on Samsung laptops > > Implemented the auto-mic jack sensing for Samsung laptops with AD1986A > codec chip (model=laptop-eapd). > > The hardware uses pin 0x1d and 0x1f for the internal and external > mics, respectively. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Reported at: https://bugs.gentoo.org/show_bug.cgi?id=235732 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4228 The discussion around this patch appears to be here: http://thread.gmane.org/gmane.linux.alsa.devel/50989/focus=51457 Anything we can do to help solve this problem? Thanks, Daniel

2 1

[alsa-devel] Jack insertion
by naveen krishna ch 18 Nov '08

18 Nov '08

Hello Alsa Devels I am working on a OMAP3 + TWL4030 combination for audio (alsa soc) I could develop the ASOC drivers with limited kcontorls and DAPM widgets for the setup. Now my issue is is there any work regarding the Headset JACK insertion/detection. Earlier there was a proposal for a jack insertion layer. Can anyone suggest me in this regard. Thank You, (: Naveen Krishna Ch :)

3 5

[alsa-devel] [PATCH 001/002] snd-powermac: enable mic on iMac G4
by Risto Suominen 18 Nov '08

18 Nov '08

2 1

[alsa-devel] [PATCH 0/3] HDMI ELD routines and proc interface
by Wu Fengguang 18 Nov '08

18 Nov '08

Hi Takashi, Here are the patches for separated HDMI ELD routines and proc interface. They have undergone basic tests on a G45 board and stress test with the patch below. [PATCH 1/3] introduce snd_print_pcm_rates() [PATCH 2/3] create hda_eld.c for ELD routines and proc interface [PATCH 3/3] ELD proc interface for HDMI sinks Here are two sample outputs from the stress testing: monitor name connection_type 2-reserved eld_version [0x0] reserved edid_version [0x0] no CEA EDID Timing Extension block present manufacture_id 0x3c93 product_id 0xc5ac port_id 0x486d85d5a3bfe049 support_hdcp 1 support_ai 0 audio_sync_delay 152 speakers [0x6f] FL/FR LFE FC RL/RR FLC/FRC RLC/RRC sad_count 8 sad0_coding_type [0x0] undefined sad0_channels 0 sad0_sampling_rates [0x0] sad1_coding_type [0x0] undefined sad1_channels 0 sad1_sampling_rates [0x0] sad2_coding_type [0x0] undefined sad2_channels 0 sad2_sampling_rates [0x0] sad3_coding_type [0x0] undefined sad3_channels 0 sad3_sampling_rates [0x0] sad4_coding_type [0x0] undefined sad4_channels 0 sad4_sampling_rates [0x0] sad5_coding_type [0x0] undefined sad5_channels 0 sad5_sampling_rates [0x0] sad6_coding_type [0x0] undefined sad6_channels 0 sad6_sampling_rates [0x0] sad7_coding_type [0x0] undefined sad7_channels 0 sad7_sampling_rates [0x0] monitor name ��p connection_type Display Port eld_version [0x0] reserved edid_version [0x7] reserved manufacture_id 0x59c7 product_id 0x3c72 port_id 0x323d7444a5fcbdba support_hdcp 0 support_ai 1 audio_sync_delay 268 speakers [0x33] FL/FR LFE RC FLC/FRC sad_count 14 sad0_coding_type [0x9] DSD (1-bit audio) sad0_channels 5 sad0_sampling_rates [0xe40] 48000 176400 192000 384000 sad1_coding_type [0x6] AAC-LC sad1_channels 2 sad1_sampling_rates [0x18a0] 44100 88200 384000 sad1_max_bitrate 1024000 sad2_coding_type [0xc] MLP (Dolby TrueHD) sad2_channels 8 sad2_sampling_rates [0x1ec0] 48000 88200 176400 192000 384000 sad3_coding_type [0x8] ATRAC sad3_channels 3 sad3_sampling_rates [0x1220] 44100 176400 sad3_max_bitrate 944000 sad4_coding_type [0xd] DST sad4_channels 3 sad4_sampling_rates [0xac0] 48000 88200 176400 384000 sad5_coding_type [0x2] AC-3 sad5_channels 3 sad5_sampling_rates [0x1a60] 44100 48000 176400 384000 sad5_max_bitrate 592000 sad6_coding_type [0xc] MLP (Dolby TrueHD) sad6_channels 3 sad6_sampling_rates [0xc40] 48000 192000 384000 sad7_coding_type [0x9] DSD (1-bit audio) sad7_channels 6 sad7_sampling_rates [0xc20] 44100 192000 384000 sad8_coding_type [0x5] MPEG2 sad8_channels 1 sad8_sampling_rates [0x1ac0] 48000 88200 176400 384000 sad8_max_bitrate 544000 sad9_coding_type [0xc] MLP (Dolby TrueHD) sad9_channels 7 sad9_sampling_rates [0x460] 44100 48000 192000 sad10_coding_type [0x1] LPCM sad10_channels 4 sad10_sampling_rates [0x1260] 44100 48000 176400 sad10_sample_bits 0x40000 sad11_coding_type [0x6] AAC-LC sad11_channels 7 sad11_sampling_rates [0x1240] 48000 176400 sad11_max_bitrate 408000 sad12_coding_type [0x0] undefined sad12_channels 0 sad12_sampling_rates [0x0] sad13_coding_type [0x0] undefined sad13_channels 0 sad13_sampling_rates [0x0] Thank you, Fengguang --- stress testing patch for ELD routines sound/pci/hda/hda_eld.c | 30 ++++++++++++++++++++---------- sound/pci/hda/patch_intelhdmi.c | 2 ++ 2 files changed, 22 insertions(+), 10 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_eld.c +++ sound-2.6/sound/pci/hda/hda_eld.c @@ -27,6 +27,15 @@ #include "hda_codec.h" #include "hda_local.h" +static unsigned long rand; + +static inline unsigned char random(void) +{ + /* See "Numerical Recipes in C", second edition, p. 284 */ + rand = rand * 1664525L + 1013904223L; + return (unsigned char) (rand >> 24); +} + enum eld_versions { ELD_VER_CEA_861D = 2, ELD_VER_PARTIAL = 31, @@ -167,6 +176,8 @@ static unsigned char hdmi_get_eld_byte(s val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_ELDD, byte_index); + val = AC_ELDD_ELD_VALID | random(); + #ifdef BE_PARANOID printk(KERN_INFO "ELD data byte %d: 0x%x\n", byte_index, val); #endif @@ -277,7 +288,6 @@ static int hdmi_update_sink_eld(struct s if (e->eld_ver != ELD_VER_CEA_861D && e->eld_ver != ELD_VER_PARTIAL) { snd_printd(KERN_INFO "Unknown ELD version %d\n", e->eld_ver); - goto out_fail; } e->eld_size = size; @@ -359,9 +369,6 @@ int snd_hdmi_get_eld(struct sink_eld *el int size; unsigned char *buf; - if (!hdmi_eld_valid(codec, nid)) - return -ENOENT; - size = snd_hdmi_get_eld_size(codec, nid); if (size == 0) { /* wfg: workaround for ASUS P5E-VM HDMI board */ @@ -516,6 +523,9 @@ int snd_hda_eld_proc_new(struct hda_code struct snd_info_entry *entry; int err; + /* Initialize random number generator */ + rand = jiffies; + snprintf(name, sizeof(name), "eld#%d", codec->addr); err = snd_card_proc_new(codec->bus->card, name, &entry); if (err < 0) --- sound-2.6.orig/sound/pci/hda/patch_intelhdmi.c +++ sound-2.6/sound/pci/hda/patch_intelhdmi.c @@ -635,6 +635,8 @@ static int patch_intel_hdmi(struct hda_c init_channel_allocations(); + hdmi_parse_eld(codec); + return 0; }

2 4

[alsa-devel] [PATCH] hda: remove redundant get_amp_nid()
by Wu Fengguang 18 Nov '08

18 Nov '08

Remove get_amp_nid(): it duplicates the one defined in hda_local.h Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/hda/patch_via.c | 2 -- 1 file changed, 2 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_via.c +++ sound-2.6/sound/pci/hda/patch_via.c @@ -142,8 +142,6 @@ enum { AUTO_SEQ_SIDE }; -#define get_amp_nid(kc) ((kc)->private_value & 0xffff) - /* Some VT1708S based boards gets the micboost setting wrong, so we have * to apply some brute-force and re-write the TLV's by software. */ static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag,

1 0

[alsa-devel] (no subject)
by Fernando Moro 18 Nov '08

18 Nov '08

Dear member of the alsa time, I am not sure if this is the email addres in which it is possible to talk about a problem with the Alsa plugins, but it is the unique address that I have found. I am writing to tell you about the compatibility of the alsa plugins. There are lots of users (like me) who have the Lexicon lambda soundcard and the lexicon alpha soundcard, but alsa plugins only support the Lexicon Omega!, and it is impossible to connect our soundcards to the computer. I think that it is a problem for the Ubuntu Studio, Musix or Studio64 users, who use this devices to record and play midi songs, and I think that linux can be better than windows making audio. I would be grateful if alsa team could help us with this problem somehow. I look forward to your response. Thank you for your attention. Your faithfully, Fernando. Sorry for my bad English, I am Spanish. _________________________________________________________________ Los más de lo más, Especial Rankings http://events.es.msn.com/dinero/listas/default.aspx

1 0

[alsa-devel] [PATCH 002/002] snd-powermac: enable mic on iMac G4 (older kernels)
by Risto Suominen 17 Nov '08

17 Nov '08

1 0

[alsa-devel] snd-usb-audio: minimum period time as module option?
by Klaus Schulz 17 Nov '08

17 Nov '08

Hi folks (probably Clemens). Would it be possible to to introduce a module-option variable to snd-usb-audio, so that it is possible to switch minimum period from outside usbaudio.c /* set the period time minimum 1ms */ /* FIXME: high-speed mode allows 125us minimum period, but many parts * in the current code assume the 1ms period. */ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 1000 * MIN_PACKS_URB, /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX); With 125us my EMU USB 0404 and also my PCM2707 devices do sound better. THX \Klaus

1 1

Re: [alsa-devel] [PATCH] ALSA: hda - Add digital BEEP generator support for Realtek codecs.
by Takashi Iwai 17 Nov '08

17 Nov '08

At Sat, 15 Nov 2008 15:20:14 +0900, Kusanagi Kouichi wrote: > > Add digital BEEP generator support for Realtek codecs. > > Signed-off-by: Kusanagi Kouichi <slash(a)ma.neweb.ne.jp> Thanks for the patch. Do these beep widgets have the mute amp bits? If not, we'd likely need to add mute switches as well (otherwise it gets too annoying) controlling on software. Check patch_sigmatel.c in the latest master branch or topic/fix/hda branch of sound git tree: git://git.kernel.org/pub/scm/linux/kernel/tiwai/sound-2.6.git Another concern is that I don't want to add the digital beep control unconditionally. On many devices, the beep is implemented in the old good analog way. We'd need to add the check of analog/digital beep in a new spec field and in the preset table not to duplicate beep controls on these. thanks, Takashi > --- > sound/pci/hda/hda_beep.h | 2 +- > sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++++++ > 2 files changed, 63 insertions(+), 1 deletions(-) > > diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h > index de4036e..b483779 100644 > --- a/sound/pci/hda/hda_beep.h > +++ b/sound/pci/hda/hda_beep.h > @@ -38,7 +38,7 @@ struct hda_beep { > int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); > void snd_hda_detach_beep_device(struct hda_codec *codec); > #else > -#define snd_hda_attach_beep_device(...) > +#define snd_hda_attach_beep_device(...) 0 > #define snd_hda_detach_beep_device(...) > #endif > #endif > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index a378c01..d586600 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -31,6 +31,7 @@ > #include "hda_codec.h" > #include "hda_local.h" > #include "hda_patch.h" > +#include "hda_beep.h" > > #define ALC880_FRONT_EVENT 0x01 > #define ALC880_DCVOL_EVENT 0x02 > @@ -2788,6 +2789,7 @@ static void alc_free(struct hda_codec *codec) > kfree(spec->kctl_alloc); > } > kfree(spec); > + snd_hda_detach_beep_device(codec); > codec->spec = NULL; /* to be sure */ > } > > @@ -3960,6 +3962,12 @@ static int patch_alc880(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC880_AUTO) > setup_preset(spec, &alc880_presets[board_config]); > > @@ -5558,6 +5566,12 @@ static int patch_alc260(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC260_AUTO) > setup_preset(spec, &alc260_presets[board_config]); > > @@ -6811,6 +6825,12 @@ static int patch_alc882(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC882_AUTO) > setup_preset(spec, &alc882_presets[board_config]); > > @@ -8916,6 +8936,12 @@ static int patch_alc883(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC883_AUTO) > setup_preset(spec, &alc883_presets[board_config]); > > @@ -10739,6 +10765,12 @@ static int patch_alc262(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC262_AUTO) > setup_preset(spec, &alc262_presets[board_config]); > > @@ -11752,6 +11784,12 @@ static int patch_alc268(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC268_AUTO) > setup_preset(spec, &alc268_presets[board_config]); > > @@ -12458,6 +12496,12 @@ static int patch_alc269(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC269_AUTO) > setup_preset(spec, &alc269_presets[board_config]); > > @@ -13624,6 +13668,12 @@ static int patch_alc861(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x23); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC861_AUTO) > setup_preset(spec, &alc861_presets[board_config]); > > @@ -14578,6 +14628,12 @@ static int patch_alc861vd(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x23); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC861VD_AUTO) > setup_preset(spec, &alc861vd_presets[board_config]); > > @@ -16412,6 +16468,12 @@ static int patch_alc662(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC662_AUTO) > setup_preset(spec, &alc662_presets[board_config]); > > -- > 1.5.6.5 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

1 0

[alsa-devel] [PATCH] hda: remove redundant get_amp_nid()
by Wu Fengguang 17 Nov '08

17 Nov '08

Remove get_amp_nid(): it duplicates the one defined in hda_local.h Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/hda/patch_via.c | 2 -- 1 file changed, 2 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_via.c +++ sound-2.6/sound/pci/hda/patch_via.c @@ -142,8 +142,6 @@ enum { AUTO_SEQ_SIDE }; -#define get_amp_nid(kc) ((kc)->private_value & 0xffff) - /* Some VT1708S based boards gets the micboost setting wrong, so we have * to apply some brute-force and re-write the TLV's by software. */ static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag,

1 0

[alsa-devel] [PATCH RESEND 3/3] ALSA SOC driver for s3c24xx with uda134x
by Christian Pellegrin 17 Nov '08

17 Nov '08

Adds audio support for SMDK2410 Signed-off-by: Christian Pellegrin <chripell(a)fsfe.org> --- arch/arm/mach-s3c2410/mach-smdk2410.c | 18 ++++++++++++++++++ 1 files changed, 18 insertions(+), 0 deletions(-) -- Generated on 20081115 against f7e1798a78a30bae1cf3ebc3e1b6f7c1bac57756..5b7f05e9b57e7e442b1054cbb4423cf9909fb204 -- diff --git a/arch/arm/mach-s3c2410/mach-smdk2410.c b/arch/arm/mach-s3c2410/mach-smdk2410.c index 152527b..0001d52 100644 --- a/arch/arm/mach-s3c2410/mach-smdk2410.c +++ b/arch/arm/mach-s3c2410/mach-smdk2410.c @@ -43,6 +43,7 @@ #include <asm/mach/irq.h> #include <mach/hardware.h> +#include <mach/regs-gpio.h> #include <asm/irq.h> #include <asm/mach-types.h> @@ -53,6 +54,8 @@ #include <plat/common-smdk.h> +#include <sound/s3c24xx_uda134x.h> + static struct map_desc smdk2410_iodesc[] __initdata = { /* nothing here yet */ }; @@ -85,12 +88,27 @@ static struct s3c2410_uartcfg smdk2410_uartcfgs[] __initdata = { } }; +static struct s3c24xx_uda134x_platform_data s3c24xx_uda134x_data = { + .l3_clk = S3C2410_GPB4, + .l3_data = S3C2410_GPB3, + .l3_mode = S3C2410_GPB2, + .model = UDA134X_UDA1341, +}; + +static struct platform_device s3c24xx_uda134x = { + .name = "s3c24xx_uda134x", + .dev = { + .platform_data = &s3c24xx_uda134x_data, + } +}; + static struct platform_device *smdk2410_devices[] __initdata = { &s3c_device_usb, &s3c_device_lcd, &s3c_device_wdt, &s3c_device_i2c, &s3c_device_iis, + &s3c24xx_uda134x, }; static void __init smdk2410_map_io(void) -- 1.4.4.4

2 1

[alsa-devel] [PATCH RESEND 2/3] ALSA SOC driver for s3c24xx with uda134x
by Christian Pellegrin 17 Nov '08

17 Nov '08

Sound card definition for for s3c24xx with uda134x Signed-off-by: Christian Pellegrin <chripell(a)fsfe.org> --- include/sound/s3c24xx_uda134x.h | 14 ++ sound/soc/s3c24xx/Kconfig | 5 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c24xx_uda134x.c | 374 +++++++++++++++++++++++++++++++++++ 4 files changed, 395 insertions(+), 0 deletions(-) -- Generated on 20081115 against f7e1798a78a30bae1cf3ebc3e1b6f7c1bac57756 -- diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h new file mode 100644 index 0000000..33df4cb --- /dev/null +++ b/include/sound/s3c24xx_uda134x.h @@ -0,0 +1,14 @@ +#ifndef _S3C24XX_UDA134X_H_ +#define _S3C24XX_UDA134X_H_ 1 + +#include <sound/uda134x.h> + +struct s3c24xx_uda134x_platform_data { + int l3_clk; + int l3_mode; + int l3_data; + void (*power) (int); + int model; +}; + +#endif diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b9f2353..fcd03ac 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -44,3 +44,8 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650 Say Y if you want to add support for SoC audio on ln2440sbc with the ALC650. +config SND_S3C24XX_SOC_S3C24XX_UDA134X + tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_UDA134X diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 0aa5fb0..96b3f3f 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o +snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o +obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c new file mode 100644 index 0000000..29a6813 --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -0,0 +1,374 @@ +/* + * Modifications by Christian Pellegrin <chripell(a)evolware.org> + * + * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/mutex.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/s3c24xx_uda134x.h> +#include <sound/uda134x.h> + +#include <asm/plat-s3c24xx/regs-iis.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda134x_codec.h" + + +/* #define ENFORCE_RATES 1 */ +/* + Unfortunately the S3C24XX in master mode has a limited capacity of + generating the clock for the codec. If you define this only rates + that are really available will be enforced. But be careful, most + user level application just want the usual sampling frequencies (8, + 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly + operation for embedded systems. So if you aren't very lucky or your + hardware engineer wasn't very forward-looking it's better to leave + this undefined. If you do so an approximate value for the requested + sampling rate in the range -/+ 5% will be chosen. If this in not + possible an error will be returned. +*/ + +static struct clk *xtal; +static struct clk *pclk; +/* this is need because we don't have a place where to keep the + * pointers to the clocks in each substream. We get the clocks only + * when we are actually using them so we don't block stuff like + * frequency change or oscillator power-off */ +static int clk_users; +static DEFINE_MUTEX(clk_lock); + +static unsigned int rates[33 * 2]; +#ifdef ENFORCE_RATES +static struct snd_pcm_hw_constraint_list hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; +#endif + +static struct platform_device *s3c24xx_uda134x_snd_device; + +int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) +{ + int ret = 0; +#ifdef ENFORCE_RATES + struct snd_pcm_runtime *runtime = substream->runtime;; +#endif + + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + if (clk_users == 0) { + xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); + if (!xtal) { + printk(KERN_ERR "%s cannot get xtal\n", __func__); + ret = -EBUSY; + } else { + pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, + "pclk"); + if (!pclk) { + printk(KERN_ERR "%s cannot get pclk\n", + __func__); + clk_put(xtal); + ret = -EBUSY; + } + } + if (!ret) { + int i, j; + + for (i = 0; i < 2; i++) { + int fs = i ? 256 : 384; + + rates[i*33] = clk_get_rate(xtal) / fs; + for (j = 1; j < 33; j++) + rates[i*33 + j] = clk_get_rate(pclk) / + (j * fs); + } + } + } + clk_users += 1; + mutex_unlock(&clk_lock); + if (!ret) { +#ifdef ENFORCE_RATES + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_constraints_rates); + if (ret < 0) + printk(KERN_ERR "%s cannot set constraints\n", + __func__); +#endif + } + return ret; +} + +void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) +{ + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + clk_users -= 1; + if (clk_users == 0) { + clk_put(xtal); + xtal = NULL; + clk_put(pclk); + pclk = NULL; + } + mutex_unlock(&clk_lock); +} + +static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + int ret = 0; + int clk_source, fs_mode; + unsigned long rate = params_rate(params); + long err, cerr; + unsigned int div; + int i, bi; + + err = 999999; + bi = 0; + for (i = 0; i < 2*33; i++) { + cerr = rates[i] - rate; + if (cerr < 0) + cerr = -cerr; + if (cerr < err) { + err = cerr; + bi = i; + } + } + if (bi / 33 == 1) + fs_mode = S3C2410_IISMOD_256FS; + else + fs_mode = S3C2410_IISMOD_384FS; + if (bi % 33 == 0) { + clk_source = S3C24XX_CLKSRC_MPLL; + div = 1; + } else { + clk_source = S3C24XX_CLKSRC_PCLK; + div = bi % 33; + } + pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); + + clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; + pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, + fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", + clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", + div, clk, err); + + if ((err * 100 / rate) > 5) { + printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " + "too different from desired (%ld%%)\n", + err * 100 / rate); + return -EINVAL; + } + + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + fs_mode); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops s3c24xx_uda134x_ops = { + .startup = s3c24xx_uda134x_startup, + .shutdown = s3c24xx_uda134x_shutdown, + .hw_params = s3c24xx_uda134x_hw_params, +}; + +static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { + .name = "UDA134X", + .stream_name = "UDA134X", + .codec_dai = &uda134x_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &s3c24xx_uda134x_ops, +}; + +static struct snd_soc_machine snd_soc_machine_s3c24xx_uda134x = { + .name = "S3C24XX_UDA134X", + .dai_link = &s3c24xx_uda134x_dai_link, + .num_links = 1, +}; + +static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins; + +static void setdat(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0); +} + +static void setclk(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0); +} + +static void setmode(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0); +} + +static struct uda134x_platform_data s3c24xx_uda134x = { + .l3 = { + .setdat = setdat, + .setclk = setclk, + .setmode = setmode, + .data_hold = 1, + .data_setup = 1, + .clock_high = 1, + .mode_hold = 1, + .mode = 1, + .mode_setup = 1, + }, +}; + +static struct snd_soc_device s3c24xx_uda134x_snd_devdata = { + .machine = &snd_soc_machine_s3c24xx_uda134x, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_uda134x, + .codec_data = &s3c24xx_uda134x, +}; + +static int s3c24xx_uda134x_setup_pin(int pin, char *fun) +{ + if (gpio_request(pin, "s3c24xx_uda134x") < 0) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "l3 %s pin already in use", fun); + return -EBUSY; + } + gpio_direction_output(pin, 0); + return 0; +} + +static int s3c24xx_uda134x_probe(struct platform_device *pdev) +{ + int ret; + + printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n"); + + s3c24xx_uda134x_l3_pins = pdev->dev.platform_data; + if (s3c24xx_uda134x_l3_pins == NULL) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "unable to find platform data\n"); + return -ENODEV; + } + s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power; + s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model; + + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data, + "data") < 0) + return -EBUSY; + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk, + "clk") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + return -EBUSY; + } + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode, + "mode") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + return -EBUSY; + } + + s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_uda134x_snd_device) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(s3c24xx_uda134x_snd_device, + &s3c24xx_uda134x_snd_devdata); + s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev; + ret = platform_device_add(s3c24xx_uda134x_snd_device); + if (ret) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); + platform_device_put(s3c24xx_uda134x_snd_device); + } + + return ret; +} + +static int s3c24xx_uda134x_remove(struct platform_device *pdev) +{ + platform_device_unregister(s3c24xx_uda134x_snd_device); + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + gpio_free(s3c24xx_uda134x_l3_pins->l3_mode); + return 0; +} + +static struct platform_driver s3c24xx_uda134x_driver = { + .probe = s3c24xx_uda134x_probe, + .remove = s3c24xx_uda134x_remove, + .driver = { + .name = "s3c24xx_uda134x", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c24xx_uda134x_init(void) +{ + return platform_driver_register(&s3c24xx_uda134x_driver); +} + +static void __exit s3c24xx_uda134x_exit(void) +{ + platform_driver_unregister(&s3c24xx_uda134x_driver); +} + + +module_init(s3c24xx_uda134x_init); +module_exit(s3c24xx_uda134x_exit); + +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); +MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); -- 1.4.4.4

2 1

[alsa-devel] [PATCH RESEND 1/3] ALSA SOC driver for s3c24xx with uda1341
by Christian Pellegrin 17 Nov '08

17 Nov '08

UDA134x codec driver Signed-off-by: Christian Pellegrin <chripell(a)fsfe.org> --- include/sound/l3.h | 18 + include/sound/uda134x.h | 26 ++ sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/l3.c | 91 ++++++ sound/soc/codecs/uda134x.c | 656 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/uda134x_codec.h | 36 ++ 7 files changed, 839 insertions(+), 0 deletions(-) -- Generated on 20081115 against f7e1798a78a30bae1cf3ebc3e1b6f7c1bac57756 -- diff --git a/include/sound/l3.h b/include/sound/l3.h new file mode 100644 index 0000000..423a08f --- /dev/null +++ b/include/sound/l3.h @@ -0,0 +1,18 @@ +#ifndef _L3_H_ +#define _L3_H_ 1 + +struct l3_pins { + void (*setdat)(int); + void (*setclk)(int); + void (*setmode)(int); + int data_hold; + int data_setup; + int clock_high; + int mode_hold; + int mode; + int mode_setup; +}; + +int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len); + +#endif diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h new file mode 100644 index 0000000..475ef8b --- /dev/null +++ b/include/sound/uda134x.h @@ -0,0 +1,26 @@ +/* + * uda134x.h -- UDA134x ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * 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. + */ + +#ifndef _UDA134X_H +#define _UDA134X_H + +#include <sound/l3.h> + +struct uda134x_platform_data { + struct l3_pins l3; + void (*power) (int); + int model; +#define UDA134X_UDA1340 1 +#define UDA134X_UDA1341 2 +#define UDA134X_UDA1344 3 +}; + +#endif /* _UDA134X_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b73c36a..27badab 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8510 if (I2C || SPI_MASTER) select SND_SOC_WM8580 if I2C @@ -65,6 +66,9 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_L3 + tristate + config SND_SOC_SSM2602 tristate @@ -84,6 +88,10 @@ config SND_SOC_TWL4030 tristate depends on TWL4030_CORE +config SND_SOC_UDA134X + tristate + select SND_SOC_L3 + config SND_SOC_UDA1380 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3b9b58a..3246b51 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,11 +3,13 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o +snd-soc-l3-objs := l3.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8580-objs := wm8580.o @@ -26,11 +28,13 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c new file mode 100644 index 0000000..5353af5 --- /dev/null +++ b/sound/soc/codecs/l3.c @@ -0,0 +1,91 @@ +/* + * L3 code + * + * Copyright (C) 2008, Christian Pellegrin <chripell(a)evolware.org> + * + * 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. + * + * + * based on: + * + * L3 bus algorithm module. + * + * Copyright (C) 2001 Russell King, All Rights Reserved. + * + * + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/delay.h> + +#include <sound/l3.h> + +/* + * Send one byte of data to the chip. Data is latched into the chip on + * the rising edge of the clock. + */ +static void sendbyte(struct l3_pins *adap, unsigned int byte) +{ + int i; + + for (i = 0; i < 8; i++) { + adap->setclk(0); + udelay(adap->data_hold); + adap->setdat(byte & 1); + udelay(adap->data_setup); + adap->setclk(1); + udelay(adap->clock_high); + byte >>= 1; + } +} + +/* + * Send a set of bytes to the chip. We need to pulse the MODE line + * between each byte, but never at the start nor at the end of the + * transfer. + */ +static void sendbytes(struct l3_pins *adap, const u8 *buf, + int len) +{ + int i; + + for (i = 0; i < len; i++) { + if (i) { + udelay(adap->mode_hold); + adap->setmode(0); + udelay(adap->mode); + } + adap->setmode(1); + udelay(adap->mode_setup); + sendbyte(adap, buf[i]); + } +} + +int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len) +{ + adap->setclk(1); + adap->setdat(1); + adap->setmode(1); + udelay(adap->mode); + + adap->setmode(0); + udelay(adap->mode_setup); + sendbyte(adap, addr); + udelay(adap->mode_hold); + + sendbytes(adap, data, len); + + adap->setclk(1); + adap->setdat(1); + adap->setmode(0); + + return len; +} +EXPORT_SYMBOL_GPL(l3_write); + +MODULE_DESCRIPTION("L3 bit-banging driver"); +MODULE_AUTHOR("Christian Pellegrin <chripell(a)evolware.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c new file mode 100644 index 0000000..04b30da --- /dev/null +++ b/sound/soc/codecs/uda134x.c @@ -0,0 +1,656 @@ +/* + * uda134x.c -- UDA134X ALSA SoC Codec driver + * + * Modifications by Christian Pellegrin <chripell(a)evolware.org> + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <sound/uda134x.h> +#include <sound/l3.h> + +#include "uda134x_codec.h" + + +#define POWER_OFF_ON_STANDBY 1 +/* + ALSA SOC usually puts the device in standby mode when it's not used + for sometime. If you define POWER_OFF_ON_STANDBY the driver will + turn off the ADC/DAC when this callback is invoked and turn it back + on when needed. Unfortunately this will result in a very light bump + (it can be audible only with good earphones). If this bothers you + just comment this line, you will have slightly higher power + consumption . Please note that sending the L3 command for ADC is + enough to make the bump, so it doesn't make difference if you + completely take off power from the codec. + */ + +#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 +#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) + +struct uda134x_priv { + int sysclk; + int dai_fmt; + + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + +/* In-data addresses are hard-coded into the reg-cache values */ +static const char uda134x_reg[UDA134X_REGS_NUM] = { + /* Extended address registers */ + 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, + /* Status, data regs */ + 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, +}; + +/* + * The codec has no support for reading its registers except for peak level... + */ +static inline unsigned int uda134x_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA134X_REGS_NUM) + return -1; + return cache[reg]; +} + +/* + * Write the register cache + */ +static inline void uda134x_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA134X_REGS_NUM) + return; + cache[reg] = value; +} + +/* + * Write to the uda134x registers + * + */ +static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + u8 addr; + u8 data = value; + struct uda134x_platform_data *pd = codec->control_data; + + pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); + + if (reg >= UDA134X_REGS_NUM) { + printk(KERN_ERR "%s unkown register: reg: %d", + __func__, reg); + return -EINVAL; + } + + uda134x_write_reg_cache(codec, reg, value); + + switch (reg) { + case UDA134X_STATUS0: + case UDA134X_STATUS1: + addr = UDA134X_STATUS_ADDR; + break; + case UDA134X_DATA000: + case UDA134X_DATA001: + case UDA134X_DATA010: + addr = UDA134X_DATA0_ADDR; + break; + case UDA134X_DATA1: + addr = UDA134X_DATA1_ADDR; + break; + default: + /* It's an extended address register */ + addr = (reg | UDA134X_EXTADDR_PREFIX); + + ret = l3_write(&pd->l3, + UDA134X_DATA0_ADDR, &addr, 1); + if (ret != 1) + return -EIO; + + addr = UDA134X_DATA0_ADDR; + data = (value | UDA134X_EXTDATA_PREFIX); + break; + } + + ret = l3_write(&pd->l3, + addr, &data, 1); + if (ret != 1) + return -EIO; + + return 0; +} + +static inline void uda134x_reset(struct snd_soc_codec *codec) +{ + u8 reset_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS0); + uda134x_write(codec, UDA134X_STATUS0, reset_reg | (1<<6)); + msleep(1); + uda134x_write(codec, UDA134X_STATUS0, reset_reg & ~(1<<6)); +} + +static int uda134x_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = uda134x_read_reg_cache(codec, UDA134X_DATA010); + + pr_debug("%s mute: %d\n", __func__, mute); + + if (mute) + mute_reg |= (1<<2); + else + mute_reg &= ~(1<<2); + + uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2)); + + return 0; +} + +static int uda134x_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + struct snd_pcm_runtime *master_runtime; + + if (uda134x->master_substream) { + master_runtime = uda134x->master_substream->runtime; + + pr_debug("%s constraining to %d bits at %d\n", __func__, + master_runtime->sample_bits, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + uda134x->slave_substream = substream; + } else + uda134x->master_substream = substream; + + return 0; +} + +static void uda134x_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + + if (uda134x->master_substream == substream) + uda134x->master_substream = uda134x->slave_substream; + + uda134x->slave_substream = NULL; +} + +static int uda134x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + u8 hw_params; + + if (substream == uda134x->slave_substream) { + pr_debug("%s ignoring hw_params for slave substream\n", + __func__); + return 0; + } + + hw_params = uda134x_read_reg_cache(codec, UDA134X_STATUS0); + hw_params &= STATUS0_SYSCLK_MASK; + hw_params &= STATUS0_DAIFMT_MASK; + + pr_debug("%s sysclk: %d, rate:%d\n", __func__, + uda134x->sysclk, params_rate(params)); + + /* set SYSCLK / fs ratio */ + switch (uda134x->sysclk / params_rate(params)) { + case 512: + break; + case 384: + hw_params |= (1<<4); + break; + case 256: + hw_params |= (1<<5); + break; + default: + printk(KERN_ERR "%s unsupported fs\n", __func__); + return -EINVAL; + } + + pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__, + uda134x->dai_fmt, params_format(params)); + + /* set DAI format and word length */ + switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + hw_params |= (1<<1); + break; + case SNDRV_PCM_FORMAT_S18_3LE: + hw_params |= (1<<2); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + hw_params |= ((1<<2) | (1<<1)); + break; + default: + printk(KERN_ERR "%s unsupported format (right)\n", + __func__); + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + hw_params |= (1<<3); + break; + default: + printk(KERN_ERR "%s unsupported format\n", __func__); + return -EINVAL; + } + + uda134x_write(codec, UDA134X_STATUS0, hw_params); + + return 0; +} + +static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda134x_priv *uda134x = codec->private_data; + + pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__, + clk_id, freq, dir); + + /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable + because the codec is slave. Of course limitations of the clock + master (the IIS controller) apply. + We'll error out on set_hw_params if it's not OK */ + if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) { + uda134x->sysclk = freq; + return 0; + } + + printk(KERN_ERR "%s unsupported sysclk\n", __func__); + return -EINVAL; +} + +static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda134x_priv *uda134x = codec->private_data; + + pr_debug("%s fmt: %08X\n", __func__, fmt); + + /* codec supports only full slave mode */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + printk(KERN_ERR "%s unsupported slave mode\n", __func__); + return -EINVAL; + } + + /* no support for clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { + printk(KERN_ERR "%s unsupported clock inversion\n", __func__); + return -EINVAL; + } + + /* We can't setup DAI format here as it depends on the word bit num */ + /* so let's just store the value for later */ + uda134x->dai_fmt = fmt; + + return 0; +} + +static int uda134x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 reg; + struct uda134x_platform_data *pd = codec->control_data; + int i; + u8 *cache = codec->reg_cache; + + pr_debug("%s bias level %d\n", __func__, level); + + switch (level) { + case SND_SOC_BIAS_ON: + /* ADC, DAC on */ + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); + break; + case SND_SOC_BIAS_PREPARE: + /* power on */ + if (pd->power) { + pd->power(1); + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda134x_reg); i++) + codec->write(codec, i, *cache++); + } + break; + case SND_SOC_BIAS_STANDBY: + /* ADC, DAC power off */ + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); + break; + case SND_SOC_BIAS_OFF: + /* power off */ + if (pd->power) + pd->power(0); + break; + } + codec->bias_level = level; + return 0; +} + +static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1", + "Minimum2", "Maximum"}; +static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *uda134x_mixmode[] = {"Differential", "Analog1", + "Analog2", "Both"}; + +static const struct soc_enum uda134x_mixer_enum[] = { +SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting), +SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph), +SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode), +}; + +static const struct snd_kcontrol_new uda1341_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), +SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0), +SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1), +SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1), + +SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0), +SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0), + +SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), +SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), + +SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), +SOC_ENUM("Input Mux", uda134x_mixer_enum[2]), + +SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0), +SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1), +SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0), + +SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0), +SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0), +SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0), +SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0), +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + +static const struct snd_kcontrol_new uda1340_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), + +SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), +SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), + +SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), + +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + +static int uda134x_add_controls(struct snd_soc_codec *codec) +{ + int err, i, n; + const struct snd_kcontrol_new *ctrls; + struct uda134x_platform_data *pd = codec->control_data; + + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + n = ARRAY_SIZE(uda1340_snd_controls); + ctrls = uda1340_snd_controls; + break; + case UDA134X_UDA1341: + n = ARRAY_SIZE(uda1341_snd_controls); + ctrls = uda1341_snd_controls; + break; + default: + printk(KERN_ERR "%s unkown codec type: %d", + __func__, pd->model); + return -EINVAL; + } + + for (i = 0; i < n; i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ctrls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +struct snd_soc_dai uda134x_dai = { + .name = "UDA134X", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA134X_RATES, + .formats = UDA134X_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA134X_RATES, + .formats = UDA134X_FORMATS, + }, + /* pcm operations */ + .ops = { + .startup = uda134x_startup, + .shutdown = uda134x_shutdown, + .hw_params = uda134x_hw_params, + }, + /* DAI operations */ + .dai_ops = { + .digital_mute = uda134x_mute, + .set_sysclk = uda134x_set_dai_sysclk, + .set_fmt = uda134x_set_dai_fmt, + } +}; +EXPORT_SYMBOL(uda134x_dai); + + +static int uda134x_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda134x_priv *uda134x; + void *codec_setup_data = socdev->codec_data; + int ret = -ENOMEM; + struct uda134x_platform_data *pd; + + printk(KERN_INFO "UDA134X SoC Audio Codec\n"); + + if (!codec_setup_data) { + printk(KERN_ERR "UDA134X SoC codec: " + "missing L3 bitbang function\n"); + return -ENODEV; + } + + pd = codec_setup_data; + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1341: + case UDA134X_UDA1344: + break; + default: + printk(KERN_ERR "UDA134X SoC codec: " + "unsupported model %d\n", + pd->model); + return -EINVAL; + } + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return ret; + + codec = socdev->codec; + + uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL); + if (uda134x == NULL) + goto priv_err; + codec->private_data = uda134x; + + codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + goto reg_err; + + mutex_init(&codec->mutex); + + codec->reg_cache_size = sizeof(uda134x_reg); + codec->reg_cache_step = 1; + + codec->name = "UDA134X"; + codec->owner = THIS_MODULE; + codec->dai = &uda134x_dai; + codec->num_dai = 1; + codec->read = uda134x_read_reg_cache; + codec->write = uda134x_write; +#ifdef POWER_OFF_ON_STANDBY + codec->set_bias_level = uda134x_set_bias_level; +#endif + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->control_data = codec_setup_data; + + if (pd->power) + pd->power(1); + + uda134x_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register pcms\n"); + goto pcm_err; + } + + ret = uda134x_add_controls(codec); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register controls\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); +reg_err: + kfree(codec->private_data); +priv_err: + kfree(codec); + return ret; +} + +/* power down chip */ +static int uda134x_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} + +#if defined(CONFIG_PM) +static int uda134x_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int uda134x_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); + return 0; +} +#else +#define uda134x_soc_suspend NULL +#define uda134x_soc_resume NULL +#endif /* CONFIG_PM */ + +struct snd_soc_codec_device soc_codec_dev_uda134x = { + .probe = uda134x_soc_probe, + .remove = uda134x_soc_remove, + .suspend = uda134x_soc_suspend, + .resume = uda134x_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x); + +MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x_codec.h b/sound/soc/codecs/uda134x_codec.h new file mode 100644 index 0000000..94f4404 --- /dev/null +++ b/sound/soc/codecs/uda134x_codec.h @@ -0,0 +1,36 @@ +#ifndef _UDA134X_CODEC_H +#define _UDA134X_CODEC_H + +#define UDA134X_L3ADDR 5 +#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0) +#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1) +#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2) + +#define UDA134X_EXTADDR_PREFIX 0xC0 +#define UDA134X_EXTDATA_PREFIX 0xE0 + +/* UDA134X registers */ +#define UDA134X_EA000 0 +#define UDA134X_EA001 1 +#define UDA134X_EA010 2 +#define UDA134X_EA011 3 +#define UDA134X_EA100 4 +#define UDA134X_EA101 5 +#define UDA134X_EA110 6 +#define UDA134X_EA111 7 +#define UDA134X_STATUS0 8 +#define UDA134X_STATUS1 9 +#define UDA134X_DATA000 10 +#define UDA134X_DATA001 11 +#define UDA134X_DATA010 12 +#define UDA134X_DATA1 13 + +#define UDA134X_REGS_NUM 14 + +#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4)) + +extern struct snd_soc_dai uda134x_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda134x; + +#endif -- 1.4.4.4

2 1

[alsa-devel] possible problem in s3c24xx-pcm.c in git-devel branch
by christian pellegrin 17 Nov '08

17 Nov '08

Hi, during the porting the uda134x driver to the git devel branch of ALSA SOC I noticed a possible problem in s3c24xx-pcm.c. s3c24xx_pcm_enqueue gained a new parameter that specifies the maximum number of periods that can be enqueued. In s3c24xx_pcm_prepare, which is called on play/capture start, this is limited to 1. Unfortunately on the 2410 the result of this change is that the DMA is stopped because the autoload of the next DMA descriptor doesn't work. I copied a log at the end of this email. Look for DMT==00 after the irq signaling the end of the first period and the subsequent fail of DMA loading since the channel is stopped. Now I fixed with: --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -253,7 +253,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) prtd->dma_pos = prtd->dma_start; /* enqueue dma buffers */ - s3c24xx_pcm_enqueue(substream, 1); + s3c24xx_pcm_enqueue(substream, 2); return ret; } but this really could be "playing with DMA" since I'm not sure why there is a 1 (since min_period is 2 I would guess 2 is the right value). Anyway with this change play and capture work well. # /opt/arm/alsa/bin/aplay /mp3z/prova.wav s3c24xx-pcm: Entered s3c24xx_pcm_open s3c24xx_uda134x_startup 0 31250, 132031, 66015, 44010, 33007, 26406, 22005, 18861, 16503, 14670, 13203, 12002, 11002, 10156, 9430, 8802, 8251, 7766, 7335, 6949, 6601, 6287, 6001, 5740, 5501, 5281, 5078, 4890, 4715, 4552, 4401, 4259, 4125, 46875, 198046, 99023, 66015, 49511, 39609, 33007, 28292, 24755, 22005, 19804, 18004, 16503, 15234, 14146, 13203, 12377, 11649, 11002, 10423, 9902, 9430, 9002, 8610, 8251, 7921, 7617, 7335, 7073, 6829, 6601, 6388, 6188, Playing WAVE '/mp3z/prova.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono s3c24xx_uda134x_hw_params desired rate 8000, 58 s3c24xx_uda134x_hw_params will use: 256FS PCLK 24 sysclk 2048000 err 79 uda134x_set_dai_fmt fmt: 00003000 s3c24xx-i2s: Entered s3c24xx_i2s_set_fmt s3c24xx-i2s: hw_params r: IISMOD: 9 s3c24xx-i2s: hw_params w: IISMOD: 9 s3c24xx-i2s: Entered s3c24xx_i2s_set_sysclk s3c24xx-i2s: Entered s3c24xx_i2s_set_clkdiv s3c24xx-i2s: Entered s3c24xx_i2s_set_clkdiv s3c24xx-i2s: Entered s3c24xx_i2s_set_clkdiv uda134x_set_dai_sysclk clk_id: 0, freq: 2048000, dir: 1 uda134x_hw_params sysclk: 2048000, rate:8000 uda134x_hw_params dai_fmt: 12288, params_format:2 uda134x_write reg: 08, value:20 s3c24xx-i2s: Entered s3c24xx_i2s_hw_params s3c24xx-i2s: hw_params r: IISMOD: 9 s3c24xx-i2s: hw_params w: IISMOD: 9 s3c24xx-pcm: Entered s3c24xx_pcm_hw_params s3c24xx-pcm: params bf0673f4, client bf067404, channel 10 dma10: s3c2410_request_dma: client=I2S PCM Stereo out, dev=00000000 mapped channel 10 to 2 dma2: s3c2410_dma_request:755: ls=0, cur=c326b020, 00000000 c326b020 dma2: s3c2410_dma_request:755: DCSRC=332a05c4, DISRC=332a0000, DSTAT=00000000 DMT=04, DCON=a0d00000 dma10: s3c2410_dma_request : requesting irq 35 s3c2410_dma_request: channel initialised, c044fdd4 s3c2410_dma_set_buffdone_fn: chan=c044fdd4, callback rtn=bf05d108 s3c24xx-pcm: Entered s3c24xx_pcm_mmap s3c24xx-pcm: Entered s3c24xx_pcm_prepare s3c2410_dma_devconfig: source=1, hwcfg=00000003, devaddr=55000010 s3c2410_dma_devconfig: mem source, devaddr=55000010, hwcfg=3 s3c2410_dma_config: chan=10, xfer_unit=2, dcon=80000000 s3c2410_dma_config: Initial dcon is 80000000 s3c2410_dma_config: New dcon is 80000000 s3c2410_dma_config: dcon now a0900000 s3c2410_dma_flush: chan c044fdd4 (2) dma2: s3c2410_dma_flush:912: ls=0, cur=c326b020, 00000000 c326b020 dma2: s3c2410_dma_flush:912: DCSRC=332a05c4, DISRC=332a0000, DSTAT=00000000 DMT=04, DCON=a0d00000 s3c2410_dma_flush: free buffer c326b020, next 00000000 s3c24xx-pcm: Entered s3c24xx_audio_buffdone dma2: s3c2410_dma_flush:939: DCSRC=332a05c4, DISRC=332a0000, DSTAT=00000000 DMT=04, DCON=a0d00000 dma2: s3c2410_dma_flush:954: DCSRC=332a05c4, DISRC=332a0000, DSTAT=00000000 DMT=04, DCON=a0d00000 s3c24xx-pcm: Entered s3c24xx_pcm_enqueue dma_max 1 s3c24xx-pcm: dma_loaded: 0 pos: 332a0000 len: 4096 dma_max: 1 dma_end: 332a3e80 s3c2410_dma_enqueue: id=c31fa300, data=332a0000, size=4096 s3c2410_dma_enqueue: buffer c326b020 queued onto empty channel s3c24xx-pcm: ret 0 period 4096 uda134x_mute mute: 0 uda134x_write reg: 0C, value:80 s3c24xx-pcm: Entered s3c24xx_pcm_trigger s3c2410_start_dma: channel=2 s3c2410_chan_loadbuffer: loading buff c326b020 (0x332a0000,0x001000) load_state is none, checking for noreload (next=00000000) dma2: s3c2410_dma_start:366: ls=1, cur=c326b020, 00000000 c326b020 dma2: s3c2410_dma_start:366: DCSRC=332a05c4, DISRC=332a0000, DSTAT=00000000 DMT=04, DCON=a0d00800 dma2: 00000002 to DMASKTRIG dma2: s3c2410_dma_start:394: ls=1, cur=c326b020, 00000000 c326b020 dma2: s3c2410_dma_start:394: DCSRC=332a05c4, DISRC=332a0000, DSTAT=00000000 DMT=02, DCON=a0d00800 dma2: s3c2410_dma_started:967: ls=1, cur=c326b020, 00000000 c326b020 dma2: s3c2410_dma_started:967: DCSRC=332a05c4, DISRC=332a0000, DSTAT=00000000 DMT=02, DCON=a0d00800 s3c24xx-i2s: Entered s3c24xx_i2s_trigger s3c24xx-i2s: Entered s3c24xx_snd_is_clkmaster s3c24xx-i2s: Entered s3c24xx_snd_txctrl s3c24xx-i2s: r: IISCON: 10f IISMOD: 9 IISFCON: 0 s3c24xx-i2s: w: IISCON: 127 IISMOD: 89 IISFCON: a000 s3c24xx-pcm: Entered s3c24xx_pcm_pointer s3c24xx-pcm: Pointer 332a0158 55000010 dma2: s3c2410_dma_irq:608: ls=1, cur=c326b020, 00000000 c326b020 dma2: s3c2410_dma_irq:608: DCSRC=332a1000, DISRC=332a0000, DSTAT=00000000 DMT=00, DCON=a0d00800 s3c24xx-pcm: Entered s3c24xx_audio_buffdone s3c24xx-pcm: Entered s3c24xx_pcm_pointer s3c24xx-pcm: Pointer 332a1000 55000010 s3c24xx-pcm: Entered s3c24xx_pcm_enqueue dma_max 0 s3c24xx-pcm: dma_loaded: 0 pos: 332a1000 len: 4096 dma_max: 2 dma_end: 332a3e80 s3c2410_dma_enqueue: id=c31fa300, data=332a1000, size=4096 s3c2410_dma_enqueue: buffer c326b040 queued onto empty channel s3c2410_chan_loadbuffer: loading buff c326b040 (0x332a1000,0x001000) load_state is none, checking for noreload (next=00000000) s3c24xx-pcm: ret 0 period 4096 s3c24xx-pcm: dma_loaded: 1 pos: 332a2000 len: 4096 dma_max: 2 dma_end: 332a3e80 s3c2410_dma_enqueue: id=c31fa300, data=332a2000, size=4096 dma2: s3c2410_dma_enqueue: buffer c326b000 queued onto non-empty channel dma2: loadbuffer:timeout loading buffer dma2: s3c2410_dma_enqueue:517: ls=1, cur=c326b040, c326b000 c326b000 dma2: s3c2410_dma_enqueue:517: DCSRC=332a1000, DISRC=332a1000, DSTAT=00000000 DMT=00, DCON=a0d00800 s3c24xx-pcm: ret -22 period 4096 dma2: timeout waiting for load (s3c2410_dma_irq) s3c24xx-pcm: Entered s3c24xx_pcm_pointer s3c24xx-pcm: Pointer 332a1000 55000010 s3c24xx-pcm: Entered s3c24xx_pcm_pointer s3c24xx-pcm: Pointer 332a1000 55000010 ^CAborted by signals3c24xx-pcm: Entered s3c24xx_pcm_trigger s3c2410_dma_dostop: dma2: s3c2410_dma_dostop:857: ls=1, cur=c326b040, c326b000 c326b000 dma2: s3c2410_dma_dostop:857: DCSRC=332a1000, DISRC=332a1000, DSTAT=00000000 DMT=00, DCON=a0d00800 s3c24xx-i2s: Entered s3c24xx_i2s_trigger s3c24xx-i2s: Entered s3c24xx_snd_txctrl s3c24xx-i2s: r: IISCON: 127 IISMOD: 89 IISFCON: a000 -- Christian Pellegrin, see http://www.evolware.org/chri/ "Real Programmers don't play tennis, or any other sport which requires you to change clothes. Mountain climbing is OK, and Real Programmers wear their climbing boots to work in case a mountain should suddenly spring up in the middle of the computer room."

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[alsa-devel] [PATCH] hda-intel: reorder HDMI audio enabling sequence
by Wu Fengguang 17 Nov '08

17 Nov '08

Reorder HDMI audio enabling sequence so that 1) the sink knows about the coming audio stream 2) unmute 3) start transferring audio samples The theory is that in the path A=>B=>C, we first make C ready, and then enable B, and lastly allow A to send audio samples. Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/hda/patch_intelhdmi.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_intelhdmi.c +++ sound-2.6/sound/pci/hda/patch_intelhdmi.c @@ -312,16 +312,16 @@ static void hdmi_write_dip_byte(struct h static void hdmi_enable_output(struct hda_codec *codec) { - /* Enable pin out and unmute */ - snd_hda_sequence_write(codec, pinout_enable_verb); - if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, PIN_NID, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); - /* Enable Audio InfoFrame Transmission */ hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_BEST); + /* Unmute */ + if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, PIN_NID, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + /* Enable pin out */ + snd_hda_sequence_write(codec, pinout_enable_verb); } static void hdmi_disable_output(struct hda_codec *codec)

2 2

[alsa-devel] [PATCH] hda: STAC_DELL_M6 EAPD
by Matthew Ranostay 17 Nov '08

17 Nov '08

Add support for EAPD on system suspend and disabling EAPD on headphone jack detection for STAC_DELL_M6 laptops. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index ec8f5ab..e0298df 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -147,6 +147,7 @@ struct sigmatel_spec { unsigned int num_mixers; int board_config; + unsigned int eapd_switch: 1; unsigned int surr_switch: 1; unsigned int line_switch: 1; unsigned int mic_switch: 1; @@ -3944,7 +3945,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) for (i = 0; i < cfg->speaker_outs; i++) stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); - if (spec->eapd_mask) + if (spec->eapd_mask && spec->eapd_switch) stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data & ~spec->eapd_mask); @@ -3959,7 +3960,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) for (i = 0; i < cfg->speaker_outs; i++) stac92xx_set_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); - if (spec->eapd_mask) + if (spec->eapd_mask && spec->eapd_switch) stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data | spec->eapd_mask); @@ -4319,6 +4320,7 @@ again: spec->num_smuxes = 0; spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER]; spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP]; + spec->eapd_switch = 0; spec->num_amps = 1; if (!spec->init) @@ -4350,6 +4352,7 @@ again: default: spec->num_dmics = STAC92HD73XX_NUM_DMICS; spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); + spec->eapd_switch = 1; } if (spec->board_config > STAC_92HD73XX_REF) { /* GPIO0 High = Enable EAPD */ @@ -4495,7 +4498,13 @@ static int stac92hd71xx_resume(struct hda_codec *codec) static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state) { + struct sigmatel_spec *spec = codec->spec; + stac92hd71xx_set_power_state(codec, AC_PWRST_D3); + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); return 0; }; @@ -4885,6 +4894,7 @@ static int patch_stac927x(struct hda_codec *codec) spec->num_pwrs = 0; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; + spec->eapd_switch = 1; err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { @@ -4965,6 +4975,7 @@ static int patch_stac9205(struct hda_codec *codec) spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; + spec->eapd_switch = 1; spec->multiout.dac_nids = spec->dac_nids; switch (spec->board_config){

2 1

Re: [alsa-devel] digital audio on ASUS P5QL-EM motherboard
by Andrei Tanas 17 Nov '08

17 Nov '08

> It seems that there are two problems: > - the SPDIF output PID for you board is 0x10 instead of 0x06 You are right about this, it works with the patch you provided, with one caveat: ALC1200 uses ALC883 patch, so instead of changing ALC882_DIGOUT_NID, I changed ALC883_DIGOUT_NID. How did you find out that it's 0x10 instead of 0x06? Has documentation on ALC1200 become available? > - codec#1 is not probed properly I was responsible for the confusion with the missing codec. I set probe_mask=1 parameter (as suggested in some forum as a troubleshooting option for ALC1200) without trying to understand what it does. The probing actually seems to work correctly. Model autodetection: lspci -nn -vvv shows: 00:1b.0 Audio device [0403]: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller [8086:3a3e] Subsystem: ASUSTeK Computer Inc. Device [1043:82fe] So this SND_PCI_QUIRK fixes it (except that it's currently relying on the temporary hack) --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8206,6 +8206,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), Thank you. ---------------------- aplay -l ---------------------- **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: INTEL HDMI [INTEL HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 ---------------------- aplay -L (with model=6stack-dig) ---------------------- front:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Intel,DEV=0 HDA Intel, ALC1200 Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) ---------------------- aplay -L (default options) ---------------------- front:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers null Discard all samples (playback) or generate zero samples (capture) ---------------------- dmesg (with model=6stack-dig ) ---------------------- [ 8.442560] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 [ 8.442686] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 8.442690] ALSA sound/pci/hda/hda_intel.c:2196: chipset global capabilities = 0x4401 [ 8.466673] ALSA sound/pci/hda/hda_intel.c:779: codec_mask = 0x3 [ 8.466753] ALSA sound/pci/hda/hda_intel.c:1256: max_slots = 4, codec_mask = 3, codec_probe_mask = -1 [ 8.466889] ALSA sound/pci/hda/hda_intel.c:1206: hda_intel: codec #0 probed OK [ 8.467018] ALSA sound/pci/hda/hda_intel.c:1206: hda_intel: codec #1 probed OK [ 8.490429] usb 3-1: reset full speed USB device using uhci_hcd and address 2 [ 8.637617] lirc_dev: lirc_register_plugin: sample_rate: 0 [ 8.641615] lirc_mceusb2[2]: Philips eHome Infrared Transceiver on usb3:2 [ 8.641689] usbcore: registered new interface driver lirc_mceusb2 [ 9.470007] ALSA sound/pci/hda/hda_intel.c:627: hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0x200f0000 [ 10.473336] ALSA sound/pci/hda/hda_intel.c:1271: hda_intel: Codec #2 probe error; disabling it... [ 11.503338] ALSA sound/pci/hda/hda_intel.c:1271: hda_intel: Codec #3 probe error; disabling it... [ 11.531911] ALSA sound/pci/hda/hda_codec.c:2587: hda_codec: model '6stack-dig' is selected [ 11.558144] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Headphone Playback Volume, skipped [ 11.558237] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Speaker Playback Volume, skipped [ 11.558330] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Mono Playback Volume, skipped [ 11.558422] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Line-Out Playback Volume, skipped [ 11.558514] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave PCM Playback Volume, skipped [ 11.558610] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Speaker Playback Switch, skipped [ 11.558702] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Mono Playback Switch, skipped ************** playback on ************** [ 159.960545] ALSA sound/pci/hda/hda_intel.c:1473: azx_pcm_prepare: bufsize=0x10000, format=0x11 [ 159.960567] ALSA sound/pci/hda/hda_codec.c:841: hda_codec_setup_stream: NID=0x10, stream=0x5, channel=0, format=0x11 [ 161.613092] ALSA sound/pci/hda/hda_codec.c:853: hda_codec_cleanup_stream: NID=0x10 [ 161.613168] ALSA sound/pci/hda/hda_codec.c:853: hda_codec_cleanup_stream: NID=0x10 ************** playback off ************** ---------------------- dmesg (no parameters) ---------------------- [ 8.972060] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 [ 8.972187] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 8.972191] ALSA sound/pci/hda/hda_intel.c:2196: chipset global capabilities = 0x4401 [ 8.996673] ALSA sound/pci/hda/hda_intel.c:779: codec_mask = 0x3 [ 8.996754] ALSA sound/pci/hda/hda_intel.c:1256: max_slots = 4, codec_mask = 3, codec_probe_mask = -1 [ 8.996896] ALSA sound/pci/hda/hda_intel.c:1206: hda_intel: codec #0 probed OK [ 8.997016] ALSA sound/pci/hda/hda_intel.c:1206: hda_intel: codec #1 probed OK [ 10.000009] ALSA sound/pci/hda/hda_intel.c:627: hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0x200f0000 [ 11.003343] ALSA sound/pci/hda/hda_intel.c:1271: hda_intel: Codec #2 probe error; disabling it... [ 12.033342] ALSA sound/pci/hda/hda_intel.c:1271: hda_intel: Codec #3 probe error; disabling it... [ 12.061914] hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... [ 12.062480] ALSA sound/pci/hda/hda_codec.c:3303: autoconfig: line_outs=4 (0x14/0x15/0x16/0x17/0x0) [ 12.062482] ALSA sound/pci/hda/hda_codec.c:3307: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 12.062484] ALSA sound/pci/hda/hda_codec.c:3311: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 12.062485] ALSA sound/pci/hda/hda_codec.c:3312: mono: mono_out=0x0 [ 12.062487] ALSA sound/pci/hda/hda_codec.c:3320: inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x0, aux=0x0 [ 12.092487] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Headphone Playback Volume, skipped [ 12.092581] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Speaker Playback Volume, skipped [ 12.092673] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Mono Playback Volume, skipped [ 12.092764] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Line-Out Playback Volume, skipped [ 12.092856] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave PCM Playback Volume, skipped [ 12.092953] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Speaker Playback Switch, skipped [ 12.093043] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Mono Playback Switch, skipped

4 5

[alsa-devel] (no subject)
by Yang, Libin 17 Nov '08

17 Nov '08

Hi Takashi, This patch is to support detecting new AMD HD Audio devices with HDA PCI class code. Would you please review it? Thanks a lot! -------------------------------------------------------------------- Signed-off-by: Libin Yang <libin.yang(a)amd.com> --- alsa-driver-1.0.18.orig/alsa-kernel/pci/hda/hda_intel.c 2008-10-29 20:41:35.000000000 +0800 +++ alsa-driver-1.0.18/alsa-kernel/pci/hda/hda_intel.c 2008-11-11 18:33:14.000000000 +0800 @@ -291,6 +291,7 @@ /* Define VIA HD Audio Device ID*/ #define VIA_HDAC_DEVICE_ID 0x3288 +#define PCI_CLASS_MULTIMEDIA_HDA 0x040300 /* */ @@ -410,6 +411,7 @@ AZX_DRIVER_ULI, AZX_DRIVER_NVIDIA, AZX_DRIVER_TERA, + AZX_DRIVER_AMD_AUTO, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -423,6 +425,7 @@ [AZX_DRIVER_ULI] = "HDA ULI M5461", [AZX_DRIVER_NVIDIA] = "HDA NVidia", [AZX_DRIVER_TERA] = "HDA Teradici", + [AZX_DRIVER_AMD_AUTO] = "HDA AMD", }; /* @@ -1198,6 +1201,7 @@ static unsigned int azx_default_codecs[AZX_NUM_DRIVERS] __devinitdata = { [AZX_DRIVER_ICH] = 3, [AZX_DRIVER_ATI] = 3, + [AZX_DRIVER_AMD_AUTO] = 3, }; static int __devinit azx_codec_create(struct azx *chip, const char *model, @@ -2146,6 +2150,7 @@ chip->playback_streams = ULI_NUM_PLAYBACK; chip->capture_streams = ULI_NUM_CAPTURE; break; + case AZX_DRIVER_AMD_AUTO: case AZX_DRIVER_ATIHDMI: chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; @@ -2373,6 +2378,9 @@ { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, + /* AMD Generic, PCI class code and Vendor ID for HD Audio */ + { PCI_VENDOR_ID_ATI, PCI_ANY_ID, PCI_ANY_ID, PCI_ANY_ID, + PCI_CLASS_MULTIMEDIA_HDA, 0xffffff, AZX_DRIVER_AMD_AUTO }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); Thanks & Regards, Libin

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[alsa-devel] Chip: SigmaTel ID 7645
by Thomas Dietterle 16 Nov '08

16 Nov '08

Sehr geehrte Damen und Herren, ich habe ein Mainboard mit eingebauter Soundkarte. Leider bekomme ich keinen vernüftigen Sound zu hören, der Klang kratzt und verliert sich im Echo. head -n 1 /proc/asound/card0/codec* Codec: SigmaTel ID 7645 hwinfo --sound 22: PCI 07.0: 0403 Audio device [Created at pci.301] UDI: /org/freedesktop/Hal/devices/pci_10de_774 Unique ID: M71A.I3CGe6sNjHC SysFS ID: /devices/pci0000:00/0000:00:07.0 SysFS BusID: 0000:00:07.0 Hardware Class: sound Model: "Elitegroup Audio device" Vendor: pci 0x10de "nVidia Corporation" Device: pci 0x0774 SubVendor: pci 0x1019 "Elitegroup Computer Systems" SubDevice: pci 0x2666 Revision: 0xa1 Driver: "HDA Intel" Driver Modules: "snd_hda_intel" Memory Range: 0xfce78000-0xfce7bfff (rw,non-prefetchable) IRQ: 20 (9832 events) Module Alias: "pci:v000010DEd00000774sv00001019sd00002666bc04sc03i00" Driver Info #0: Driver Status: snd_hda_intel is active Driver Activation Cmd: "modprobe snd_hda_intel" Config Status: cfg=no, avail=yes, need=no, active=unknown lsmod| grep snd snd_pcm_oss 50432 0 snd_mixer_oss 20096 1 snd_pcm_oss snd_seq 54452 0 snd_seq_device 12172 1 snd_seq snd_hda_intel 273180 0 snd_pcm 82564 2 snd_pcm_oss,snd_hda_intel snd_timer 26756 2 snd_seq,snd_pcm snd 58164 7 snd_pcm_oss,snd_mixer_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_pcm,snd_timer soundcore 11460 1 snd snd_page_alloc 14472 2 snd_hda_intel,snd_pcm || Suse 10.3 uname -a Linux linux-winxp 2.6.22.19-0.1-default #1 SMP 2008-10-14 22:17:43 +0200 i686 athlon i386 GNU/Linux Woran könnte es ihrer Meinung noch liegen?

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[alsa-devel] Alsa driver won't compile on Ubuntu kernel 2.6.24-16-generic
by Klaus Schulz 16 Nov '08

16 Nov '08

Hi folks. As you know I am maintaining an Alsa-Installation script over at Ubuntu forums: http://ubuntuforums.org/showthread.php?t=962695 Admin: I intend to post problems over here if they're mentioned more than once, if you don't mind. I ask people to upload the Alsa log file. Perhaps you can give me hint how to set this process up properly. !These are not my problems. I just try to channel at least some of them. Here is the first problem: Alsa driver won't compile on Ubuntu kernel 2.6.24-16-generic make -C /lib/modules/2.6.24-16-generic/build SUBDIRS=/usr/src/Alsa-1.0.18a/alsa-driver-1.0.18a CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/linux-headers-2.6.24-16-generic' CC [M] /usr/src/Alsa-1.0.18a/alsa-driver-1.0.18a/acore/hrtimer.o In file included from /usr/src/Alsa-1.0.18a/alsa-driver-1.0.18a/acore/hrtimer.c:2: /usr/src/Alsa-1.0.18a/alsa-driver-1.0.18a/acore/../alsa-kernel/core/hrtimer.c: In function âsnd_hrtimer_callbackâ: /usr/src/Alsa-1.0.18a/alsa-driver-1.0.18a/acore/../alsa-kernel/core/hrtimer.c:29: error: implicit declaration of function âhrtimer_forward_nowâ make[3]: *** [/usr/src/Alsa-1.0.18a/alsa-driver-1.0.18a/acore/hrtimer.o] Error 1 make[2]: *** [/usr/src/Alsa-1.0.18a/alsa-driver-1.0.18a/acore] Error 2 make[1]: *** [_module_/usr/src/Alsa-1.0.18a/alsa-driver-1.0.18a] Error 2 make[1]: Leaving directory `/usr/src/linux-headers-2.6.24-16-generic' make: *** [compile] Error Any ideas? Cheers Klaus

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[alsa-devel] ALC883 recording broken
by Dmitry Nezhevenko 16 Nov '08

16 Nov '08

Hi. I'm trying to get sound recording work on my MSI M673 laptop. Currently I'm running 2.6.27 kernel with in-kernel alsa. It looks like something going really wrong. I've connected headphones to laptop, unmuted mic and configured it's gain to maximum. After this I've tried to change "Mic boost" value. First of all, "Mic boost" under "Capture" view of alsamixer don't work at all (I can't hear any changes). "Mic boost" under "Playback" works (at least I can hear less noise with 0%, more noise with 33%, more with 67%. and, that's strange, but less with 100%. (100% looks like same as 33%). I've tried all model=xxx variants for my ALC883 codec but with no success. arecord just records silence. (Mic was unmuted and Mic boost is set to maximum). Also just after modprobing snd_hda_intel with model=acer-aspire some led in headphones jack lights.. I don't know what does this means. Ubuntu 7.10 with 2.6.22 kernel and alsa 1.0.14 works a bit different. Recording works with model=acer-aspire but playback is broken (just silence). It really looks like some codec misconfiguration (probably it tries to record from incorrect line or something like this). I've found one thread at LKML (http://lkml.org/lkml/2008/6/9/331) about this codec. And it refers to some hda-verb utility that should be able to adjust some codec settings. Is there any manual available? Or any other suggestions, what I should try to debug this? I'm attaching alsa-info.sh output -- WBR, Dmitry

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Re: [alsa-devel] Fwd: Undelivered Mail Returned to Sender
by Andrei Tanas 16 Nov '08

16 Nov '08

>> >> And here's "cat /proc/asound/card0/codec#0" >> > >> > This is the regular codec. >> > Are there /proc/asound/card0/codec#{1,2,3}? >> >> There was only one. >> >> > The HDMI codec would look like this one: >> > % cat /proc/asound/card0/codec\#3 >> >> >> I saw nothing like that. >> Does your main board have an external physical HDMI interface? >> If true, then there is an ALSA bug. Yes, there's physical HDMI port on the card, which, according to the documentation, supports both video and audio. However, there's also "SPDIF OUT Mode Setting" in the BIOS allowing to choose between "HDMI Output" or "SPDIF Output", suggesting that only one can be used at a time. The description is not terribly clear though. >> Will it help to enable SND_VERBOSE_PRINTK and SND_DEBUG? > Sure it could help, and be sure to use the latest sound-2.6 git tree :-) Ok, I've done this. Here's "aplay -l" **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 And here's "dmesg" [ 8.513848] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 [ 8.514001] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 8.514005] ALSA sound/pci/hda/hda_intel.c:2187: chipset global capabilities = 0x4401 [ 8.543345] ALSA sound/pci/hda/hda_intel.c:779: codec_mask = 0x3 [ 8.543487] ALSA sound/pci/hda/hda_intel.c:1206: hda_intel: codec #0 probed OK [ 8.545462] ALSA sound/pci/hda/hda_codec.c:2587: hda_codec: model '6stack-dig' is selected [ 8.574845] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Headphone Playback Volume, skipped [ 8.574947] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Speaker Playback Volume, skipped [ 8.575043] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Mono Playback Volume, skipped [ 8.575135] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Line-Out Playback Volume, skipped [ 8.575231] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave PCM Playback Volume, skipped [ 8.575337] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Speaker Playback Switch, skipped [ 8.575435] ALSA sound/pci/hda/hda_codec.c:1276: Cannot find slave Mono Playback Switch, skipped [ 105.312682] ALSA sound/pci/hda/hda_codec.c:853: hda_codec_cleanup_stream: NID=0x6 [ 106.707933] ALSA sound/pci/hda/hda_intel.c:1464: azx_pcm_prepare: bufsize=0x10000, format=0x11 [ 106.707951] ALSA sound/pci/hda/hda_codec.c:841: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x11 .......... [ 110.479724] ALSA sound/pci/hda/hda_intel.c:1464: azx_pcm_prepare: bufsize=0x10000, format=0x11 [ 110.479740] ALSA sound/pci/hda/hda_codec.c:841: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x11 [ 200.715073] ALSA sound/pci/hda/hda_codec.c:853: hda_codec_cleanup_stream: NID=0x6

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[alsa-devel] [PATCH] hda: STAC_VREF_EVENT value change
by Matthew Ranostay 15 Nov '08

15 Nov '08

Changed value for STAC_VREF_EVENT from 0x40 to 0x00 because the unsol response value is only 6-bits width and the former value was 1<<6 which is an overrun. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index ec8f5ab..86e5cd9 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -36,10 +36,10 @@ #include "hda_patch.h" #include "hda_beep.h" +#define STAC_VREF_EVENT 0x00 #define STAC_INSERT_EVENT 0x10 #define STAC_PWR_EVENT 0x20 #define STAC_HP_EVENT 0x30 -#define STAC_VREF_EVENT 0x40 enum { STAC_REF,

2 1

[alsa-devel] xfi fatality pro gamer
by Dmitri Seletski 15 Nov '08

15 Nov '08

Hello boys and girls. I have a card (actually 3 of them, hda-intel, that doesn't work properly, usb-audio, that doesn't work properly and xfi fatality gamer pro, which you guessed by now doesn't work properly). I may test things from any of those cards if you have any interest. I know my fatality card is rather new on the market(not exactly, but...) and not very well supported. I would like to test it if you need more testers. I have got driver released on 6th of November 2008. I tryed OSS with it, it sucked badly, but at least proved that card is in working conditions. With this driver that creative released i am getting my sound applications failed miserably. I know it's crap, so instead of complaining I decided to do something about it this time. I can provide dmesg output. Anyway, let me know if you need a guinea pig(I know how to load a distro from live cd, worst case scenario, chrootig or something). Restecpa me brothers and sistas. Dmitri

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[alsa-devel] [PATCH RESEND 0/3] ALSA SOC driver for s3c24xx with uda134x
by christian pellegrin 15 Nov '08

15 Nov '08

This patch adds support for the UDA134X codec and a sound card definition for one of these UDAs connected to the s3c24xx. It is *heavily* based on the one made by Zoltan Devai but: 1 since the UDA is the only use of the L3 protocol I can find I just embedded a stripped-down L3 bit-banging algorithm from the original RMK work. It is really small. 2 the driver has the possibility to specify the pins used by codec via platform data so it can work on SMDK2410, SMDK2440 or any custom design. 3 it tries to guess the right clock source and divider so it is not tied to a particular crystal used. 4 it fixes some bugs. -- Christian Pellegrin, see http://www.evolware.org/chri/ "Real Programmers don't play tennis, or any other sport which requires you to change clothes. Mountain climbing is OK, and Real Programmers wear their climbing boots to work in case a mountain should suddenly spring up in the middle of the computer room."

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[alsa-devel] How does usbaudio synchronize on playback?
by John Crowley 15 Nov '08

15 Nov '08

All: How is the playback data in usbaudio.c synchronized between the application writing data to the runtime->dma_area buffer and the prepare_playback_urb function copying this into URBs? To start, a series of no-data URBs are sent out. The completion routine for each URB then calls prepare_playback_urb, which copies more data into the URB, and this URB is then passed to the USB system to be sent to the device. prepare_playback_urb copies data from the runtime->dma_area to the urb->transfer_buffer area. It maintains subs->hwptr_done to record what data has been copied, and wraps at runtime->buffer_size since this is a circular buffer. It also recognizes when runtime->period_size has been exceeded and invokes snd_pcm_period_elapsed to tell the application that more playback data is needed. What I don't understand is how does prepare_playback_urb ensure that more playback data has actually been written into the buffer? In our case, we are streaming audio over a WAN, so we increased the number of URBs to 16, with up to 32 packets per URB, to get a lot of data going on-the-fly over a high latency link. The problem is that we are getting erroneous data, and suspect that it is caused by this synchronization issue. When the initial no-data URBs are completed, the prepare_playback_urb logic copies a lot of data from the buffer into new URBs -- but apparently no playback data has been written into the buffer yet by the application. If we insert logic to arbitrarily slow down the prepare_playback_urb logic, then the data appears to be correct. But this is a (very) weak solution. We also tried to save the runtime->control->appl_ptr value in order to calculate how much new application data is available, but discovered that appl_ptr is evidently not being updated (always zero). (running 2.6.23 kernel). Are we missing a key concept? Is there a way to reliably determine how much data the application has actually written into the buffer? Thanks for your help ... John Crowley ScreenPC, Inc. 101 Merritt 7, 5th Fl Norwalk, CT 06851 Office: 203.842.0460 ext 602 Cell: 203.856.2396 This message and any accompanying attachments contains information from ScreenPC,Inc. that is confidential and/or privileged. The information is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply email and destroy all copies of the original email. No virus found in this outgoing message. Checked by AVG. Version: 7.5.549 / Virus Database: 270.9.2 - Release Date: 11/11/2008 12:00 AM

2 2

[alsa-devel] Using period events
by Bankim Bhavsar 14 Nov '08

14 Nov '08

Hi, I used async notification mechanism provided by alsa-lib. Since they use signals I ran into all sorts of issues. I should have read this post before http://mailman.alsa-project.org/pipermail/alsa-devel/2008-May/008030.html Definitely not the way to go ahead. I second the argument to remove that feature from alsa-lib altogether. We need a mechanism wherein the alsa subsystem notifies a polling thread that a "period" of sound bytes has been completed. Using snd_pcm_sw_params_set_period_event() will suffice? I took a cursory look at the alsa-lib and it seems a timer is used for this purpose. Is this well supported on majority of sound cards out there (alsa version 1.0.17+)? I want to make sure before I use this alternative approach. Thanks, Bankim.

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Re: [alsa-devel] Bugs on aspire one A150
by Maxim Levitsky 14 Nov '08

14 Nov '08

Alan Jenkins wrote: > Maxim Levitsky wrote: >> I have just bought an Aspire one A150, XP version, >> as it was the only available here, and installed ubuntu on it. >> >> Bugs I discovered so far: >> >> >> ** 1 - embedded controler works in polling mode, due to this: >> >> [ 0.708571] ACPI: EC: non-query interrupt received, switching to interrupt mode >> [ 1.224043] ACPI: EC: missing confirmations, switch off interrupt mode. >> >> >> Maybe this is the reason for the fact that gnome power manager freezes when I unplug >> the AC, and freezes often when I try to see battery status. >> >> >> (Note: same is seen on my acer aspire 5720) > > That sounds like a known issue. It has been resolved by "ACPI: EC: revert msleep patch". Happily Len submitted it for mainline this week. You will also find it if you try the acpi-test git tree. We're all hoping 2.6.28 will be much improved in terms of reliable operation of different ECs :). > >> ** 2 - wireless: not to mention the fact that ath5k wasn't installed by default in ubuntu... >> wireless more or less works, but kernel log is full of backtraces. > > Well, that doesn't tell us much. Did they still happen after upgrading to 2.6.28-rc3? Can we see them? > >> Was able to connect to my WPA2 access point. >> Sometimes wireless fails completely, especially after suspend to ram. >> Advanced features like monitor/injection work, but when I changed the card's >> mac address it stopped working. >> I also noticed that if I then start airodump, then wireless works with new mac. >> >> >> >> ** 3 - internal mic doesn't work. >> tried model=acer, model=auto. >> Overall it seems that alsa misprograms O/B realteck >> codec. >> I talk about this later. >> Have same issue on my acer 5720 >> >> >> >> ** 4 - wireless led doesn't work. >> ath5k devs, can you fix this? >> >> >> ** 5 - coretemp doesn't show cpu temperature, >> I have seen somewhere that atom support same thermal diode as core2 >> and only patch to detect it is needed. >> Please include such path in 2.6.28 if exists. Patch for that does exits, but doesn't apply to latest git, I will apply in manually. This should go to .28 I think, this is trivial thing. >> >> >> ** 6 - both card readers are missing from lspci, is this normal? > > A similar bug has been reported as a regression: > > <http://bugzilla.kernel.org/show_bug.cgi?id=11828> > > so one assumes that it worked on the machines with linux pre-installed. Hopefully without requiring any hacks. > > It seems that for now a workaround may be to pass the option debug_quirks=1 to the sdhci module... > > <http://marc.info/?l=linux-kernel&m=122509648027303&w=2> > > ...or that it may help if you insert an SD card before booting. > > Apparently the reporter also investigated pcie hotplug. Probably the BIOS doesn't provide the normal support. You can try "modprobe pciehp pciehp_force=1", maybe it helps the kernel discover the devices. It worked for something else on my EeePC. But then it will reportedly disappear the ethernet controller. > > However, at the moment pciehp can cause delays of 10s of seconds during resume. Got, a SD card, and with help of pciehp it works almost perfectly in both slots. Almost, due to the fact that R/O switch is ignored. (Just as I expected, a s/w switch, couldn't they think more, and include a hardware switch?) On my main acer R/O switch works. However I have to use pciehp_force=1, I understand that probably acpi tables are broken, but could you add a workaround (dmi quirk?) Also, there is a acpiphp driver, which doesn't work here. pciehp is supposed to support so called native mode, don't yet know what it is, but here it still uses acpi. > >> ** 7 - opengl is broken, I tried running neverball in fullscreen, but it shows wrong/corrupted >> textures (compiz is running, and working well thought) I almost sure why this happens, opengl application is out of textures, I need to install gem enabled intel driver here, and see how it works. Best regards, Maxim Levitsky

1 0

[alsa-devel] problem with snd_hda_intel and ALC889a
by G.Bitzer 14 Nov '08

14 Nov '08

Hi All, I have a problem I think caused by ALSA. But first the facts. My box runs Gentoo Linux, kernel 2.6.25-gentoo-r7, amd64, on a Gigabyte board with the following sound hardware (excerpt from lspci -vv): 00:14.2 Audio device: ATI Technologies Inc SB600 Azalia Subsystem: Giga-byte Technology Device a002 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 32, Cache Line Size: 4 bytes Interrupt: pin ? routed to IRQ 16 Region 0: Memory at fe024000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) There is a AMD 790X chipset on the board, and a ALC889a codec. According to 'grep Code /proc/asound/card0/codec\#3' a 'Codec: Realtek ALC885' is detected. All software is self compiled, as usual with Gentoo. I can play .ogg or .mp3 files with audacious, no problem so far, also viewing/listening to .wmv files or YouTube works. But when pcm sounds are played from some games i constantly get dropouts and errors like snd_pcm_writei failed: Broken pipe preparing audio device for output snd_pcm_writei failed: Broken pipe preparing audio device for output ... or snd_pcm_writei failed: Resource temporarily unavailable snd_pcm_writei failed: Resource temporarily unavailable snd_pcm_writei failed: Resource temporarily unavailable snd_pcm_writei failed: Resource temporarily unavailable snd_pcm_writei short write: 987 out of 1024 snd_pcm_writei failed: Resource temporarily unavailable snd_pcm_writei failed: Resource temporarily unavailable idAudioHardwareALSA::Write: 1024 frames overflowed and dropped snd_pcm_writei short write: 1242 out of 2048 snd_pcm_writei failed: Resource temporarily unavailable idAudioHardwareALSA::Write: 1024 frames overflowed and dropped snd_pcm_writei failed: Resource temporarily unavailable snd_pcm_writei failed: Resource temporarily unavailable After disabling 'Aggressive power-saving on HD-audio SND_HDA_POWER_SAVE' and 'AC97 Power-Saving Mode SND_AC97_POWER_SAVE' I got less dropouts, but they were still noticable. Last but not least, my alsa is media-libs/alsa-lib-1.0.16 media-sound/alsa-headers-1.0.16 media-sound/alsa-utils-1.0.16 This are the newest ones marked as stable in portage. What can I do ? Are there any parameters for the kernel modules, to get good pcm sound ? Thanks in advance PS: I can send a report of alsa-info.sh, if this is helps

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[alsa-devel] [PATCH RESEND 1/2] ALSA SOC driver for s3c24xx with uda134x
by Christian Pellegrin 14 Nov '08

14 Nov '08

This patch adds a sound card definition for one of these UDAs connected to the s3c24xx. It is *heavily* based on the one made by Zoltan Devai but: * the driver has the possibility to specify the pins used by codec via platform data so it can work on SMDK2410, SMDK2440 or any custom design. * it tries to guess the right clock source and divider so it is not tied to a particular crystal used. Generated on 20081113 against f7e1798a78a30bae1cf3ebc3e1b6f7c1bac57756 Signed-off-by: Christian Pellegrin <chripell(a)fsfe.org> --- include/sound/s3c24xx_uda134x.h | 14 ++ sound/soc/s3c24xx/Kconfig | 5 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c24xx_uda134x.c | 347 +++++++++++++++++++++++++++++++++++ 4 files changed, 368 insertions(+), 0 deletions(-) diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h new file mode 100644 index 0000000..33df4cb --- /dev/null +++ b/include/sound/s3c24xx_uda134x.h @@ -0,0 +1,14 @@ +#ifndef _S3C24XX_UDA134X_H_ +#define _S3C24XX_UDA134X_H_ 1 + +#include <sound/uda134x.h> + +struct s3c24xx_uda134x_platform_data { + int l3_clk; + int l3_mode; + int l3_data; + void (*power) (int); + int model; +}; + +#endif diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b9f2353..fcd03ac 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -44,3 +44,8 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650 Say Y if you want to add support for SoC audio on ln2440sbc with the ALC650. +config SND_S3C24XX_SOC_S3C24XX_UDA134X + tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_UDA134X diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 0aa5fb0..96b3f3f 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o +snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o +obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c new file mode 100644 index 0000000..2d9a40d --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -0,0 +1,347 @@ +/* + * Modifications by Christian Pellegrin <chripell(a)evolware.org> + * + * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/mutex.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/s3c24xx_uda134x.h> +#include <sound/uda134x.h> + +#include <asm/plat-s3c24xx/regs-iis.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" + + +/* #define ENFORCE_RATES 1 */ +/* + Unfortunately the S3C24XX in master mode has a limited capacity of + generating the clock for the codec. If you define this only rates + that are really available will be enforced. But be careful, most + user level application just want the usual sampling frequencies (8, + 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly + operation for embedded systems. So if you aren't very lucky or your + hardware engineer wasn't very forward-looking it's better to leave + this undefined. If you do so an approximate value for the requested + sampling rate in the range -/+ 5% will be chosen. If this in not + possible an error will be returned. +*/ + +static struct clk *xtal; +static struct clk *pclk; +/* this is need because we don't have a place where to keep the + * pointers to the clocks in each substream. We get the clocks only + * when we are actually using them so we don't block stuff like + * frequency change or oscillator power-off */ +static int clk_users; +static DEFINE_MUTEX(clk_lock); + +static unsigned int rates[33 * 2]; +#ifdef ENFORCE_RATES +static struct snd_pcm_hw_constraint_list hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; +#endif + +static struct platform_device *s3c24xx_uda134x_snd_device; + +int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) +{ + int ret = 0; +#ifdef ENFORCE_RATES + struct snd_pcm_runtime *runtime = substream->runtime;; +#endif + + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + if (clk_users == 0) { + xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); + if (!xtal) { + printk(KERN_ERR "%s cannot get xtal\n", __func__); + ret = -EBUSY; + } else { + pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, + "pclk"); + if (!pclk) { + printk(KERN_ERR "%s cannot get pclk\n", + __func__); + clk_put(xtal); + ret = -EBUSY; + } + } + if (!ret) { + int i, j; + + for (i = 0; i < 2; i++) { + int fs = i ? 256 : 384; + + rates[i*33] = clk_get_rate(xtal) / fs; + for (j = 1; j < 33; j++) + rates[i*33 + j] = clk_get_rate(pclk) / + (j * fs); + } + } + } + clk_users += 1; + mutex_unlock(&clk_lock); + if (!ret) { +#ifdef ENFORCE_RATES + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_constraints_rates); + if (ret < 0) + printk(KERN_ERR "%s cannot set constraints\n", + __func__); +#endif + } + return ret; +} + +void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) +{ + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + clk_users -= 1; + if (clk_users == 0) { + clk_put(xtal); + xtal = NULL; + clk_put(pclk); + pclk = NULL; + } + mutex_unlock(&clk_lock); +} + +static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + int ret = 0; + int clk_source, fs_mode; + unsigned long rate = params_rate(params); + long err, cerr; + unsigned int div; + int i, bi; + + err = 999999; + bi = 0; + for (i = 0; i < 2*33; i++) { + cerr = rates[i] - rate; + if (cerr < 0) + cerr = -cerr; + if (cerr < err) { + err = cerr; + bi = i; + } + } + if (bi / 33 == 1) + fs_mode = S3C2410_IISMOD_256FS; + else + fs_mode = S3C2410_IISMOD_384FS; + if (bi % 33 == 0) { + clk_source = S3C24XX_CLKSRC_MPLL; + div = 1; + } else { + clk_source = S3C24XX_CLKSRC_PCLK; + div = bi % 33; + } + pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); + + clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; + pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, + fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", + clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", + div, clk, err); + + if ((err * 100 / rate) > 5) { + printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " + "too different from desired (%ld%%)\n", + err * 100 / rate); + return -EINVAL; + } + + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + fs_mode); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops s3c24xx_uda134x_ops = { + .startup = s3c24xx_uda134x_startup, + .shutdown = s3c24xx_uda134x_shutdown, + .hw_params = s3c24xx_uda134x_hw_params, +}; + +static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { + .name = "UDA134X", + .stream_name = "UDA134X", + .codec_dai = &uda134x_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &s3c24xx_uda134x_ops, +}; + +static struct snd_soc_machine snd_soc_machine_s3c24xx_uda134x = { + .name = "S3C24XX_UDA134X", + .dai_link = &s3c24xx_uda134x_dai_link, + .num_links = 1, +}; + +static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins; + +static void setdat(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0); +} + +static void setclk(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0); +} + +static void setmode(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0); +} + +static struct uda134x_platform_data s3c24xx_uda134x = { + .l3 = { + .setdat = setdat, + .setclk = setclk, + .setmode = setmode, + .data_hold = 1, + .data_setup = 1, + .clock_high = 1, + .mode_hold = 1, + .mode = 1, + .mode_setup = 1, + }, +}; + +static struct snd_soc_device s3c24xx_uda134x_snd_devdata = { + .machine = &snd_soc_machine_s3c24xx_uda134x, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_uda134x, + .codec_data = &s3c24xx_uda134x, +}; + +static int s3c24xx_uda134x_probe(struct platform_device *pdev) +{ + int ret; + + printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n"); + + s3c24xx_uda134x_l3_pins = pdev->dev.platform_data; + if (s3c24xx_uda134x_l3_pins == NULL) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "unable to find platform data\n"); + return -ENODEV; + } + s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power; + s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model; + gpio_direction_output(s3c24xx_uda134x_l3_pins->l3_data, 0); + gpio_direction_output(s3c24xx_uda134x_l3_pins->l3_clk, 0); + gpio_direction_output(s3c24xx_uda134x_l3_pins->l3_mode, 0); + + s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_uda134x_snd_device) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(s3c24xx_uda134x_snd_device, + &s3c24xx_uda134x_snd_devdata); + s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev; + ret = platform_device_add(s3c24xx_uda134x_snd_device); + if (ret) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); + platform_device_put(s3c24xx_uda134x_snd_device); + } + + return ret; +} + +static int s3c24xx_uda134x_remove(struct platform_device *pdev) +{ + platform_device_unregister(s3c24xx_uda134x_snd_device); + return 0; +} + +static struct platform_driver s3c24xx_uda134x_driver = { + .probe = s3c24xx_uda134x_probe, + .remove = s3c24xx_uda134x_remove, + .driver = { + .name = "s3c24xx_uda134x", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c24xx_uda134x_init(void) +{ + return platform_driver_register(&s3c24xx_uda134x_driver); +} + +static void __exit s3c24xx_uda134x_exit(void) +{ + platform_driver_unregister(&s3c24xx_uda134x_driver); +} + + +module_init(s3c24xx_uda134x_init); +module_exit(s3c24xx_uda134x_exit); + +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); +MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); -- 1.4.4.4

2 2

[alsa-devel] [PATCH RESEND 0/2] ALSA SOC driver for uda134x codec
by Christian Pellegrin 14 Nov '08

14 Nov '08

This patch adds support for the UDA1341 codec. It is *heavily* based on the one made by Zoltan Devai but: since the UDA is the only use of the L3 protocol I can find I just embedded a stripped-down L3 bit-banging algorithm from the original RMK work. It is really small. Generated on 20081113 against f7e1798a78a30bae1cf3ebc3e1b6f7c1bac57756 Signed-off-by: Christian Pellegrin <chripell(a)fsfe.org> --- include/sound/l3.h | 18 ++ include/sound/uda134x.h | 58 ++++ sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/l3.c | 91 +++++++ sound/soc/codecs/uda134x.c | 636 ++++++++++++++++++++++++++++++++++++++++++++ 6 files changed, 815 insertions(+), 0 deletions(-) diff --git a/include/sound/l3.h b/include/sound/l3.h new file mode 100644 index 0000000..423a08f --- /dev/null +++ b/include/sound/l3.h @@ -0,0 +1,18 @@ +#ifndef _L3_H_ +#define _L3_H_ 1 + +struct l3_pins { + void (*setdat)(int); + void (*setclk)(int); + void (*setmode)(int); + int data_hold; + int data_setup; + int clock_high; + int mode_hold; + int mode; + int mode_setup; +}; + +int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len); + +#endif diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h new file mode 100644 index 0000000..df29064 --- /dev/null +++ b/include/sound/uda134x.h @@ -0,0 +1,58 @@ +/* + * uda134x.h -- UDA134x ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * 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. + */ + +#ifndef _UDA134X_H +#define _UDA134X_H + +#include <sound/l3.h> + +#define UDA134X_L3ADDR 5 +#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0) +#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1) +#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2) + +#define UDA134X_EXTADDR_PREFIX 0xC0 +#define UDA134X_EXTDATA_PREFIX 0xE0 + +/* UDA134X registers */ +#define UDA134X_EA000 0 +#define UDA134X_EA001 1 +#define UDA134X_EA010 2 +#define UDA134X_EA011 3 +#define UDA134X_EA100 4 +#define UDA134X_EA101 5 +#define UDA134X_EA110 6 +#define UDA134X_EA111 7 +#define UDA134X_STATUS0 8 +#define UDA134X_STATUS1 9 +#define UDA134X_DATA000 10 +#define UDA134X_DATA001 11 +#define UDA134X_DATA010 12 +#define UDA134X_DATA1 13 + +#define UDA134X_REGS_NUM 14 + +#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4)) + +extern struct snd_soc_dai uda134x_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda134x; + +struct uda134x_platform_data { + struct l3_pins l3; + void (*power) (int); + int model; +#define UDA134X_UDA1340 0 +#define UDA134X_UDA1341 1 +#define UDA134X_UDA1344 2 +}; + +#endif /* _UDA134X_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b73c36a..7756e5b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8510 if (I2C || SPI_MASTER) select SND_SOC_WM8580 if I2C @@ -84,6 +85,13 @@ config SND_SOC_TWL4030 tristate depends on TWL4030_CORE +config SND_SOC_L3 + tristate + +config SND_SOC_UDA134X + tristate + select SND_SOC_L3 + config SND_SOC_UDA1380 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3b9b58a..9af993e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -8,6 +8,8 @@ snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-l3-objs := l3.o +snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8580-objs := wm8580.o @@ -31,6 +33,8 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o +obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c new file mode 100644 index 0000000..93dbbe9 --- /dev/null +++ b/sound/soc/codecs/l3.c @@ -0,0 +1,91 @@ +/* + * L3 code + * + * Copyright (C) 2008, Christian Pellegrin <chripell(a)evolware.org> + * + * 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. + * + * + * based on: + * + * L3 bus algorithm module. + * + * Copyright (C) 2001 Russell King, All Rights Reserved. + * + * + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/delay.h> + +#include <sound/l3.h> + +/* + * Send one byte of data to the chip. Data is latched into the chip on + * the rising edge of the clock. + */ +static void sendbyte(struct l3_pins *adap, unsigned int byte) +{ + int i; + + for (i = 0; i < 8; i++) { + adap->setclk(0); + udelay(adap->data_hold); + adap->setdat(byte & 1); + udelay(adap->data_setup); + adap->setclk(1); + udelay(adap->clock_high); + byte >>= 1; + } +} + +/* + * Send a set of bytes to the chip. We need to pulse the MODE line + * between each byte, but never at the start nor at the end of the + * transfer. + */ +static void sendbytes(struct l3_pins *adap, const u8 *buf, + int len) +{ + int i; + + for (i = 0; i < len; i++) { + if (i) { + udelay(adap->mode_hold); + adap->setmode(0); + udelay(adap->mode); + } + adap->setmode(1); + udelay(adap->mode_setup); + sendbyte(adap, buf[i]); + } +} + +int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len) +{ + adap->setclk(1); + adap->setdat(1); + adap->setmode(1); + udelay(adap->mode); + + adap->setmode(0); + udelay(adap->mode_setup); + sendbyte(adap, addr); + udelay(adap->mode_hold); + + sendbytes(adap, data, len); + + adap->setclk(1); + adap->setdat(1); + adap->setmode(0); + + return len; +} +EXPORT_SYMBOL(l3_write); + +MODULE_DESCRIPTION("L3 bit-banging driver"); +MODULE_AUTHOR("Christian Pellegrin <chripell(a)evolware.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c new file mode 100644 index 0000000..b067e44 --- /dev/null +++ b/sound/soc/codecs/uda134x.c @@ -0,0 +1,636 @@ +/* + * uda134x.c -- UDA134X ALSA SoC Codec driver + * + * Modifications by Christian Pellegrin <chripell(a)evolware.org> + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <sound/uda134x.h> +#include <sound/l3.h> + +#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 +#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) + +struct uda134x_priv { + int sysclk; + int dai_fmt; + + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + +/* In-data addresses are hard-coded into the reg-cache values */ +static const char uda134x_reg[UDA134X_REGS_NUM] = { + /* Extended address registers */ + 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, + /* Status, data regs */ + 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, +}; + +/* + * The codec has no support for reading its registers except for peak level... + */ +static inline unsigned int uda134x_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA134X_REGS_NUM) + return -1; + return cache[reg]; +} + +/* + * Write the register cache + */ +static inline void uda134x_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA134X_REGS_NUM) + return; + cache[reg] = value; +} + +/* + * Write to the uda134x registers + * + */ +static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + u8 addr; + u8 data = value; + struct uda134x_platform_data *pd = codec->control_data; + + pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); + + if (reg >= UDA134X_REGS_NUM) { + printk(KERN_ERR "%s unkown register: reg: %d", + __func__, reg); + return -EINVAL; + } + + uda134x_write_reg_cache(codec, reg, value); + + switch (reg) { + case UDA134X_STATUS0: + case UDA134X_STATUS1: + addr = UDA134X_STATUS_ADDR; + break; + case UDA134X_DATA000: + case UDA134X_DATA001: + case UDA134X_DATA010: + addr = UDA134X_DATA0_ADDR; + break; + case UDA134X_DATA1: + addr = UDA134X_DATA1_ADDR; + break; + default: + /* It's an extended address register */ + addr = (reg | UDA134X_EXTADDR_PREFIX); + + ret = l3_write(&pd->l3, + UDA134X_DATA0_ADDR, &addr, 1); + if (ret != 1) + return -EIO; + + addr = UDA134X_DATA0_ADDR; + data = (value | UDA134X_EXTDATA_PREFIX); + break; + } + + ret = l3_write(&pd->l3, + addr, &data, 1); + if (ret != 1) + return -EIO; + + return 0; +} + +static inline void uda134x_reset(struct snd_soc_codec *codec) +{ + u8 reset_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS0); + uda134x_write(codec, UDA134X_STATUS0, reset_reg | (1<<6)); + msleep(1); + uda134x_write(codec, UDA134X_STATUS0, reset_reg & ~(1<<6)); +} + +static int uda134x_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = uda134x_read_reg_cache(codec, UDA134X_DATA010); + + pr_debug("%s mute: %d\n", __func__, mute); + + if (mute) + mute_reg |= (1<<2); + else + mute_reg &= ~(1<<2); + + uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2)); + + return 0; +} + +static int uda134x_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + struct snd_pcm_runtime *master_runtime; + + if (uda134x->master_substream) { + master_runtime = uda134x->master_substream->runtime; + + pr_debug("%s costrining to %d bits at %d\n", __func__, + master_runtime->sample_bits, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + uda134x->slave_substream = substream; + } else + uda134x->master_substream = substream; + + return 0; +} + +static void uda134x_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + + if (uda134x->master_substream == substream) + uda134x->master_substream = uda134x->slave_substream; + + uda134x->slave_substream = NULL; +} + +static int uda134x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + u8 hw_params; + + if (substream == uda134x->slave_substream) { + pr_debug("%s ignoring hw_params for slave substream\n", + __func__); + return 0; + } + + hw_params = uda134x_read_reg_cache(codec, UDA134X_STATUS0); + hw_params &= STATUS0_SYSCLK_MASK; + hw_params &= STATUS0_DAIFMT_MASK; + + pr_debug("%s sysclk: %d, rate:%d\n", __func__, + uda134x->sysclk, params_rate(params)); + + /* set SYSCLK / fs ratio */ + switch (uda134x->sysclk / params_rate(params)) { + case 512: + break; + case 384: + hw_params |= (1<<4); + break; + case 256: + hw_params |= (1<<5); + break; + default: + printk(KERN_ERR "%s unsupported fs\n", __func__); + return -EINVAL; + } + + pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__, + uda134x->dai_fmt, params_format(params)); + + /* set DAI format and word length */ + switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + hw_params |= (1<<1); + break; + case SNDRV_PCM_FORMAT_S18_3LE: + hw_params |= (1<<2); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + hw_params |= ((1<<2) | (1<<1)); + break; + default: + printk(KERN_ERR "%s unsupported format (right)\n", + __func__); + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + hw_params |= (1<<3); + break; + default: + printk(KERN_ERR "%s unsupported format\n", __func__); + return -EINVAL; + } + + uda134x_write(codec, UDA134X_STATUS0, hw_params); + + return 0; +} + +static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda134x_priv *uda134x = codec->private_data; + + pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__, + clk_id, freq, dir); + + /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable + because the codec is slave. Of course limitations of the clock + master (the IIS controller) apply. + We'll error out on set_hw_params if it's not OK */ + if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) { + uda134x->sysclk = freq; + return 0; + } + + printk(KERN_ERR "%s unsupported sysclk\n", __func__); + return -EINVAL; +} + +static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda134x_priv *uda134x = codec->private_data; + + pr_debug("%s fmt: %08X\n", __func__, fmt); + + /* codec supports only full slave mode */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + printk(KERN_ERR "%s unsupported slave mode\n", __func__); + return -EINVAL; + } + + /* no support for clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { + printk(KERN_ERR "%s unsupported clock inversion\n", __func__); + return -EINVAL; + } + + /* We can't setup DAI format here as it depends on the word bit num */ + /* so let's just store the value for later */ + uda134x->dai_fmt = fmt; + + return 0; +} + +static int uda134x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 reg; + struct uda134x_platform_data *pd = codec->control_data; + int i; + u8 *cache = codec->reg_cache; + + pr_debug("%s bias level %d\n", __func__, level); + + switch (level) { + case SND_SOC_BIAS_ON: + /* power on */ + if (pd->power) + pd->power(1); + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda134x_reg); i++) + codec->write(codec, i, *cache++); + /* ADC, DAC on */ + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + /* mute, ADC, DAC power off */ + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); + /* power off */ + if (pd->power) + pd->power(0); + break; + } + codec->bias_level = level; + return 0; +} + +static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1", + "Minimum2", "Maximum"}; +static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *uda134x_mixmode[] = {"Differential", "Analog1", + "Analog2", "Both"}; + +static const struct soc_enum uda134x_mixer_enum[] = { +SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting), +SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph), +SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode), +}; + +static const struct snd_kcontrol_new uda1341_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), +SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0), +SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1), +SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1), + +SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0), +SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0), + +SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), +SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), + +SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), +SOC_ENUM("Input Mux", uda134x_mixer_enum[2]), + +SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0), +SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1), +SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0), + +SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0), +SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0), +SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0), +SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0), +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + +static const struct snd_kcontrol_new uda1340_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), + +SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), +SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), + +SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), + +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + +static int uda134x_add_controls(struct snd_soc_codec *codec) +{ + int err, i, n; + const struct snd_kcontrol_new *ctrls; + struct uda134x_platform_data *pd = codec->control_data; + + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + n = ARRAY_SIZE(uda1340_snd_controls); + ctrls = uda1340_snd_controls; + break; + case UDA134X_UDA1341: + n = ARRAY_SIZE(uda1341_snd_controls); + ctrls = uda1341_snd_controls; + break; + default: + BUG(); + } + + for (i = 0; i < n; i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ctrls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +struct snd_soc_dai uda134x_dai = { + .name = "UDA134X", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA134X_RATES, + .formats = UDA134X_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA134X_RATES, + .formats = UDA134X_FORMATS, + }, + /* pcm operations */ + .ops = { + .startup = uda134x_startup, + .shutdown = uda134x_shutdown, + .hw_params = uda134x_hw_params, + }, + /* DAI operations */ + .dai_ops = { + .digital_mute = uda134x_mute, + .set_sysclk = uda134x_set_dai_sysclk, + .set_fmt = uda134x_set_dai_fmt, + } +}; +EXPORT_SYMBOL(uda134x_dai); + + +static int uda134x_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda134x_priv *uda134x; + void *codec_setup_data = socdev->codec_data; + int ret = -ENOMEM; + struct uda134x_platform_data *pd; + + printk(KERN_INFO "UDA134X SoC Audio Codec\n"); + + if (!codec_setup_data) { + printk(KERN_ERR "UDA134X SoC codec: " + "missing L3 bitbang function\n"); + return -ENODEV; + } + + pd = codec_setup_data; + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1341: + case UDA134X_UDA1344: + break; + default: + printk(KERN_ERR "UDA134X SoC codec: " + "unsupported model\n"); + return -EINVAL; + } + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return ret; + + codec = socdev->codec; + + uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL); + if (uda134x == NULL) + goto priv_err; + codec->private_data = uda134x; + + codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + goto reg_err; + + mutex_init(&codec->mutex); + + codec->reg_cache_size = sizeof(uda134x_reg); + codec->reg_cache_step = 1; + + codec->name = "UDA134X"; + codec->owner = THIS_MODULE; + codec->dai = &uda134x_dai; + codec->num_dai = 1; + codec->read = uda134x_read_reg_cache; + codec->write = uda134x_write; + codec->set_bias_level = uda134x_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->control_data = codec_setup_data; + + if (pd->power) + pd->power(1); + + uda134x_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register pcms\n"); + goto pcm_err; + } + + ret = uda134x_add_controls(codec); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register controls\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); +reg_err: + kfree(codec->private_data); +priv_err: + kfree(codec); + return ret; +} + +/* power down chip */ +static int uda134x_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} + +#if defined(CONFIG_PM) +static int uda134x_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int uda134x_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); + return 0; +} +#endif /* CONFIG_PM */ + +struct snd_soc_codec_device soc_codec_dev_uda134x = { + .probe = uda134x_soc_probe, + .remove = uda134x_soc_remove, +#if defined(CONFIG_PM) + .suspend = uda134x_soc_suspend, + .resume = uda134x_soc_resume, +#else + .suspend = NULL, + .resume = NULL, +#endif /* CONFIG_PM */ +}; +EXPORT_SYMBOL(soc_codec_dev_uda134x); + +MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); +MODULE_LICENSE("GPL"); -- 1.4.4.4

2 2

Re: [alsa-devel] Asus P5QL-EM motherboard (ALC1200 codec) - no sound through S/PDIF
by Andrei Tanas 14 Nov '08

14 Nov '08

> I cannot find any 'HDMI' words in your report below. > Can you provide some more info about it? Such as the full output of > 'aplay -l' and 'cat /proc/asound/card0/codec\#3'. Here's "aplay -L" with autodetection htpc@htpc:~$ aplay -L default:CARD=Intel HDA Intel, ALC1200 Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=Intel,DEV=0 HDA Intel HDMI Audio Output null Discard all samples (playback) or generate zero samples (capture) And here's with "options snd-hda-intel model=6stack-dig probe_mask=1" default:CARD=Intel HDA Intel, ALC1200 Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Intel,DEV=0 HDA Intel, ALC1200 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=Intel,DEV=0 HDA Intel HDMI Audio Output null Discard all samples (playback) or generate zero samples (capture) And here's "cat /proc/asound/card0/codec#0" Codec: Realtek ALC888 Address: 0 Vendor Id: 0x10ec0888 Subsystem Id: 0x104382fe Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Copyright Non-Audio GenLevel Digital category: 0x2 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x19 0x19] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x13 0x13] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x99430130: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000036: IN OUT Detect Trigger Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000036: IN OUT Detect Trigger Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19840: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19c50: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181304f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4015e601: [N/A] Speaker at Ext N/A Conn = Optical, Color = White DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b

3 4

[alsa-devel] [PATCH 0/4] ASoC updates
by Mark Brown 14 Nov '08

14 Nov '08

The following changes since commit fb0ef645f2c546f8297b2fbf9b2b8fff4a7455e8: Naresh Medisetty (1): ASoC: DaVinci: Audio: Fix swapping of channels at start of stereo playback are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Jarkko Nikula (2): ASoC: Fix supported sample rates of TWL4030 audio codec ASoC: OMAP: Add more supported sample rates into McBSP DAI driver Mark Brown (2): ASoC: Revert "ASoC: Add new parameter to s3c24xx_pcm_enqueue" ASoC: Add WM8728 codec driver sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/wm8728.c | 574 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8728.h | 30 ++ sound/soc/omap/omap-mcbsp.c | 4 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 12 +- 7 files changed, 616 insertions(+), 12 deletions(-) create mode 100644 sound/soc/codecs/wm8728.c create mode 100644 sound/soc/codecs/wm8728.h

2 5

[alsa-devel] [PATCH] ASoC: Fix for master playback/capture volume range for TWL4030 codec
by Peter Ujfalusi 14 Nov '08

14 Nov '08

FGAIN for playback is in range of 0-0x3f, while for capture GAIN it is in the range of 0-0x1f. The original value of 128 (0x7f) would modify the CGAIN also for playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 90f3b4d..65aeabe 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -192,10 +192,10 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 0, 127, 0), + 0, 0x3f, 0), SOC_DOUBLE_R("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, - 0, 127, 0), + 0, 0x1f, 0), }; /* add non dapm controls */ -- 1.6.0.3

3 2

[alsa-devel] Regression on alsa-driver-1.0.18a on HDA for Dell Inspiron 1521
by Manuel Naranjo 14 Nov '08

14 Nov '08

Hello guys, I just upgraded my alsa from 1.0.16 to 1.0.18a (actually I tried 1.0.18 for some days, and got lots of kernel crashes, which apparently 1.0.18a fixes). Thing is that when I try to use my headphones with 1.0.18a they don't work. I'm attaching alsa-info.sh outputs for both 1.0.16 and 1.0.18a too see if this helps. Headphones detection works pretty well, when ever I plug the jack internal speakers stop working but nothing is coming from the jack. I can hear an static sounds, so it seems like the headphones are getting the bias current, but not the signal itself. Please let me know how can I help you to debug this problem. Regards, Manuel

4 16

[alsa-devel] Why no Marco definitions for rate 24000 and 12000?
by cocala 14 Nov '08

14 Nov '08

Who can tell me why not define add SNDRV_PCM_RATE_24000 and SNDRV_PCM_RATE_12000?

3 4

[alsa-devel] snd_pcm_ops
by John Kacur 14 Nov '08

14 Nov '08

As a newcomer to alsa, I have a few more probably simple questions. I noticed in the SOC drivers, that when the struct snd_pcm_ops is filled in, .copy is omitted, for example in file s3c24xx-pcm.c, you see the following initialization. static struct snd_pcm_ops s3c24xx_pcm_ops = { .open = s3c24xx_pcm_open, .close = s3c24xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = s3c24xx_pcm_hw_params, .hw_free = s3c24xx_pcm_hw_free, .prepare = s3c24xx_pcm_prepare, .trigger = s3c24xx_pcm_trigger, .pointer = s3c24xx_pcm_pointer, .mmap = s3c24xx_pcm_mmap, }; How is this supposed to work if somewhere down the line substream->ops->copy is called? Thanks in advance for any pointers.

2 1

Re: [alsa-devel] Roland/Edirol M-16DX
by Clemens Ladisch 14 Nov '08

14 Nov '08

James Trevelyan wrote: > ... > However, what I did notice is that in all circumstances the windows > driver was capturing at the same time as playback, even when I was not > asking it to record. This suggests to me that the comment in the driver > source about synchronising playback to capture has some relevance ... Indeed. The driver would have to send the data at the exact speed of the device's internal clock, and the only way to determine that clock's speed is to capture data. So far I haven't found the time to rewrite the driver to support this synchronization mechanism. Best regards, Clemens

1 0

[alsa-devel] [RFC][PATCH] ELD routines and proc interface
by Wu Fengguang 14 Nov '08

14 Nov '08

Create hda_eld.c for ELD routines and proc interface. ELD handling routines can be shared by all HDMI codecs, and they are large enough to make a standalone source file. Some notes on the code: - Shall we show an empty file if the ELD content is not valid? Well it's not that simple. There could be partially populated ELD, and there may be malformed ELD provided by buggy drivers/monitors. So I tend to expose ELD as it is. - hdmi_show_eld() and hdmi_print_eld_info() are somehow duplicated The plan is to shrink hdmi_show_eld()/hdmi_show_short_audio_desc(), and to show a compact message at monitor hotplug and driver initialization time. - The ELD retrieval routines rely on the Intel HDA interface, others are/could be universal and independent ones. - The ELD proc interface code could be moved to hda_proc.c if necessary. - How do we name the proc file? If there are going to be two HDMI pins per codec, then the current naming scheme (eld#<codec no>) will fail. - The ELD proc file content is designed to be easy for scripts and human reading. Its lines all have the pattern: <item_name>\t[\t]*<item_value> where <item_name> is a keyword in c language, while <item_value> could be any contents, including white spaces. <item_value> could also be a null value. Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/Kconfig | 4 sound/pci/hda/Makefile | 1 sound/pci/hda/hda_eld.c | 532 ++++++++++++++++++++++++++++++ sound/pci/hda/hda_local.h | 62 ++- sound/pci/hda/patch_intelhdmi.c | 484 --------------------------- 5 files changed, 600 insertions(+), 483 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_intelhdmi.c +++ sound-2.6/sound/pci/hda/patch_intelhdmi.c @@ -30,7 +30,6 @@ #include <linux/delay.h> #include <linux/slab.h> #include <sound/core.h> -#include <asm/unaligned.h> #include "hda_codec.h" #include "hda_local.h" #include "hda_patch.h" @@ -40,43 +39,6 @@ #define INTEL_HDMI_EVENT_TAG 0x08 -/* - * CEA Short Audio Descriptor data - */ -struct cea_sad { - int channels; - int format; /* (format == 0) indicates invalid SAD */ - int rates; - int sample_bits; /* for LPCM */ - int max_bitrate; /* for AC3...ATRAC */ - int profile; /* for WMAPRO */ -}; - -#define ELD_FIXED_BYTES 20 -#define ELD_MAX_MNL 16 -#define ELD_MAX_SAD 16 - -/* - * ELD: EDID Like Data - */ -struct sink_eld { - int eld_size; - int baseline_len; - int eld_ver; /* (eld_ver == 0) indicates invalid ELD */ - int cea_edid_ver; - char monitor_name[ELD_MAX_MNL + 1]; - int manufacture_id; - int product_id; - u64 port_id; - int support_hdcp; - int support_ai; - int conn_type; - int aud_synch_delay; - int spk_alloc; - int sad_count; - struct cea_sad sad[ELD_MAX_SAD]; -}; - struct intel_hdmi_spec { struct hda_multi_out multiout; struct hda_pcm pcm_rec; @@ -127,160 +89,9 @@ struct hdmi_audio_infoframe { }; /* - * SS1:SS0 index => sample size - */ -static int cea_sample_sizes[4] = { - 0, /* 0: Refer to Stream Header */ - AC_SUPPCM_BITS_16, /* 1: 16 bits */ - AC_SUPPCM_BITS_20, /* 2: 20 bits */ - AC_SUPPCM_BITS_24, /* 3: 24 bits */ -}; - -/* - * SF2:SF1:SF0 index => sampling frequency - */ -static int cea_sampling_frequencies[8] = { - 0, /* 0: Refer to Stream Header */ - SNDRV_PCM_RATE_32000, /* 1: 32000Hz */ - SNDRV_PCM_RATE_44100, /* 2: 44100Hz */ - SNDRV_PCM_RATE_48000, /* 3: 48000Hz */ - SNDRV_PCM_RATE_88200, /* 4: 88200Hz */ - SNDRV_PCM_RATE_96000, /* 5: 96000Hz */ - SNDRV_PCM_RATE_176400, /* 6: 176400Hz */ - SNDRV_PCM_RATE_192000, /* 7: 192000Hz */ -}; - -enum eld_versions { - ELD_VER_CEA_861D = 2, - ELD_VER_PARTIAL = 31, -}; - -static char *eld_versoin_names[32] = { - "0-reserved", - "1-reserved", - "CEA-861D or below", - "3-reserved", - [4 ... 30] = "reserved", - [31] = "partial" -}; - -enum cea_edid_versions { - CEA_EDID_VER_NONE = 0, - CEA_EDID_VER_CEA861 = 1, - CEA_EDID_VER_CEA861A = 2, - CEA_EDID_VER_CEA861BCD = 3, - CEA_EDID_VER_RESERVED = 4, -}; - -static char *cea_edid_version_names[8] = { - "no CEA EDID Timing Extension block present", - "CEA-861", - "CEA-861-A", - "CEA-861-B, C or D", - "4-reserved", - [5 ... 7] = "reserved" -}; - -/* - * CEA Speaker Allocation data block bits - */ -#define CEA_SA_FLR (0 << 0) -#define CEA_SA_LFE (1 << 1) -#define CEA_SA_FC (1 << 2) -#define CEA_SA_RLR (1 << 3) -#define CEA_SA_RC (1 << 4) -#define CEA_SA_FLRC (1 << 5) -#define CEA_SA_RLRC (1 << 6) -/* the following are not defined in ELD yet */ -#define CEA_SA_FLRW (1 << 7) -#define CEA_SA_FLRH (1 << 8) -#define CEA_SA_TC (1 << 9) -#define CEA_SA_FCH (1 << 10) - -static char *cea_speaker_allocation_names[] = { - /* 0 */ "FL/FR", - /* 1 */ "LFE", - /* 2 */ "FC", - /* 3 */ "RL/RR", - /* 4 */ "RC", - /* 5 */ "FLC/FRC", - /* 6 */ "RLC/RRC", - /* 7 */ "FLW/FRW", - /* 8 */ "FLH/FRH", - /* 9 */ "TC", - /* 10 */ "FCH", -}; - -static char *eld_connection_type_names[4] = { - "HDMI", - "Display Port", - "2-reserved", - "3-reserved" -}; - -enum cea_audio_coding_types { - AUDIO_CODING_TYPE_REF_STREAM_HEADER = 0, - AUDIO_CODING_TYPE_LPCM = 1, - AUDIO_CODING_TYPE_AC3 = 2, - AUDIO_CODING_TYPE_MPEG1 = 3, - AUDIO_CODING_TYPE_MP3 = 4, - AUDIO_CODING_TYPE_MPEG2 = 5, - AUDIO_CODING_TYPE_AACLC = 6, - AUDIO_CODING_TYPE_DTS = 7, - AUDIO_CODING_TYPE_ATRAC = 8, - AUDIO_CODING_TYPE_SACD = 9, - AUDIO_CODING_TYPE_EAC3 = 10, - AUDIO_CODING_TYPE_DTS_HD = 11, - AUDIO_CODING_TYPE_MLP = 12, - AUDIO_CODING_TYPE_DST = 13, - AUDIO_CODING_TYPE_WMAPRO = 14, - AUDIO_CODING_TYPE_REF_CXT = 15, - /* also include valid xtypes below */ - AUDIO_CODING_TYPE_HE_AAC = 15, - AUDIO_CODING_TYPE_HE_AAC2 = 16, - AUDIO_CODING_TYPE_MPEG_SURROUND = 17, -}; - -enum cea_audio_coding_xtypes { - AUDIO_CODING_XTYPE_HE_REF_CT = 0, - AUDIO_CODING_XTYPE_HE_AAC = 1, - AUDIO_CODING_XTYPE_HE_AAC2 = 2, - AUDIO_CODING_XTYPE_MPEG_SURROUND = 3, - AUDIO_CODING_XTYPE_FIRST_RESERVED = 4, -}; - -static char *cea_audio_coding_type_names[] = { - /* 0 */ "undefined", - /* 1 */ "LPCM", - /* 2 */ "AC-3", - /* 3 */ "MPEG1", - /* 4 */ "MP3", - /* 5 */ "MPEG2", - /* 6 */ "AAC-LC", - /* 7 */ "DTS", - /* 8 */ "ATRAC", - /* 9 */ "DSD(1-bit audio)", - /* 10 */ "Dolby Digital Plus(E-AC-3/DD+)", - /* 11 */ "DTS-HD", - /* 12 */ "Dolby TrueHD(MLP)", - /* 13 */ "DST", - /* 14 */ "WMAPro", - /* 15 */ "HE-AAC", - /* 16 */ "HE-AACv2", - /* 17 */ "MPEG Surround", -}; - - -/* * HDMI routines */ -static int hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid) -{ - return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, - AC_DIPSIZE_ELD_BUF); -} - #ifdef BE_PARANOID static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t nid, int *packet_index, int *byte_index) @@ -375,294 +186,13 @@ static void hdmi_setup_channel_mapping(s } -/* - * ELD(EDID Like Data) routines - */ - -static int hdmi_present_sense(struct hda_codec *codec, hda_nid_t nid) -{ - return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); -} - -static void hdmi_debug_present_sense(struct hda_codec *codec) -{ -#ifdef CONFIG_SND_DEBUG_VERBOSE - int eldv; - int present; - - present = hdmi_present_sense(codec, PIN_NID); - eldv = (present & AC_PINSENSE_ELDV); - present = (present & AC_PINSENSE_PRESENCE); - - printk(KERN_INFO "pinp = %d, eldv = %d\n", !!present, !!eldv); -#endif -} - -static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, int byte_index) -{ - unsigned int val; - - val = snd_hda_codec_read(codec, PIN_NID, 0, - AC_VERB_GET_HDMI_ELDD, byte_index); - -#ifdef BE_PARANOID - printk(KERN_INFO "ELD data byte %d: 0x%x\n", byte_index, val); -#endif - - if ((val & AC_ELDD_ELD_VALID) == 0) { - snd_printd(KERN_INFO "Invalid ELD data byte %d\n", - byte_index); - val = 0; - } - - return val & AC_ELDD_ELD_DATA; -} - -static inline unsigned char grab_bits(const unsigned char *buf, - int byte, int lowbit, int bits) -{ - BUG_ON(lowbit > 7); - BUG_ON(bits > 8); - BUG_ON(bits <= 0); - - return (buf[byte] >> lowbit) & ((1 << bits) - 1); -} - -static void hdmi_update_short_audio_desc(struct cea_sad *a, - const unsigned char *buf) -{ - int i; - int val; - - val = grab_bits(buf, 1, 0, 7); - a->rates = 0; - for (i = 0; i < 7; i++) - if (val & (1 << i)) - a->rates |= cea_sampling_frequencies[i + 1]; - - a->channels = grab_bits(buf, 0, 0, 3); - a->channels++; - - a->format = grab_bits(buf, 0, 3, 4); - switch (a->format) { - case AUDIO_CODING_TYPE_REF_STREAM_HEADER: - snd_printd(KERN_INFO - "audio coding type 0 not expected in ELD\n"); - break; - - case AUDIO_CODING_TYPE_LPCM: - val = grab_bits(buf, 2, 0, 3); - a->sample_bits = 0; - for (i = 0; i < 3; i++) - if (val & (1 << i)) - a->sample_bits |= cea_sample_sizes[i + 1]; - break; - - case AUDIO_CODING_TYPE_AC3: - case AUDIO_CODING_TYPE_MPEG1: - case AUDIO_CODING_TYPE_MP3: - case AUDIO_CODING_TYPE_MPEG2: - case AUDIO_CODING_TYPE_AACLC: - case AUDIO_CODING_TYPE_DTS: - case AUDIO_CODING_TYPE_ATRAC: - a->max_bitrate = grab_bits(buf, 2, 0, 8); - a->max_bitrate *= 8000; - break; - - case AUDIO_CODING_TYPE_SACD: - break; - - case AUDIO_CODING_TYPE_EAC3: - break; - - case AUDIO_CODING_TYPE_DTS_HD: - break; - - case AUDIO_CODING_TYPE_MLP: - break; - - case AUDIO_CODING_TYPE_DST: - break; - - case AUDIO_CODING_TYPE_WMAPRO: - a->profile = grab_bits(buf, 2, 0, 3); - break; - - case AUDIO_CODING_TYPE_REF_CXT: - a->format = grab_bits(buf, 2, 3, 5); - if (a->format == AUDIO_CODING_XTYPE_HE_REF_CT || - a->format >= AUDIO_CODING_XTYPE_FIRST_RESERVED) { - snd_printd(KERN_INFO - "audio coding xtype %d not expected in ELD\n", - a->format); - a->format = 0; - } else - a->format += AUDIO_CODING_TYPE_HE_AAC - - AUDIO_CODING_XTYPE_HE_AAC; - break; - } -} - -static int hdmi_update_sink_eld(struct hda_codec *codec, - const unsigned char *buf, int size) -{ - struct intel_hdmi_spec *spec = codec->spec; - struct sink_eld *e = &spec->sink; - int mnl; - int i; - - e->eld_ver = grab_bits(buf, 0, 3, 5); - if (e->eld_ver != ELD_VER_CEA_861D && - e->eld_ver != ELD_VER_PARTIAL) { - snd_printd(KERN_INFO "Unknown ELD version %d\n", e->eld_ver); - goto out_fail; - } - - e->eld_size = size; - e->baseline_len = grab_bits(buf, 2, 0, 8); - mnl = grab_bits(buf, 4, 0, 5); - e->cea_edid_ver = grab_bits(buf, 4, 5, 3); - - e->support_hdcp = grab_bits(buf, 5, 0, 1); - e->support_ai = grab_bits(buf, 5, 1, 1); - e->conn_type = grab_bits(buf, 5, 2, 2); - e->sad_count = grab_bits(buf, 5, 4, 4); - - e->aud_synch_delay = grab_bits(buf, 6, 0, 8); - e->spk_alloc = grab_bits(buf, 7, 0, 7); - - e->port_id = get_unaligned_le64(buf + 8); - - /* not specified, but the spec's tendency is little endian */ - e->manufacture_id = get_unaligned_le16(buf + 16); - e->product_id = get_unaligned_le16(buf + 18); - - if (mnl > ELD_MAX_MNL) { - snd_printd(KERN_INFO "MNL is reserved value %d\n", mnl); - goto out_fail; - } else if (ELD_FIXED_BYTES + mnl > size) { - snd_printd(KERN_INFO "out of range MNL %d\n", mnl); - goto out_fail; - } else - strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl); - - for (i = 0; i < e->sad_count; i++) { - if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) { - snd_printd(KERN_INFO "out of range SAD %d\n", i); - goto out_fail; - } - hdmi_update_short_audio_desc(e->sad + i, - buf + ELD_FIXED_BYTES + mnl + 3 * i); - } - - return 0; - -out_fail: - e->eld_ver = 0; - return -EINVAL; -} - -static int hdmi_get_eld(struct hda_codec *codec) -{ - int i; - int ret; - int size; - unsigned char *buf; - - i = hdmi_present_sense(codec, PIN_NID) & AC_PINSENSE_ELDV; - if (!i) - return -ENOENT; - - size = hdmi_get_eld_size(codec, PIN_NID); - if (size == 0) { - /* wfg: workaround for ASUS P5E-VM HDMI board */ - snd_printd(KERN_INFO "ELD buf size is 0, force 128\n"); - size = 128; - } - if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { - snd_printd(KERN_INFO "Invalid ELD buf size %d\n", size); - return -ERANGE; - } - - buf = kmalloc(size, GFP_KERNEL); - if (!buf) - return -ENOMEM; - - for (i = 0; i < size; i++) - buf[i] = hdmi_get_eld_byte(codec, i); - - ret = hdmi_update_sink_eld(codec, buf, size); - - kfree(buf); - return ret; -} - -static void hdmi_show_short_audio_desc(struct cea_sad *a) -{ - printk(KERN_INFO "coding type: %s\n", - cea_audio_coding_type_names[a->format]); - printk(KERN_INFO "channels: %d\n", a->channels); - printk(KERN_INFO "sampling frequencies: 0x%x\n", a->rates); - - if (a->format == AUDIO_CODING_TYPE_LPCM) - printk(KERN_INFO "sample bits: 0x%x\n", a->sample_bits); - - if (a->max_bitrate) - printk(KERN_INFO "max bitrate: %d HZ\n", a->max_bitrate); - - if (a->profile) - printk(KERN_INFO "profile: %d\n", a->profile); -} - -static void hdmi_show_eld(struct hda_codec *codec) -{ - int i; - int j; - struct intel_hdmi_spec *spec = codec->spec; - struct sink_eld *e = &spec->sink; - char buf[80]; - - printk(KERN_INFO "ELD buffer size is %d\n", e->eld_size); - printk(KERN_INFO "ELD baseline len is %d*4\n", e->baseline_len); - printk(KERN_INFO "vendor block len is %d\n", - e->eld_size - e->baseline_len * 4 - 4); - printk(KERN_INFO "ELD version is %s\n", - eld_versoin_names[e->eld_ver]); - printk(KERN_INFO "CEA EDID version is %s\n", - cea_edid_version_names[e->cea_edid_ver]); - printk(KERN_INFO "manufacture id is 0x%x\n", e->manufacture_id); - printk(KERN_INFO "product id is 0x%x\n", e->product_id); - printk(KERN_INFO "port id is 0x%llx\n", (long long)e->port_id); - printk(KERN_INFO "HDCP support is %d\n", e->support_hdcp); - printk(KERN_INFO "AI support is %d\n", e->support_ai); - printk(KERN_INFO "SAD count is %d\n", e->sad_count); - printk(KERN_INFO "audio sync delay is %x\n", e->aud_synch_delay); - printk(KERN_INFO "connection type is %s\n", - eld_connection_type_names[e->conn_type]); - printk(KERN_INFO "monitor name is %s\n", e->monitor_name); - - j = 0; - for (i = 0; i < ARRAY_SIZE(cea_speaker_allocation_names); i++) { - if (e->spk_alloc & (1 << i)) - j += snprintf(buf + j, sizeof(buf) - j, " %s", - cea_speaker_allocation_names[i]); - } - buf[j] = '\0'; /* necessary when j == 0 */ - printk(KERN_INFO "speaker allocations: (0x%x)%s\n", e->spk_alloc, buf); - - for (i = 0; i < e->sad_count; i++) - hdmi_show_short_audio_desc(e->sad + i); -} - -/* - * Be careful, ELD buf could be totally rubbish! - */ static void hdmi_parse_eld(struct hda_codec *codec) { - hdmi_debug_present_sense(codec); + struct intel_hdmi_spec *spec = codec->spec; + struct sink_eld *eld = &spec->sink; - if (!hdmi_get_eld(codec)) - hdmi_show_eld(codec); + if (!hdmi_get_eld(eld, codec, PIN_NID)) + hdmi_show_eld(eld); } @@ -880,6 +410,11 @@ static int intel_hdmi_build_controls(str static int intel_hdmi_init(struct hda_codec *codec) { + struct intel_hdmi_spec *spec = codec->spec; + struct sink_eld *eld = &spec->sink; + + snd_hda_eld_proc_new(codec, eld); + /* disable audio output as early as possible */ hdmi_disable_output(codec); @@ -927,3 +462,4 @@ struct hda_codec_preset snd_hda_preset_i { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ }; + --- /dev/null +++ sound-2.6/sound/pci/hda/hda_eld.c @@ -0,0 +1,532 @@ +/* + * Generic routines and proc interface for ELD(EDID Like Data) information + * + * Copyright(c) 2008 Intel Corporation. + * + * Authors: + * Wu Fengguang <wfg(a)linux.intel.com> + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <sound/core.h> +#include <asm/unaligned.h> +#include "hda_codec.h" +#include "hda_local.h" + +enum eld_versions { + ELD_VER_CEA_861D = 2, + ELD_VER_PARTIAL = 31, +}; + +static char *eld_versoin_names[32] = { + "reserved", + "reserved", + "CEA-861D or below", + [3 ... 30] = "reserved", + [31] = "partial" +}; + +enum cea_edid_versions { + CEA_EDID_VER_NONE = 0, + CEA_EDID_VER_CEA861 = 1, + CEA_EDID_VER_CEA861A = 2, + CEA_EDID_VER_CEA861BCD = 3, + CEA_EDID_VER_RESERVED = 4, +}; + +static char *cea_edid_version_names[8] = { + "no CEA EDID Timing Extension block present", + "CEA-861", + "CEA-861-A", + "CEA-861-B, C or D", + [4 ... 7] = "reserved" +}; + +/* + * CEA Speaker Allocation data block bits + */ +#define CEA_SA_FLR (0 << 0) +#define CEA_SA_LFE (1 << 1) +#define CEA_SA_FC (1 << 2) +#define CEA_SA_RLR (1 << 3) +#define CEA_SA_RC (1 << 4) +#define CEA_SA_FLRC (1 << 5) +#define CEA_SA_RLRC (1 << 6) +/* the following are not defined in ELD yet */ +#define CEA_SA_FLRW (1 << 7) +#define CEA_SA_FLRH (1 << 8) +#define CEA_SA_TC (1 << 9) +#define CEA_SA_FCH (1 << 10) + +static char *cea_speaker_allocation_names[] = { + /* 0 */ "FL/FR", + /* 1 */ "LFE", + /* 2 */ "FC", + /* 3 */ "RL/RR", + /* 4 */ "RC", + /* 5 */ "FLC/FRC", + /* 6 */ "RLC/RRC", + /* 7 */ "FLW/FRW", + /* 8 */ "FLH/FRH", + /* 9 */ "TC", + /* 10 */ "FCH", +}; + +static char *eld_connection_type_names[4] = { + "HDMI", + "Display Port", + "2-reserved", + "3-reserved" +}; + +enum cea_audio_coding_types { + AUDIO_CODING_TYPE_REF_STREAM_HEADER = 0, + AUDIO_CODING_TYPE_LPCM = 1, + AUDIO_CODING_TYPE_AC3 = 2, + AUDIO_CODING_TYPE_MPEG1 = 3, + AUDIO_CODING_TYPE_MP3 = 4, + AUDIO_CODING_TYPE_MPEG2 = 5, + AUDIO_CODING_TYPE_AACLC = 6, + AUDIO_CODING_TYPE_DTS = 7, + AUDIO_CODING_TYPE_ATRAC = 8, + AUDIO_CODING_TYPE_SACD = 9, + AUDIO_CODING_TYPE_EAC3 = 10, + AUDIO_CODING_TYPE_DTS_HD = 11, + AUDIO_CODING_TYPE_MLP = 12, + AUDIO_CODING_TYPE_DST = 13, + AUDIO_CODING_TYPE_WMAPRO = 14, + AUDIO_CODING_TYPE_REF_CXT = 15, + /* also include valid xtypes below */ + AUDIO_CODING_TYPE_HE_AAC = 15, + AUDIO_CODING_TYPE_HE_AAC2 = 16, + AUDIO_CODING_TYPE_MPEG_SURROUND = 17, +}; + +enum cea_audio_coding_xtypes { + AUDIO_CODING_XTYPE_HE_REF_CT = 0, + AUDIO_CODING_XTYPE_HE_AAC = 1, + AUDIO_CODING_XTYPE_HE_AAC2 = 2, + AUDIO_CODING_XTYPE_MPEG_SURROUND = 3, + AUDIO_CODING_XTYPE_FIRST_RESERVED = 4, +}; + +static char *cea_audio_coding_type_names[] = { + /* 0 */ "undefined", + /* 1 */ "LPCM", + /* 2 */ "AC-3", + /* 3 */ "MPEG1", + /* 4 */ "MP3", + /* 5 */ "MPEG2", + /* 6 */ "AAC-LC", + /* 7 */ "DTS", + /* 8 */ "ATRAC", + /* 9 */ "DSD(1-bit audio)", + /* 10 */ "Dolby Digital Plus(E-AC-3/DD+)", + /* 11 */ "DTS-HD", + /* 12 */ "Dolby TrueHD(MLP)", + /* 13 */ "DST", + /* 14 */ "WMAPro", + /* 15 */ "HE-AAC", + /* 16 */ "HE-AACv2", + /* 17 */ "MPEG Surround", +}; + +/* + * The following two lists are shared between + * - HDMI audio infoframe (source to sink) + * - CEA E-EDID extension (sink to source) + */ + +/* + * SS1:SS0 index => sample size + */ +static int cea_sample_sizes[4] = { + 0, /* 0: Refer to Stream Header */ + AC_SUPPCM_BITS_16, /* 1: 16 bits */ + AC_SUPPCM_BITS_20, /* 2: 20 bits */ + AC_SUPPCM_BITS_24, /* 3: 24 bits */ +}; + +/* + * SF2:SF1:SF0 index => sampling frequency + */ +static int cea_sampling_frequencies[8] = { + 0, /* 0: Refer to Stream Header */ + SNDRV_PCM_RATE_32000, /* 1: 32000Hz */ + SNDRV_PCM_RATE_44100, /* 2: 44100Hz */ + SNDRV_PCM_RATE_48000, /* 3: 48000Hz */ + SNDRV_PCM_RATE_88200, /* 4: 88200Hz */ + SNDRV_PCM_RATE_96000, /* 5: 96000Hz */ + SNDRV_PCM_RATE_176400, /* 6: 176400Hz */ + SNDRV_PCM_RATE_192000, /* 7: 192000Hz */ +}; + +static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid, + int byte_index) +{ + unsigned int val; + + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_HDMI_ELDD, byte_index); + +#ifdef BE_PARANOID + printk(KERN_INFO "ELD data byte %d: 0x%x\n", byte_index, val); +#endif + + if ((val & AC_ELDD_ELD_VALID) == 0) { + snd_printd(KERN_INFO "Invalid ELD data byte %d\n", + byte_index); + val = 0; + } + + return val & AC_ELDD_ELD_DATA; +} + +static inline unsigned char grab_bits(const unsigned char *buf, + int byte, int lowbit, int bits) +{ + BUG_ON(lowbit > 7); + BUG_ON(bits > 8); + BUG_ON(bits <= 0); + + return (buf[byte] >> lowbit) & ((1 << bits) - 1); +} + +static void hdmi_update_short_audio_desc(struct cea_sad *a, + const unsigned char *buf) +{ + int i; + int val; + + val = grab_bits(buf, 1, 0, 7); + a->rates = 0; + for (i = 0; i < 7; i++) + if (val & (1 << i)) + a->rates |= cea_sampling_frequencies[i + 1]; + + a->channels = grab_bits(buf, 0, 0, 3); + a->channels++; + + a->format = grab_bits(buf, 0, 3, 4); + switch (a->format) { + case AUDIO_CODING_TYPE_REF_STREAM_HEADER: + snd_printd(KERN_INFO + "audio coding type 0 not expected in ELD\n"); + break; + + case AUDIO_CODING_TYPE_LPCM: + val = grab_bits(buf, 2, 0, 3); + a->sample_bits = 0; + for (i = 0; i < 3; i++) + if (val & (1 << i)) + a->sample_bits |= cea_sample_sizes[i + 1]; + break; + + case AUDIO_CODING_TYPE_AC3: + case AUDIO_CODING_TYPE_MPEG1: + case AUDIO_CODING_TYPE_MP3: + case AUDIO_CODING_TYPE_MPEG2: + case AUDIO_CODING_TYPE_AACLC: + case AUDIO_CODING_TYPE_DTS: + case AUDIO_CODING_TYPE_ATRAC: + a->max_bitrate = grab_bits(buf, 2, 0, 8); + a->max_bitrate *= 8000; + break; + + case AUDIO_CODING_TYPE_SACD: + break; + + case AUDIO_CODING_TYPE_EAC3: + break; + + case AUDIO_CODING_TYPE_DTS_HD: + break; + + case AUDIO_CODING_TYPE_MLP: + break; + + case AUDIO_CODING_TYPE_DST: + break; + + case AUDIO_CODING_TYPE_WMAPRO: + a->profile = grab_bits(buf, 2, 0, 3); + break; + + case AUDIO_CODING_TYPE_REF_CXT: + a->format = grab_bits(buf, 2, 3, 5); + if (a->format == AUDIO_CODING_XTYPE_HE_REF_CT || + a->format >= AUDIO_CODING_XTYPE_FIRST_RESERVED) { + snd_printd(KERN_INFO + "audio coding xtype %d not expected in ELD\n", + a->format); + a->format = 0; + } else + a->format += AUDIO_CODING_TYPE_HE_AAC - + AUDIO_CODING_XTYPE_HE_AAC; + break; + } +} + +/* + * Be careful, ELD buf could be totally rubbish! + */ +static int hdmi_update_sink_eld(struct sink_eld *e, + const unsigned char *buf, int size) +{ + int mnl; + int i; + + e->eld_ver = grab_bits(buf, 0, 3, 5); + if (e->eld_ver != ELD_VER_CEA_861D && + e->eld_ver != ELD_VER_PARTIAL) { + snd_printd(KERN_INFO "Unknown ELD version %d\n", e->eld_ver); + goto out_fail; + } + + e->eld_size = size; + e->baseline_len = grab_bits(buf, 2, 0, 8); + mnl = grab_bits(buf, 4, 0, 5); + e->cea_edid_ver = grab_bits(buf, 4, 5, 3); + + e->support_hdcp = grab_bits(buf, 5, 0, 1); + e->support_ai = grab_bits(buf, 5, 1, 1); + e->conn_type = grab_bits(buf, 5, 2, 2); + e->sad_count = grab_bits(buf, 5, 4, 4); + + e->aud_synch_delay = grab_bits(buf, 6, 0, 8) * 2; + e->spk_alloc = grab_bits(buf, 7, 0, 7); + + e->port_id = get_unaligned_le64(buf + 8); + + /* not specified, but the spec's tendency is little endian */ + e->manufacture_id = get_unaligned_le16(buf + 16); + e->product_id = get_unaligned_le16(buf + 18); + + if (mnl > ELD_MAX_MNL) { + snd_printd(KERN_INFO "MNL is reserved value %d\n", mnl); + goto out_fail; + } else if (ELD_FIXED_BYTES + mnl > size) { + snd_printd(KERN_INFO "out of range MNL %d\n", mnl); + goto out_fail; + } else + strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl); + + for (i = 0; i < e->sad_count; i++) { + if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) { + snd_printd(KERN_INFO "out of range SAD %d\n", i); + goto out_fail; + } + hdmi_update_short_audio_desc(e->sad + i, + buf + ELD_FIXED_BYTES + mnl + 3 * i); + } + + return 0; + +out_fail: + e->eld_ver = 0; + return -EINVAL; +} + +static int hdmi_present_sense(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); +} + +static int hdmi_eld_valid(struct hda_codec *codec, hda_nid_t nid) +{ + int eldv; + int present; + + present = hdmi_present_sense(codec, nid); + eldv = (present & AC_PINSENSE_ELDV); + present = (present & AC_PINSENSE_PRESENCE); + +#ifdef CONFIG_SND_DEBUG_VERBOSE + printk(KERN_INFO "pinp = %d, eldv = %d\n", !!present, !!eldv); +#endif + + return eldv && present; +} + +int hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, + AC_DIPSIZE_ELD_BUF); +} + +int hdmi_get_eld(struct sink_eld *eld, + struct hda_codec *codec, hda_nid_t nid) +{ + int i; + int ret; + int size; + unsigned char *buf; + + if (!hdmi_eld_valid(codec, nid)) + return -ENOENT; + + size = hdmi_get_eld_size(codec, nid); + if (size == 0) { + /* wfg: workaround for ASUS P5E-VM HDMI board */ + snd_printd(KERN_INFO "ELD buf size is 0, force 128\n"); + size = 128; + } + if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { + snd_printd(KERN_INFO "Invalid ELD buf size %d\n", size); + return -ERANGE; + } + + buf = kmalloc(size, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + for (i = 0; i < size; i++) + buf[i] = hdmi_get_eld_byte(codec, nid, i); + + ret = hdmi_update_sink_eld(eld, buf, size); + + kfree(buf); + return ret; +} + +static void hdmi_show_short_audio_desc(struct cea_sad *a) +{ + printk(KERN_INFO "coding type: %s\n", + cea_audio_coding_type_names[a->format]); + printk(KERN_INFO "channels: %d\n", a->channels); + printk(KERN_INFO "sampling frequencies: 0x%x\n", a->rates); + + if (a->format == AUDIO_CODING_TYPE_LPCM) + printk(KERN_INFO "sample bits: 0x%x\n", a->sample_bits); + + if (a->max_bitrate) + printk(KERN_INFO "max bitrate: %d HZ\n", a->max_bitrate); + + if (a->profile) + printk(KERN_INFO "profile: %d\n", a->profile); +} + +static void hdmi_print_speaker_allocations(int spk_alloc, char *buf, int buflen) +{ + int i, j; + + for (i = 0, j = 0; i < ARRAY_SIZE(cea_speaker_allocation_names); i++) { + if (spk_alloc & (1 << i)) + j += snprintf(buf + j, buflen - j, " %s", + cea_speaker_allocation_names[i]); + } + buf[j] = '\0'; /* necessary when j == 0 */ +} + +void hdmi_show_eld(struct sink_eld *e) +{ + int i; + char buf[80]; + + printk(KERN_INFO "ELD buffer size is %d\n", e->eld_size); + printk(KERN_INFO "ELD baseline len is %d*4\n", e->baseline_len); + printk(KERN_INFO "vendor block len is %d\n", + e->eld_size - e->baseline_len * 4 - 4); + printk(KERN_INFO "ELD version is %s\n", + eld_versoin_names[e->eld_ver]); + printk(KERN_INFO "CEA EDID version is %s\n", + cea_edid_version_names[e->cea_edid_ver]); + printk(KERN_INFO "manufacture id is 0x%x\n", e->manufacture_id); + printk(KERN_INFO "product id is 0x%x\n", e->product_id); + printk(KERN_INFO "port id is 0x%llx\n", (long long)e->port_id); + printk(KERN_INFO "HDCP support is %d\n", e->support_hdcp); + printk(KERN_INFO "AI support is %d\n", e->support_ai); + printk(KERN_INFO "SAD count is %d\n", e->sad_count); + printk(KERN_INFO "audio sync delay is %x\n", e->aud_synch_delay); + printk(KERN_INFO "connection type is %s\n", + eld_connection_type_names[e->conn_type]); + printk(KERN_INFO "monitor name is %s\n", e->monitor_name); + + hdmi_print_speaker_allocations(e->spk_alloc, buf, sizeof(buf)); + printk(KERN_INFO "speaker allocations: (0x%x)%s\n", e->spk_alloc, buf); + + for (i = 0; i < e->sad_count; i++) + hdmi_show_short_audio_desc(e->sad + i); +} + +#ifdef CONFIG_PROC_FS + +static void hdmi_print_sad_info(int i, struct cea_sad *a, + struct snd_info_buffer *buffer) +{ + snd_iprintf(buffer, "sad%d_coding_type\t%s\n", + i, cea_audio_coding_type_names[a->format]); + snd_iprintf(buffer, "sad%d_channels\t\t%d\n", i, a->channels); + snd_iprintf(buffer, "sad%d_sampling_rates\t0x%x\n", i, a->rates); + + if (a->format == AUDIO_CODING_TYPE_LPCM) + snd_iprintf(buffer, "sad%d_sample_bits\t0x%x\n", i, a->sample_bits); + + if (a->max_bitrate) + snd_iprintf(buffer, "sad%d_max_bitrate\t%d HZ\n", + i, a->max_bitrate); + + if (a->profile) + snd_iprintf(buffer, "sad%d_profile\t\t%d\n", i, a->profile); +} + +static void hdmi_print_eld_info(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct sink_eld *e = entry->private_data; + char buf[80]; + int i; + + snd_iprintf(buffer, "monitor name\t\t%s\n", e->monitor_name); + snd_iprintf(buffer, "connection_type\t\t%s\n", + eld_connection_type_names[e->conn_type]); + snd_iprintf(buffer, "eld_version\t\t%d:%s\n", e->eld_ver, + eld_versoin_names[e->eld_ver]); + snd_iprintf(buffer, "edid_version\t\t%d:%s\n", e->cea_edid_ver, + cea_edid_version_names[e->cea_edid_ver]); + snd_iprintf(buffer, "manufacture_id\t\t0x%x\n", e->manufacture_id); + snd_iprintf(buffer, "product_id\t\t0x%x\n", e->product_id); + snd_iprintf(buffer, "port_id\t\t\t0x%llx\n", (long long)e->port_id); + snd_iprintf(buffer, "support_hdcp\t\t%d\n", e->support_hdcp); + snd_iprintf(buffer, "support_ai\t\t%d\n", e->support_ai); + snd_iprintf(buffer, "audio_sync_delay\t%d ms\n", e->aud_synch_delay); + + hdmi_print_speaker_allocations(e->spk_alloc, buf, sizeof(buf)); + snd_iprintf(buffer, "speakers\t\t%s\n", buf); + + snd_iprintf(buffer, "sad_count\t\t%d\n", e->sad_count); + + for (i = 0; i < e->sad_count; i++) + hdmi_print_sad_info(i, e->sad + i, buffer); +} + +int snd_hda_eld_proc_new(struct hda_codec *codec, struct sink_eld *eld) +{ + char name[32]; + struct snd_info_entry *entry; + int err; + + snprintf(name, sizeof(name), "eld#%d", codec->addr); + err = snd_card_proc_new(codec->bus->card, name, &entry); + if (err < 0) + return err; + + snd_info_set_text_ops(entry, eld, hdmi_print_eld_info); + return 0; +} + +#endif --- sound-2.6.orig/sound/pci/Kconfig +++ sound-2.6/sound/pci/Kconfig @@ -582,6 +582,10 @@ config SND_HDA_CODEC_INTELHDMI Say Y here to include INTEL HDMI HD-audio codec support in snd-hda-intel driver, such as Eaglelake integrated HDMI. +config SND_HDA_ELD + def_bool y + depends on SND_HDA_CODEC_INTELHDMI + config SND_HDA_CODEC_CONEXANT bool "Build Conexant HD-audio codec support" depends on SND_HDA_INTEL --- sound-2.6.orig/sound/pci/hda/Makefile +++ sound-2.6/sound/pci/hda/Makefile @@ -4,6 +4,7 @@ snd-hda-intel-y := hda_intel.o # designed to be individual modules snd-hda-intel-y += hda_codec.o snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o +snd-hda-intel-$(CONFIG_SND_HDA_ELD) += hda_eld.o snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o snd-hda-intel-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o --- sound-2.6.orig/sound/pci/hda/hda_local.h +++ sound-2.6/sound/pci/hda/hda_local.h @@ -276,15 +276,6 @@ static inline int snd_hda_parse_generic_ #endif /* - * generic proc interface - */ -#ifdef CONFIG_PROC_FS -int snd_hda_codec_proc_new(struct hda_codec *codec); -#else -static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; } -#endif - -/* * Misc */ int snd_hda_check_board_config(struct hda_codec *codec, int num_configs, @@ -440,4 +431,57 @@ int snd_hda_check_amp_list_power(struct #define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) #define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) +/* + * CEA Short Audio Descriptor data + */ +struct cea_sad { + int channels; + int format; /* (format == 0) indicates invalid SAD */ + int rates; + int sample_bits; /* for LPCM */ + int max_bitrate; /* for AC3...ATRAC */ + int profile; /* for WMAPRO */ +}; + +#define ELD_FIXED_BYTES 20 +#define ELD_MAX_MNL 16 +#define ELD_MAX_SAD 16 + +/* + * ELD: EDID Like Data + */ +struct sink_eld { + int eld_size; + int baseline_len; + int eld_ver; /* (eld_ver == 0) indicates invalid ELD */ + int cea_edid_ver; + char monitor_name[ELD_MAX_MNL + 1]; + int manufacture_id; + int product_id; + u64 port_id; + int support_hdcp; + int support_ai; + int conn_type; + int aud_synch_delay; + int spk_alloc; + int sad_count; + struct cea_sad sad[ELD_MAX_SAD]; +}; + +int hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid); +int hdmi_get_eld(struct sink_eld *eld, struct hda_codec *codec, hda_nid_t nid); +void hdmi_show_eld(struct sink_eld *eld); + +/* + * generic proc interface + */ +#ifdef CONFIG_PROC_FS +int snd_hda_codec_proc_new(struct hda_codec *codec); +int snd_hda_eld_proc_new(struct hda_codec *codec, struct sink_eld *eld); +#else +static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; } +static inline int snd_hda_eld_proc_new(struct hda_codec *codec, + struct sink_eld *eld) { return 0; } +#endif + #endif /* __SOUND_HDA_LOCAL_H */

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[alsa-devel] [PATCH] ASoC: Fix for master playback/capture volume range for TWL4030 codec
by Peter Ujfalusi 14 Nov '08

14 Nov '08

FGAIN for playback/capture is in range of 0-0x3f. The original value of 128 (0x7f) would also modify the CGAIN, which gives undesired results. Separate control for CGAIN? Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 90f3b4d..95946d2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -192,10 +192,10 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 0, 127, 0), + 0, 0x3f, 0), SOC_DOUBLE_R("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, - 0, 127, 0), + 0, 0x3f, 0), }; /* add non dapm controls */ -- 1.6.0.3

1 1

[alsa-devel] sound-2.6.git master branch rebased
by Takashi Iwai 14 Nov '08

14 Nov '08

Hi, I had to rebase master branch on sound git tree now due to a temporary drop of topic/cs5535audio branch. The old master is saved as old-master branch for a while in case you need it. Sorry for inconvenience. Takashi

1 0

Re: [alsa-devel] S24_BLE does not play using aplay.
by Sean McNamara 13 Nov '08

13 Nov '08

Hi, >>> On Thu, Nov 13, 2008 at 2:07 PM, Baek Chang <baeksan(a)ccrma.stanford.edu> >>> wrote: >>> > Thanks for the reply. I get the following error when trying your >>> > suggestion >>> > >>> > aplay --device=plug:pulse LPCM_48khz_ST_24bit.wav >>> > >>> > Playing WAVE 'LPCM_48khz_ST_24bit.wav' : Signed 24 bit Little Endian in >>> > 3bytes, Rate 48000 Hz, Stereo >>> > aplay: set_params:1022: Unable to install hw params: >>> > ACCESS: RW_INTERLEAVED >>> > FORMAT: S24_3LE >>> > SUBFORMAT: STD >>> > SAMPLE_BITS: 24 >>> > FRAME_BITS: 48 >>> > CHANNELS: 2 >>> > RATE: 48000 >>> > PERIOD_TIME: 125000 >>> > PERIOD_SIZE: 6000 >>> > PERIOD_BYTES: 36000 >>> > PERIODS: 4 >>> > BUFFER_TIME: 500000 >>> > BUFFER_SIZE: 24000 >>> > BUFFER_BYTES: 144000 >>> > TICK_TIME: 0 >>> > >>> > >>> > Any ideas? >>> > >>> > Thanks >>> > Baek >>> > Some initial debugging shows: 1. It's not in alsa-plugins/pulse, because if you send it 24-bit anything, it won't recognize that format and error out with a different message (rather than just throwing error back at aplay). Things fall down before pcm_pulse.c ever evaluates the sample format. 2. It's not in alsa-lib/pcm/pcm_plug.c, because plug intelligently works with other (non-pulse) devices: Works: aplay --device=plug:hw:0 LPCM_48khz_ST_24bit.wav and aplay --device=plug:dmix:0 LPCM_48khz_ST_24bit.wav In theory, these statements should be both true: My hw doesn't support S24_3LE, but plug converts it for me. My pulse doesn't support S24_3LE, but plug converts it for me. But there is one more layer between plug and the actual pulse plugin code: ioplug! It's non-trivial to figure out the guts of functions such as snd_pcm_ioplug_hw_refine when you have refine math going on. The refine code goes through a lot of contortions trying to get ioplug to accept a conversion, but no joy.... it appears pcm_extplug.c, the "External Filter Plugin SDK" is unhappy. So basically, the problem occurs before alsa-plugins/pulse even gets a chance to detect and deal with the problem. pcm_ioplug.c does something similar to what pcm_plug.c does with refines, just more cryptically. Since it was developed so long ago, it may not have been tested with S24_3LE, or with S24 at all. One possible ugliness is that there is no cmpxchg instruction for 24-bit values, so you'd have to do something else for synchronization. Maybe that's why it's not supported? It sounds like pulse (or any ioplug-user, in general) is not going to support S24_3LE until someone really digs in and finds out why this is erroring out. Keep in mind that there are two code paths that both hit the same problem, namely plug:pulse and pulse alone. The error returned is different, and it seems simpler to figure out why you get an error with plain pulse, rather than complicating things with plug. HTH, Sean >>> > >>> > On Thu, Nov 13, 2008 at 11:04 AM, Sean McNamara <smcnam(a)gmail.com> >>> > wrote: >>> >> >>> >> Hmm, sounds like the ALSA<->Pulse bindings don't support this sample >>> >> format. Have you tried >>> >> >>> >> aplay --device=plug:pulse foo.wav >>> >> >>> >> Maybe the ALSA plug layer can convert it to a sample format that the >>> >> pulse plugin will accept. I don't have a file like this so I can't >>> >> test. >>> >> >>> >> Sean >>> >> >>> >> On Thu, Nov 13, 2008 at 1:44 PM, Baek Chang >>> >> <baeksan(a)ccrma.stanford.edu> >>> >> wrote: >>> >> > I am using pulseaudio as the sound server in conjunction with alsa. >>> >> > My >>> >> > alsa >>> >> > sinks are mapped to pulseaudio in asound.conf. Using paplay with a >>> >> > 24 >>> >> > bit >>> >> > file seems to work fine, because pulseaudio just converts the S24 >>> >> > bit >>> >> > data >>> >> > to 32 bit float. Is there a way to bypass using plughw with alsa, >>> >> > so >>> >> > that >>> >> > alsa can just send the 24 bit pcm data to pulseaudio, then have >>> >> > pulseaudio >>> >> > do the neccessary conversion? >>> >> > >>> >> > Thanks >>> >> > Baek >>> >> > >>> >> > On Fri, Nov 7, 2008 at 6:08 AM, Clemens Ladisch <clemens(a)ladisch.de> >>> >> > wrote: >>> >> > >>> >> >> Baek Chang wrote: >>> >> >> > Playing WAVE 'sine_24_48k.wav' : Signed 24 bit Little Endian in >>> >> >> > 3bytes, >>> >> >> Rate 48000 Hz, Stereo >>> >> >> > aplay: set_params:961: Sample format non available >>> >> >> >>> >> >> This error message indicates that the device does not support the >>> >> >> sample >>> >> >> format. >>> >> >> >>> >> >> You didn't show the aplay command line, but I'd guess that you used >>> >> >> a >>> >> >> "hw" device. Try "plughw" or "default" to get automatic sample >>> >> >> format >>> >> >> conversion. >>> >> >> >>> >> >> >>> >> >> HTH >>> >> >> Clemens >>> >> >> >>> >> >> >>> >> > >>> >> > >>> >> > -- >>> >> > Baek Chang >>> >> > baeksan(a)ccrma.stanford.edu >>> >> > 407-267-6701 >>> >> > _______________________________________________ >>> >> > Alsa-devel mailing list >>> >> > Alsa-devel(a)alsa-project.org >>> >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>> >> > >>> >> >>> > >>> > >>> > >>> > -- >>> > Baek Chang >>> > baeksan(a)ccrma.stanford.edu >>> > 407-267-6701 >>> > >> >> >> >> -- >> Baek Chang >> baeksan(a)ccrma.stanford.edu >> 407-267-6701 > > > > -- > Baek Chang > baeksan(a)ccrma.stanford.edu > 407-267-6701 >

1 0

[alsa-devel] Creative Sound Blaster X-Fi and X-Fi Titanium Series Linux 32bit/64bit Driver Source released
by Bjoern Olausson 13 Nov '08

13 Nov '08

Creative released a driver today.... I am hasitating to install it.. the last one was so crappy and I just installed my gentoo new ;-) But it looks really like the SOURCE, so maybe it will help to pimp the ALSA driver (would rather see an implementation in alsa then an driver from Creative) http://forums.creative.com/creativelabs/board/message?board.id=soundblaster… http://support.creative.com/downloads/download.aspx?nDownloadId=10792 Kind regards Bjoern

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[alsa-devel] S24_BLE does not play using aplay.
by Baek Chang 13 Nov '08

13 Nov '08

Hi, I am testing some things with aplay and I am trying to play a stereo Signed 24-bit 3 byte, 48kHz PCM file. I get the following error: Playing WAVE 'sine_24_48k.wav' : Signed 24 bit Little Endian in 3bytes, Rate 48000 Hz, Stereo aplay: set_params:961: Sample format non available Does aplay only work for S24_LE in 4 bytes? Thanks in advance Baek

2 2

[alsa-devel] Asus P5QL-EM motherboard (ALC1200 codec) - no sound through S/PDIF
by Andrei Tanas 13 Nov '08

13 Nov '08

Hello, I can't get the S/PDIF out to work on P5QL-EM motherboard. First of all, it doesn't detect the board configuration correctly, the motherboard has an HDMI output and an S/PDIF header (I have a TV connected to one and an amplifier to the other, in case one output will "go live"). aplay -L normally only shows th HDMI port, however if I add this options snd-hda-intel model=6stack-dig probe_mask=1 to /etc/modprobe.d/alsa-base, it will show both. Problem is, there seems to be no signal coming out of any of the two digital ports (they are not muted in mixer) in either PCM or AC3 pass-through mode. I compiled the "alsa-kernel" hoping that the latest commits related to ALC1200 would help, but it hasn't helped. Here's the output of alsa-info.sh: http://www.alsa-project.org/db/?f=b62b56c5ce0aa383fef1be0f55b87b4e27e8ab63 Again, it's running "alsa-kernel", checked in just 12 hours ago. Let me know what I can do to help, I am a developer but Alsa is not my territory. Thanks

3 14

[alsa-devel] [PATCH RESEND 2/2] Adds audo definition for SMDK2410
by Christian Pellegrin 13 Nov '08

13 Nov '08

Adds audio support for SMDK2410 Generated on 20081113 against f7e1798a78a30bae1cf3ebc3e1b6f7c1bac57756..5b7f05e9b57e7e442b1054cbb4423cf9909fb204 Signed-off-by: Christian Pellegrin <chripell(a)fsfe.org> --- arch/arm/mach-s3c2410/mach-smdk2410.c | 18 ++++++++++++++++++ 1 files changed, 18 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-s3c2410/mach-smdk2410.c b/arch/arm/mach-s3c2410/mach-smdk2410.c index 152527b..0001d52 100644 --- a/arch/arm/mach-s3c2410/mach-smdk2410.c +++ b/arch/arm/mach-s3c2410/mach-smdk2410.c @@ -43,6 +43,7 @@ #include <asm/mach/irq.h> #include <mach/hardware.h> +#include <mach/regs-gpio.h> #include <asm/irq.h> #include <asm/mach-types.h> @@ -53,6 +54,8 @@ #include <plat/common-smdk.h> +#include <sound/s3c24xx_uda134x.h> + static struct map_desc smdk2410_iodesc[] __initdata = { /* nothing here yet */ }; @@ -85,12 +88,27 @@ static struct s3c2410_uartcfg smdk2410_uartcfgs[] __initdata = { } }; +static struct s3c24xx_uda134x_platform_data s3c24xx_uda134x_data = { + .l3_clk = S3C2410_GPB4, + .l3_data = S3C2410_GPB3, + .l3_mode = S3C2410_GPB2, + .model = UDA134X_UDA1341, +}; + +static struct platform_device s3c24xx_uda134x = { + .name = "s3c24xx_uda134x", + .dev = { + .platform_data = &s3c24xx_uda134x_data, + } +}; + static struct platform_device *smdk2410_devices[] __initdata = { &s3c_device_usb, &s3c_device_lcd, &s3c_device_wdt, &s3c_device_i2c, &s3c_device_iis, + &s3c24xx_uda134x, }; static void __init smdk2410_map_io(void) -- 1.4.4.4

1 0

[alsa-devel] [PATCH] ASoC: DaVinci: Audio: Fix swapping of channels at start of stereo playback
by naresh＠ti.com 13 Nov '08

13 Nov '08

From: Naresh Medisetty <naresh(a)ti.com> Fixes swapping of channels at start of stereo playback. Channel swap can be observed while playing left-only or right-only audio data. The channel swap is fixed by handling the XSYNCERR condition. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- sound/soc/davinci/davinci-i2s.c | 49 ++++++++++++++++++++++++++++++++++++-- 1 files changed, 46 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 5ebf1ff..d8279ad 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -110,16 +110,59 @@ static void davinci_mcbsp_start(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_platform *platform = socdev->platform; u32 w; + int ret; /* Start the sample generator and enable transmitter/receiver */ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Stop the DMA to avoid data loss */ + /* while the transmitter is out of reset to handle XSYNCERR */ + if (platform->pcm_ops->trigger) { + ret = platform->pcm_ops->trigger(substream, + SNDRV_PCM_TRIGGER_STOP); + if (ret < 0) + printk(KERN_DEBUG "Playback DMA stop failed\n"); + } + + /* Enable the transmitter */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); - else + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* wait for any unexpected frame sync error to occur */ + udelay(100); + + /* Disable the transmitter to clear any outstanding XSYNCERR */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* Restart the DMA */ + if (platform->pcm_ops->trigger) { + ret = platform->pcm_ops->trigger(substream, + SNDRV_PCM_TRIGGER_START); + if (ret < 0) + printk(KERN_DEBUG "Playback DMA start failed\n"); + } + /* Enable the transmitter */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + } else { + + /* Enable the reciever */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } + /* Start frame sync */ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); -- 1.5.4.1

3 4

[alsa-devel] [PATCH] ARM DaVinci: Add Right-Justified mode and Codec clock master to davinci-i2s
by hugo.villeneuve＠lyrtech.com 13 Nov '08

13 Nov '08

From: Hugo Villeneuve <hugo.villeneuve(a)lyrtech.com> ARM DaVinci: Add Right-Justified mode and Codec clock master to davinci-i2s The TI DVEVM board uses the SND_SOC_DAIFMT_CBM_CFM & I2S formats, but the Lyrtech SFFSDR board uses the SND_SOC_DAIFMT_CBM_CFS & RIGHT-JUSTIFIED formats. Signed-off-by: Hugo Villeneuve <hugo.villeneuve(a)lyrtech.com> --- sound/soc/davinci/davinci-i2s.c | 40 ++++++++++++++++++++++++++++++++------ 1 files changed, 33 insertions(+), 7 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index abb5fed..d814ec8 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -59,6 +59,7 @@ #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) #define DAVINCI_MCBSP_PCR_FSRP (1 << 2) #define DAVINCI_MCBSP_PCR_FSXP (1 << 3) +#define DAVINCI_MCBSP_PCR_SCLKME (1 << 7) #define DAVINCI_MCBSP_PCR_CLKRM (1 << 8) #define DAVINCI_MCBSP_PCR_CLKXM (1 << 9) #define DAVINCI_MCBSP_PCR_FSRM (1 << 10) @@ -171,6 +172,16 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, DAVINCI_MCBSP_SRGR_FSGM); break; + case SND_SOC_DAIFMT_CBM_CFS: + /* McBSP CLKR pin is the input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate Generator. */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, + DAVINCI_MCBSP_PCR_SCLKME | + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, + DAVINCI_MCBSP_SRGR_FSGM); + break; case SND_SOC_DAIFMT_CBM_CFM: davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0); break; @@ -205,6 +216,28 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, + DAVINCI_MCBSP_RCR_RFRLEN1(1) | + DAVINCI_MCBSP_RCR_RDATDLY(0)); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, + DAVINCI_MCBSP_XCR_XFRLEN1(1) | + DAVINCI_MCBSP_XCR_XDATDLY(0) | + DAVINCI_MCBSP_XCR_XFIG); + break; + case SND_SOC_DAIFMT_I2S: + default: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, + DAVINCI_MCBSP_RCR_RFRLEN1(1) | + DAVINCI_MCBSP_RCR_RDATDLY(1)); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, + DAVINCI_MCBSP_XCR_XFRLEN1(1) | + DAVINCI_MCBSP_XCR_XDATDLY(1) | + DAVINCI_MCBSP_XCR_XFIG); + break; + } + return 0; } @@ -223,13 +256,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, - DAVINCI_MCBSP_RCR_RFRLEN1(1) | - DAVINCI_MCBSP_RCR_RDATDLY(1)); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, - DAVINCI_MCBSP_XCR_XFRLEN1(1) | - DAVINCI_MCBSP_XCR_XDATDLY(1) | - DAVINCI_MCBSP_XCR_XFIG); i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);

4 3

[alsa-devel] Git branches with Gentoo fixes
by flameeyes＠gmail.com 13 Nov '08

13 Nov '08

Hi all, Since I have just had to fix (again) an --as-needed issues on alsa-tools on the Gentoo ebuild, I've decided to clone the git repo and see to try pushing (again, I'm sure I tried before) the Gentoo-specific patches. You can find two git branches for alsa-tools and alsa-utils at these addresses: git://git.flameeyes.eu/others/alsa/alsa-tools.git git://git.flameeyes.eu/others/alsa/alsa-utils.git For the patches coming from users, the proper Author setting was done so that they are reported coming from the correct contributor. HTH, -- Diego "Flameeyes" Pettenò http://blog.flameeyes.eu/

3 12

[alsa-devel] [topic/asoc][PATCH] ASoC: OMAP: Add more supported sample rates into McBSP DAI driver
by Jarkko Nikula 13 Nov '08

13 Nov '08

Originally it was put too tight limits to support only 44.1 kHz and 48 kHz sample rates in McBSP DAI driver. Extend it now to 8 kHz - 96 kHz. With 96 kHz and 2*16 bits, bit clock is 3.072 MHz < 3.125 MHz (I2S max?). Tested on Nokia N810 with TVL320AIC33 from rates 8 - 96 kHz and on Texas Instruments Beagle with TWL4030 from rates 8 - 48 kHz. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Steve Sakoman <steve(a)sakoman.com> Cc: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/omap-mcbsp.c | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8485a8a..3d4060b 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -36,9 +36,7 @@ #include "omap-mcbsp.h" #include "omap-pcm.h" -#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | \ - SNDRV_PCM_RATE_KNOT) +#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) struct omap_mcbsp_data { unsigned int bus_id; -- 1.5.6.5

4 3

[alsa-devel] 2.6.28-rc2-mm1 - sound semi-busted on Dell Latitude D820
by Valdis.Kletnieks＠vt.edu 13 Nov '08

13 Nov '08

Sometime between -mmotm1017 and -rc2-mm1, something odd happened to the sound subsystem. On my Dell Latitude D820, audio works OK if it's sitting there standalone, or in its docking station. However, when I plug in headphones, it goes silent. Sorry for not having any more detail, but I've not had a chance to bisect this down to a specific commit. I did run alsa-info for the two kernels, here's a diff between them. I also attached a copy of the full output for working -mmotm1017. (- lines are working -mmotm-1017, + are busted -rc2-mm1) --- alsa-info-mmotm1017 2008-11-10 14:17:04.000000000 -0500 +++ alsa-info-rc3-mm1 2008-11-10 14:13:17.000000000 -0500 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.51 !!################################ -!!Script ran on: Mon Nov 10 14:16:47 EST 2008 +!!Script ran on: Mon Nov 10 14:12:28 EST 2008 !!Linux Distribution @@ -15,7 +15,7 @@ !!Kernel Information !!------------------ -Kernel release: 2.6.27-mmotm1017 +Kernel release: 2.6.28-rc3-mm1 Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 @@ -25,7 +25,7 @@ !!ALSA Version !!------------ -Driver version: 1.0.18rc3 +Driver version: 1.0.18 Library version: 1.0.18rc3 Utilities version: 1.0.18rc3 @@ -39,7 +39,7 @@ !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel - HDA Intel at 0xefffc000 irq 314 + HDA Intel at 0xefffc000 irq 41 !!PCI Soundcards installed in the system @@ -165,8 +165,8 @@ Pin Default 0x0321121f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf - Pin-ctls: 0xc0: OUT HP - Unsolicited: tag=30, enabled=1 + Pin-ctls: 0x00: + Unsolicited: tag=3d, enabled=1 Connection: 1 0x0b Node 0x0e [Pin Complex] wcaps 0x400181: Stereo @@ -234,12 +234,12 @@ !!ALSA Device nodes !!----------------- Looks like it lost the bits that setup the headphone jack.

2 6

Re: [alsa-devel] [RFC][PATCH] ELD routines and proc interface
by Wu Fengguang 13 Nov '08

13 Nov '08

On Thu, Nov 13, 2008 at 10:21:53AM +0800, Wu Fengguang wrote: > Create hda_eld.c for ELD routines and proc interface. Here is a simple example of the proc interface, in which the ELD data structure has not been populated yet: # cat /proc/asound/card0/eld\#3 monitor name connection_type HDMI eld_version 0:reserved edid_version 0:no CEA EDID Timing Extension block present manufacture_id 0x0 product_id 0x0 port_id 0x0 support_hdcp 0 support_ai 0 audio_sync_delay 0 speakers sad_count 0

1 0

[alsa-devel] [PATCH] handle SiI1392 HDMI codec in patch_intelhdmi.c
by Wu Fengguang 13 Nov '08

13 Nov '08

Move the handling of SiI1392 HDMI codec from patch_atihdmi.c to patch_intelhdmi.c, which makes our ASUS P5E-VM HDMI board work. Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/hda/patch_atihdmi.c | 1 - sound/pci/hda/patch_intelhdmi.c | 1 + 2 files changed, 1 insertion(+), 1 deletion(-) --- sound-2.6.orig/sound/pci/hda/patch_atihdmi.c +++ sound-2.6/sound/pci/hda/patch_atihdmi.c @@ -193,7 +193,6 @@ struct hda_codec_preset snd_hda_preset_a { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, { .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi }, { .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_atihdmi }, - { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_atihdmi }, { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_atihdmi }, {} /* terminator */ }; --- sound-2.6.orig/sound/pci/hda/patch_intelhdmi.c +++ sound-2.6/sound/pci/hda/patch_intelhdmi.c @@ -924,5 +924,6 @@ struct hda_codec_preset snd_hda_preset_i { .id = 0x80862801, .name = "INTEL G45 DEVBLC", .patch = patch_intel_hdmi }, { .id = 0x80862802, .name = "INTEL G45 DEVCTG", .patch = patch_intel_hdmi }, { .id = 0x80862803, .name = "INTEL G45 DEVELK", .patch = patch_intel_hdmi }, + { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ };

1 0

[alsa-devel] [PATCH 0/3] enable HDMI audio output for HDMI monitors V2
by Wu Fengguang 13 Nov '08

13 Nov '08

Hello, We can now enjoy music on HDMI monitors that are attached to Intel G35/G45 chipsets with the following X.org intel driver patches [PATCH 1/3] introduce i830_hdmi_priv.has_hdmi_sink [PATCH 2/3] enable Intel G45 integrated HDMI audio output [PATCH 3/3] enable Intel G35 SDVO HDMI audio output _and_ the corresponding ALSA patch posted at http://mailman.alsa-project.org/pipermail/alsa-devel/2008-November/012158.h… The patches are tested OK on Intel DG45ID board, HP 2230s notebook and ASUS P5E-VM board. Since the HDMI audio requires both audio and video driver support to function well, I'd like to provide an overview here. 1) driver components: A) ALSA: audio driver (the above link) B) Xorg: audio output enabling (this patchset) C) Xorg: EDID/ELD information (patches to be submitted by Ma Ling) 2) summary of the feature sets: - basic 2-channel audio: (A) is required, (B) is mostly required, (C) is not needed - 2+ multichannel audio: not tested yet; in theory we need (C) to get HDMI monitor's speaker allocation configuration; there are also bandwidth constraints that should be coordinated between audio/video drivers in the future. - non-LPCM audio: not tested yet; need more work in ALSA code. 3) summary of the work flow: - basic audio output: (A) and (B) - ALSA HDMI driver: enable pin out and unmute - ALSA HDMI driver: fill audio infoframe and enable its transmission - Xorg intel driver: enable audio output - ELD info for advanced audio capabilities: (A) and (C) - Xorg xserver: get/parse/store EDID extensions - Xorg xserver: transform EDID into ELD - Xorg intel driver: feed ELD to hardware - Xorg intel driver: set ELD-Valid flag to inform audio driver of new ELD - ALSA HDMI driver: response to unsolicited response triggered by ELDV - ALSA HDMI driver: get ELD from hardware - ALSA HDMI driver: parse and show ELD info - ALSA HDMI driver: update hardware capabilities/constraints according to ELD (TBD) That describes my understandings of HDMI audio, comments and discussions are warmly welcome. Thank you, Fengguang --

3 5

Re: [alsa-devel] 2.6.27.2 hda-intel weak sound from speakers
by Takashi Iwai 12 Nov '08

12 Nov '08

At Wed, 12 Nov 2008 12:36:09 -0500, Stephen Clark wrote: > > Hi, > > I am running linux 2.6.27.2 ontop of a Fedora 8 distro. The sound from the > speakers is barely audible with all the mixer settings at max, this includes > the pulseaudio settings. This is on an ASUS laptop with intel-hda. Try to pass either model=asus-a7j or model=asus-a7m. One of them may give a bit better result. Takashi

2 1

[alsa-devel] maximum tolerable sampling frequency difference for a driver
by chri 12 Nov '08

12 Nov '08

Let's say that the driver cannot set the exact sampling frequency asked via hwparms but something near (because of the granularity of the result of the clock division). What is the maximum difference that should be accepted (in Hz or in %)? I guess that there are 2 factors playing a role: 1) if the frequency is too different then the upper layers report a undeflow/overflow 2) the sound just doesn't sound good (just a bit too presto or too adagio) So what difference should be accepted instead of returning -EINVAL? Thanks, -- Christian Pellegrin, see http://www.evolware.org/chri/ "Real Programmers don't play tennis, or any other sport which requires you to change clothes. Mountain climbing is OK, and Real Programmers wear their climbing boots to work in case a mountain should suddenly spring up in the middle of the computer room."

4 4

[alsa-devel] [PATCH 0/4] ASoC updates
by Mark Brown 12 Nov '08

12 Nov '08

The following changes since commit e18c94d20224f3df584531a48d944d8cccfda46d: Grazvydas Ignotas (1): ALSA: ASoC: TWL4030 codec - fix 256*Fs clock are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Christian Pellegrin (1): ASoC: s3c24xx 8 bit sound fix Hugo Villeneuve (1): ASoC: Add Right-Justified mode and Codec clock master to davinci-i2s Naresh Medisetty (1): ASoC: DaVinci: Audio: Fix swapping of channels at start of stereo playback Troy Kisky (1): ASoC: TLV320AIC23B Support more sample rates sound/soc/codecs/tlv320aic23.c | 222 +++++++++++++++++++++++++++++++-------- sound/soc/davinci/davinci-i2s.c | 89 ++++++++++++++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 7 ++ 3 files changed, 263 insertions(+), 55 deletions(-)

5 13

[alsa-devel] [topic/asoc][PATCH] ASoC: Fix supported sample rates of TWL4030 audio codec
by Jarkko Nikula 12 Nov '08

12 Nov '08

TWL4030 currently supports rates between 8 kHz and 48 kHz and sets the codec mode register accordingly in twl4030_hw_params. Expose this info so that ASoC can match other rates than 44.1 kHz or 48 kHz as well. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Steve Sakoman <steve(a)sakoman.com> --- sound/soc/codecs/twl4030.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 90f3b4d..c1893d2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -504,7 +504,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -#define TWL4030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) struct snd_soc_dai twl4030_dai = { -- 1.5.6.5

3 2

[alsa-devel] RFC: add possibility to change text identification string
by Jaroslav Kysela 12 Nov '08

12 Nov '08

Hi all, based on discussion today, I prepared a patch to handle udev-style card renaming. Comments are welcome. Jaroslav

2 4

Re: [alsa-devel] Regression on alsa-driver-1.0.18a on HDA for Dell Inspiron 1521
by Takashi Iwai 12 Nov '08

12 Nov '08

At Wed, 12 Nov 2008 12:14:45 -0200, Manuel Naranjo wrote: > > > > I guess this has been fixed on my tree. > > Try the snapshot tarball below > > ftp://ftp.kernel.org/pub/kernel/linux/people/tiwai/snapshot/alsa-driver-sna… > > > > > > Takashi > > > Ok thanks, I give it a try, the url was wrong it should be: > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… Oh yes, thanks for correction. BTW, .gz is the original tarball, and .bz2 is the one created automatically on kernel.org. Thus .bz2 might be missing sometimes. That's why I showed .gz in the above. Takashi

1 0

[alsa-devel] amixer error
by Arun KS 12 Nov '08

12 Nov '08

Hi, I am getting an error for amixer, but my amixer controls works fine. Here is the log: [root@OMAP2530EVM ~]# [root@OMAP2530EVM ~]# [root@OMAP2530EVM ~]# [root@OMAP2530EVM ~]# amixer amixer: Mixer default load error: Operation not permitted [root@OMAP2530EVM ~]# [root@OMAP2530EVM ~]# [root@OMAP2530EVM ~]# [root@OMAP2530EVM ~]# amixer controls numid=2,iface=MIXER,name='PCM Playback Volume' numid=22,iface=MIXER,name='Line Input Function' numid=20,iface=MIXER,name='Line Out Function' numid=9,iface=MIXER,name='Mic Bias Level' numid=23,iface=MIXER,name='Mic Function' numid=5,iface=MIXER,name='HP DAC Playback Switch' numid=3,iface=MIXER,name='HP Driver Gain' numid=1,iface=MIXER,name='ICE1724 EEPROM' numid=21,iface=MIXER,name='Jack Function' numid=6,iface=MIXER,name='LO DAC Playback Switch' numid=4,iface=MIXER,name='LO Driver Gain' numid=18,iface=MIXER,name='Left DAC_L Mixer HP Switch' numid=19,iface=MIXER,name='Left DAC_L Mixer LO Switch' numid=13,iface=MIXER,name='Left PGA Mixer Line1L Switch' numid=14,iface=MIXER,name='Left PGA Mixer Line2L Switch' numid=15,iface=MIXER,name='Left PGA Mixer Line3L Switch' numid=7,iface=MIXER,name='PGA Capture Volume' numid=8,iface=MIXER,name='Program Registers' numid=16,iface=MIXER,name='Right DAC_R Mixer HP Switch' numid=17,iface=MIXER,name='Right DAC_R Mixer LO Switch' numid=10,iface=MIXER,name='Right PGA Mixer Line1R Switch' numid=11,iface=MIXER,name='Right PGA Mixer Line2R Switch' numid=12,iface=MIXER,name='Right PGA Mixer Line3R Switch' [root@OMAP2530EVM ~]# Can somebody tell me what all can be the reasons for this? Should I need to enable some option in the kernel? Regards, Arun

4 7

[alsa-devel] alsa-driver 1.0.18a is out
by Jaroslav Kysela 12 Nov '08

12 Nov '08

Hi all, I released alsa-driver package 1.0.18a. The changelog is here: http://www.alsa-project.org/main/index.php/Changes_v1.0.18_v1.0.18a Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

1 0

Re: [alsa-devel] [PATCH 1/3] introduce i830_hdmi_priv.has_hdmi_sink
by Wu Fengguang 12 Nov '08

12 Nov '08

On Wed, Nov 12, 2008 at 03:27:47PM +0800, Wang, Zhenyu Z wrote: > On 2008.11.07 14:23:39 +0800, Wu Fengguang wrote: > > HDMI is compatible with DVI, and we've seen many boards that > > use HDMI port for DVI output. > > > > So Zhenyu proposed this flag: i830_hdmi_priv.has_hdmi_sink > > to indicate the presence of HDMI capable monitors. > > > > Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> > We should check if the symbol is available for xserver compatiblity. > So below is updated patch for check this. > + > + if (xf86LoaderCheckSymbol("xf86MonitorIsHDMI") && > + xf86MonitorIsHDMI(edid_mon)) > + dev_priv->has_hdmi_sink = TRUE; I'm OK with the backward compatibility check. The side effect is that audio output won't be enabled for old xservers. Thanks, Fengguang

1 0

[alsa-devel] to get the detail information of a codec
by Wayne 12 Nov '08

12 Nov '08

Hi all, Now I'm writing a program which can list some capabilities of my codec, such as what rates and formats my codec supports. By typing 'cat /proc/asound/card0/codec#0', you can see the detail of your card. I want to write a program which can show these information by alsa-lib. However, I couldn't find a function in alsa-lib which can tell me the formats and rates. Is there a way to get this detail? By the way, my card is intel-hda and the codec is alc268. Another question is I've seen the word "element" in many function, such as snd_ctl_event_elem_get_device. What "element" means in alsa world? I couldn't find any artile mentioning it!!! Thanks all!!! Wayne _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

1 0

[alsa-devel] Dell Vostro 1710 laptop
by Tim Uckun 12 Nov '08

12 Nov '08

Hey all. I have a dell vostro 1710 laptop running hardy. The sound has never worked perfectly on this model. The built in mic doesn't work but a mic plugged into the jack works. The speakers sound terrible and they are not loud enough. The sound also dies frequently especially I am on youtube a lot (could be due to a flash/firefox problem too). I have to restart the alsa service in order to get it going again. Until recently I was running the option model=dell-m42 but a recent kernel upgrade broke that. I fiddled with some more options and settled on model=dell. I don't know if any of these problems will be solved by upgrading ubuntu but I wanted to report the problems anyway. I have been googling around I can't see anybody else using a vostro 1710 with linux. The alsa information is here. http://www.alsa-project.org/db/?f=a27b1377d3d2f472364ead7a04a1db7360affecd

1 0

[alsa-devel] [PATCH] ALSA SOC driver for s3c24xx with uda1341
by Christian Pellegrin 12 Nov '08

12 Nov '08

This patch adds support for the UDA1341 codec and a sound card definition for one of these UDAs connected to the s3c24xx. It is *heavily* based on the one made by Zoltan Devai but: 1 since the UDA is the only use of the L3 protocol I can find I just embedded a stripped-down L3 bit-banging algorithm from the original RMK work. It is really small. 2 the driver has the possibility to specify the pins used by codec via platform data so it can work on SMDK2410, SMDK2440 or any custom design. 3 it tries to guess the right clock source and divider so it is not tied to a particular crystal used. 4 it fixes some bugs. Thank you for reviews/comments. Generated on 20081108 against v2.6.27 Signed-off-by: Christian Pellegrin <chripell(a)fsfe.org> --- include/sound/s3c24xx_uda1341.h | 41 ++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/uda1341/Makefile | 3 + sound/soc/codecs/uda1341/l3.c | 106 +++++ sound/soc/codecs/uda1341/l3.h | 8 + sound/soc/codecs/uda1341/uda1341.c | 537 ++++++++++++++++++++++ sound/soc/codecs/uda1341/uda1341.h | 47 ++ sound/soc/codecs/uda1341/uda1341_platform_data.h | 18 + sound/soc/s3c24xx/Kconfig | 7 +- sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c24xx_uda1341.c | 314 +++++++++++++ 12 files changed, 1086 insertions(+), 1 deletions(-) diff --git a/include/sound/s3c24xx_uda1341.h b/include/sound/s3c24xx_uda1341.h new file mode 100644 index 0000000..bd80bd3 --- /dev/null +++ b/include/sound/s3c24xx_uda1341.h @@ -0,0 +1,41 @@ +#ifndef _S3C24XX_UDA1341_H_ +#define _S3C24XX_UDA1341_H_ 1 + +/* + +Example usage (pins for SMDK2410). Put something like this in your +machine file: + +... + +static struct s3c24xx_uda1341_platform_data s3c24xx_uda1341_data = { + .l3_clk = S3C2410_GPB4, + .l3_data = S3C2410_GPB3, + .l3_mode = S3C2410_GPB2, +}; + +static struct platform_device s3c24xx_uda1341 = { + .name = "s3c24xx_uda1341", + .dev = { + .platform_data = &s3c24xx_uda1341_data, + } +}; + +... + +static struct platform_device *smdk2410_devices[] __initdata = { +... + &s3c24xx_uda1341, +... +}; + + */ + +struct s3c24xx_uda1341_platform_data { + int l3_clk; + int l3_mode; + int l3_data; + void (*power) (int); +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1db04a2..4b483b7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -50,3 +50,6 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_TLV320AIC3X tristate depends on I2C + +config SND_SOC_UDA1341 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d7b97ab..cbace60 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,3 +23,4 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_UDA1341) += uda1341/ diff --git a/sound/soc/codecs/uda1341/Makefile b/sound/soc/codecs/uda1341/Makefile new file mode 100644 index 0000000..f9869ba --- /dev/null +++ b/sound/soc/codecs/uda1341/Makefile @@ -0,0 +1,3 @@ +snd-soc-uda1341-objs := l3.o uda1341.o + +obj-$(CONFIG_SND_SOC_UDA1341) += snd-soc-uda1341.o \ No newline at end of file diff --git a/sound/soc/codecs/uda1341/l3.c b/sound/soc/codecs/uda1341/l3.c new file mode 100644 index 0000000..b65b352 --- /dev/null +++ b/sound/soc/codecs/uda1341/l3.c @@ -0,0 +1,106 @@ +/* + * L3 code + * + * Copyright (C) 2008, Christian Pellegrin <chripell(a)evolware.org> + * + * 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. + * + * + * based on: + * + * L3 bus algorithm module. + * + * Copyright (C) 2001 Russell King, All Rights Reserved. + * + * + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/errno.h> +#include <linux/sched.h> + +#include "l3.h" + +/*#define L3_DEBUG 1*/ +#ifdef L3_DEBUG +#define DBG(format, arg...) \ + printk(KERN_DEBUG "L3: %s:" format "\n" , __func__, ## arg) +#else +#define DBG(format, arg...) do {} while (0) +#endif + +#define setdat(adap, val) (adap->setdat(adap, val)) +#define setclk(adap, val) (adap->setclk(adap, val)) +#define setmode(adap, val) (adap->setmode(adap, val)) + +/* + * Send one byte of data to the chip. Data is latched into the chip on + * the rising edge of the clock. + */ +static void sendbyte(struct uda1341_platform_data *adap, unsigned int byte) +{ + int i; + + DBG("%02x", byte); + + for (i = 0; i < 8; i++) { + setclk(adap, 0); + udelay(adap->data_hold); + setdat(adap, byte & 1); + udelay(adap->data_setup); + setclk(adap, 1); + udelay(adap->clock_high); + byte >>= 1; + } +} + +/* + * Send a set of bytes to the chip. We need to pulse the MODE line + * between each byte, but never at the start nor at the end of the + * transfer. + */ +static void sendbytes(struct uda1341_platform_data *adap, const u8 *buf, + int len) +{ + int i; + + for (i = 0; i < len; i++) { + if (i) { + udelay(adap->mode_hold); + setmode(adap, 0); + udelay(adap->mode); + } + setmode(adap, 1); + udelay(adap->mode_setup); + sendbyte(adap, buf[i]); + } +} + +int l3_write(struct uda1341_platform_data *adap, u8 addr, u8 *data, int len) +{ + setclk(adap, 1); + setdat(adap, 1); + setmode(adap, 1); + udelay(adap->mode); + + setmode(adap, 0); + udelay(adap->mode_setup); + sendbyte(adap, addr); + udelay(adap->mode_hold); + + sendbytes(adap, data, len); + + setclk(adap, 1); + setdat(adap, 1); + setmode(adap, 0); + + return len; +} + + diff --git a/sound/soc/codecs/uda1341/l3.h b/sound/soc/codecs/uda1341/l3.h new file mode 100644 index 0000000..2c31e0c --- /dev/null +++ b/sound/soc/codecs/uda1341/l3.h @@ -0,0 +1,8 @@ +#ifndef _L3_H_ +#define _L3_H_ 1 + +#include "uda1341_platform_data.h" + +int l3_write(struct uda1341_platform_data *adap, u8 addr, u8 *data, int len); + +#endif diff --git a/sound/soc/codecs/uda1341/uda1341.c b/sound/soc/codecs/uda1341/uda1341.c new file mode 100644 index 0000000..93cadf0 --- /dev/null +++ b/sound/soc/codecs/uda1341/uda1341.c @@ -0,0 +1,537 @@ +/* + * uda1341.c -- UDA1341 ALSA SoC Codec driver + * + * Modifications by Christian Pellegrin <chripell(a)evolware.org> + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "uda1341.h" +#include "l3.h" + +/*#define UDA1341_DEBUG 1*/ +#ifdef UDA1341_DEBUG +#define DBG(format, arg...) \ + printk(KERN_DEBUG "UDA1341: %s:" format "\n" , __func__, ## arg) +#else +#define DBG(format, arg...) do {} while (0) +#endif + +#define UDA1341_RATES SNDRV_PCM_RATE_8000_48000 +#define UDA1341_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) + +struct uda1341_priv { + int sysclk; + int dai_fmt; +}; + +/* In-data addresses are hard-coded into the reg-cache values */ +static const char uda1341_reg[UDA1341_REGS_NUM] = { + /* Extended address registers */ + 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, + /* Status, data regs */ + 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, +}; + +/* + * The codec has no support for reading its registers except for peak level... + */ +static inline unsigned int uda1341_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA1341_REGS_NUM) + return -1; + return cache[reg]; +} + +/* + * Write the register cache + */ +static inline void uda1341_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA1341_REGS_NUM) + return; + cache[reg] = value; +} + +/* + * Write to the uda1341 registers + * + */ +static int uda1341_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + u8 addr; + u8 data = value; + + DBG("reg: %02X, value:%02X", reg, value); + + if (reg >= UDA1341_REGS_NUM) { + DBG("Unkown register: reg: %d", reg); + return -EINVAL; + } + + uda1341_write_reg_cache(codec, reg, value); + + switch (reg) { + case UDA1341_STATUS0: + case UDA1341_STATUS1: + addr = UDA1341_STATUS_ADDR; + break; + case UDA1341_DATA000: + case UDA1341_DATA001: + case UDA1341_DATA010: + addr = UDA1341_DATA0_ADDR; + break; + case UDA1341_DATA1: + addr = UDA1341_DATA1_ADDR; + break; + default: + /* It's an extended address register */ + addr = (reg | UDA1341_EXTADDR_PREFIX); + + ret = l3_write((struct uda1341_platform_data *) + codec->control_data, + UDA1341_DATA0_ADDR, &addr, 1); + if (ret != 1) + return -EIO; + + addr = UDA1341_DATA0_ADDR; + data = (value | UDA1341_EXTDATA_PREFIX); + break; + } + + ret = l3_write((struct uda1341_platform_data *) codec->control_data, + addr, &data, 1); + if (ret != 1) + return -EIO; + + return 0; +} + +static inline void uda1341_reset(struct snd_soc_codec *codec) +{ + u8 reset_reg = uda1341_read_reg_cache(codec, UDA1341_STATUS0); + uda1341_write(codec, UDA1341_STATUS0, reset_reg | (1<<6)); + mdelay(1); + uda1341_write(codec, UDA1341_STATUS0, reset_reg & ~(1<<6)); +} + +static int uda1341_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = uda1341_read_reg_cache(codec, UDA1341_DATA010); + + DBG("mute: %d", mute); + + if (mute) + mute_reg |= (1<<2); + else + mute_reg &= ~(1<<2); + + uda1341_write(codec, UDA1341_DATA010, mute_reg & ~(1<<2)); + + return 0; +} + +static int uda1341_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + u8 hw_params = uda1341_read_reg_cache(codec, UDA1341_STATUS0); + hw_params &= STATUS0_SYSCLK_MASK; + hw_params &= STATUS0_DAIFMT_MASK; + + DBG("sysclk: %d, rate:%d", uda1341->sysclk, params_rate(params)); + + /* set SYSCLK / fs ratio */ + switch (uda1341->sysclk / params_rate(params)) { + case 512: + break; + case 384: + hw_params |= (1<<4); + break; + case 256: + hw_params |= (1<<5); + break; + default: + return -EINVAL; + break; + } + + DBG("dai_fmt: %d, params_format:%d", uda1341->dai_fmt, + params_format(params)); + + + /* set DAI format and word length */ + switch (uda1341->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + hw_params |= (1<<1); + break; + case SNDRV_PCM_FORMAT_S18_3LE: + hw_params |= (1<<2); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + hw_params |= ((1<<2) | (1<<1)); + break; + default: + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + hw_params |= (1<<3); + break; + default: + return -EINVAL; + } + + uda1341_write(codec, UDA1341_STATUS0, hw_params); + + return 0; +} + +static int uda1341_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + DBG("clk_id: %d, freq: %d, dir: %d", clk_id, freq, dir); + + /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable + we'll error out on set_hw_params if it's not OK */ + if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) { + uda1341->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int uda1341_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda1341_priv *uda1341 = codec->private_data; + + DBG("fmt: %08X", fmt); + + /* codec supports only full slave mode */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + /* no support for clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + + /* We can't setup DAI format here as it depends on the word bit num */ + /* so let's just store the value for later */ + uda1341->dai_fmt = fmt; + + return 0; +} + +static int uda1341_dapm_event(struct snd_soc_codec *codec, int event) +{ + u8 reg; + + DBG("event: %08X", event); + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + /* ADC, DAC on */ + reg = uda1341_read_reg_cache(codec, UDA1341_STATUS1); + uda1341_write(codec, UDA1341_STATUS1, reg | 0x03); + break; + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* mute, ADC, DAC power off */ + reg = uda1341_read_reg_cache(codec, UDA1341_STATUS1); + uda1341_write(codec, UDA1341_STATUS1, reg & ~(0x03)); + break; + } + return 0; +} + +static const char *uda1341_dsp_setting[] = {"Flat", "Minimum1", + "Minimum2", "Maximum"}; +static const char *uda1341_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *uda1341_mixmode[] = {"DD", "Input 2", + "Input 2", "Digital mixer"}; + +static const struct soc_enum uda1341_mixer_enum[] = { +SOC_ENUM_SINGLE(UDA1341_DATA010, 0, 0x03, uda1341_dsp_setting), +SOC_ENUM_SINGLE(UDA1341_DATA010, 3, 0x03, uda1341_deemph), +SOC_ENUM_SINGLE(UDA1341_EA010, 0, 0x03, uda1341_mixmode), +}; + +static const struct snd_kcontrol_new uda1341_snd_controls[] = { +SOC_SINGLE("Playback Volume", UDA1341_DATA000, 0, 0x3F, 1), +SOC_SINGLE("Mic gain", UDA1341_EA010, 2, 0x07, 0), +SOC_SINGLE("Channel 1 mixer gain", UDA1341_EA000, 0, 0x1F, 1), +SOC_SINGLE("Channgel 2 mixer gain", UDA1341_EA001, 0, 0x1F, 1), +SOC_SINGLE("Input channel 2 amp gain", UDA1341_EA101, 0, 0x1F, 0), + +SOC_SINGLE("Bass boost", UDA1341_DATA001, 2, 0xF, 0), +SOC_SINGLE("Treble", UDA1341_DATA001, 0, 3, 0), + +SOC_ENUM("DSP setting", uda1341_mixer_enum[0]), +SOC_ENUM("Playback De-emphasis", uda1341_mixer_enum[1]), +SOC_ENUM("Mixer mode", uda1341_mixer_enum[2]), + +/* This should be an ext control with own handler, if one wants + to set the values in 0.5dB steps instead of 3dB */ +SOC_SINGLE("AGC output level", UDA1341_EA110, 0, 0x03, 1), +SOC_SINGLE("AGC time const", UDA1341_EA110, 2, 0x07, 0), + +SOC_SINGLE("DAC Gain switch", UDA1341_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC Gain switch", UDA1341_STATUS1, 5, 1, 0), +SOC_SINGLE("ADC Polarity switch", UDA1341_STATUS1, 4, 1, 0), +SOC_SINGLE("DAC Polarity switch", UDA1341_STATUS1, 3, 1, 0), +SOC_SINGLE("Double speed playback switch", UDA1341_STATUS1, 2, 1, 0), +}; + +static int uda1341_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(uda1341_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&uda1341_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +struct snd_soc_dai uda1341_dai = { + .name = "UDA1341", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1341_RATES, + .formats = UDA1341_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1341_RATES, + .formats = UDA1341_FORMATS, + }, + /* pcm operations */ + .ops = { + .hw_params = uda1341_hw_params, + }, + /* DAI operations */ + .dai_ops = { + .digital_mute = uda1341_mute, + .set_sysclk = uda1341_set_dai_sysclk, + .set_fmt = uda1341_set_dai_fmt, + } +}; +EXPORT_SYMBOL(uda1341_dai); + + +static int uda1341_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda1341_priv *uda1341; + void *codec_setup_data = socdev->codec_data; + int ret = -ENOMEM; + struct uda1341_platform_data *pd; + + printk(KERN_INFO "UDA1341 SoC Audio Codec\n"); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return ret; + + codec = socdev->codec; + + uda1341 = kzalloc(sizeof(struct uda1341_priv), GFP_KERNEL); + if (uda1341 == NULL) + goto priv_err; + + codec->private_data = uda1341; + + codec->reg_cache = kmemdup(uda1341_reg, sizeof(uda1341_reg), + GFP_KERNEL); + + if (codec->reg_cache == NULL) + goto reg_err; + + mutex_init(&codec->mutex); + + codec->reg_cache_size = sizeof(uda1341_reg); + codec->reg_cache_step = 1; + + codec->name = "UDA1341"; + codec->owner = THIS_MODULE; + codec->dai = &uda1341_dai; + codec->num_dai = 1; + codec->read = uda1341_read_reg_cache; + codec->write = uda1341_write; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + if (!codec_setup_data) { + printk(KERN_ERR "UDA1341 SoC codec: " + "missing L3 bitbang function\n"); + ret = -ENODEV; + goto pcm_err; + } + + codec->control_data = codec_setup_data; + pd = (struct uda1341_platform_data *) codec_setup_data; + + if (pd->power) + pd->power(1); + + uda1341_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register pcms\n"); + goto pcm_err; + } + + ret = uda1341_add_controls(codec); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register controls\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "UDA1341: failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); +reg_err: + kfree(codec->private_data); +priv_err: + kfree(codec); + return ret; +} + +/* power down chip */ +static int uda1341_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct uda1341_platform_data *pd; + + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + + pd = (struct uda1341_platform_data *) codec->control_data; + if (pd->power) + pd->power(0); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} + +#if defined(CONFIG_PM) +static int uda1341_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct uda1341_platform_data *pd; + + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + + pd = (struct uda1341_platform_data *) codec->control_data; + if (pd->power) + pd->power(0); + + return 0; +} + +static int uda1341_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct uda1341_platform_data *pd; + int i; + u8 *cache = codec->reg_cache; + + pd = (struct uda1341_platform_data *) codec->control_data; + if (pd->power) + pd->power(1); + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda1341_reg); i++) + codec->write(codec, i, *cache++); + uda1341_dapm_event(codec, SNDRV_CTL_POWER_D0); + return 0; +} +#endif /* CONFIG_PM */ + +struct snd_soc_codec_device soc_codec_dev_uda1341 = { + .probe = uda1341_soc_probe, + .remove = uda1341_soc_remove, +#if defined(CONFIG_PM) + .suspend = uda1341_soc_suspend, + .resume = uda1341_soc_resume, +#endif /* CONFIG_PM */ +}; +EXPORT_SYMBOL(soc_codec_dev_uda1341); + +MODULE_DESCRIPTION("UDA1341 ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda1341/uda1341.h b/sound/soc/codecs/uda1341/uda1341.h new file mode 100644 index 0000000..5eccfc4 --- /dev/null +++ b/sound/soc/codecs/uda1341/uda1341.h @@ -0,0 +1,47 @@ +/* + * uda1341.h -- UDA1341 ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * 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. + */ + +#ifndef _UDA1341_H +#define _UDA1341_H + +#define UDA1341_L3ADDR 5 +#define UDA1341_DATA0_ADDR ((UDA1341_L3ADDR << 2) | 0) +#define UDA1341_DATA1_ADDR ((UDA1341_L3ADDR << 2) | 1) +#define UDA1341_STATUS_ADDR ((UDA1341_L3ADDR << 2) | 2) + +#define UDA1341_EXTADDR_PREFIX 0xC0 +#define UDA1341_EXTDATA_PREFIX 0xE0 + +/* UDA1341 registers */ +#define UDA1341_EA000 0 +#define UDA1341_EA001 1 +#define UDA1341_EA010 2 +#define UDA1341_EA011 3 +#define UDA1341_EA100 4 +#define UDA1341_EA101 5 +#define UDA1341_EA110 6 +#define UDA1341_EA111 7 +#define UDA1341_STATUS0 8 +#define UDA1341_STATUS1 9 +#define UDA1341_DATA000 10 +#define UDA1341_DATA001 11 +#define UDA1341_DATA010 12 +#define UDA1341_DATA1 13 + +#define UDA1341_REGS_NUM 14 + +#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4)) + +extern struct snd_soc_dai uda1341_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda1341; + +#endif /* _UDA1341_H */ diff --git a/sound/soc/codecs/uda1341/uda1341_platform_data.h b/sound/soc/codecs/uda1341/uda1341_platform_data.h new file mode 100644 index 0000000..26081cb --- /dev/null +++ b/sound/soc/codecs/uda1341/uda1341_platform_data.h @@ -0,0 +1,18 @@ +#ifndef _UDA1341_PLATFORM_DATA_H_ +#define _UDA1341_PLATFORM_DATA_H_ 1 + +struct uda1341_platform_data { + void (*setdat) (struct uda1341_platform_data *, int); + void (*setclk) (struct uda1341_platform_data *, int); + void (*setmode) (struct uda1341_platform_data *, int); + int data_hold; + int data_setup; + int clock_high; + int mode_hold; + int mode; + int mode_setup; + void *priv; + void (*power) (int); +}; + +#endif diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b9f2353..cc8fa5e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -16,7 +16,7 @@ config SND_S3C2443_SOC_AC97 tristate select AC97_BUS select SND_SOC_AC97_BUS - + config SND_S3C24XX_SOC_NEO1973_WM8753 tristate "SoC I2S Audio support for NEO1973 - WM8753" depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 @@ -44,3 +44,8 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650 Say Y if you want to add support for SoC audio on ln2440sbc with the ALC650. +config SND_S3C24XX_SOC_S3C24XX_UDA1341 + tristate "SoC I2S Audio support UDA1341 wired to a S3C24XX" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_UDA1341 \ No newline at end of file diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 0aa5fb0..155f5a4 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o +snd-soc-s3c24xx-uda1341-objs := s3c24xx_uda1341.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o +obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA1341) += snd-soc-s3c24xx-uda1341.o diff --git a/sound/soc/s3c24xx/s3c24xx_uda1341.c b/sound/soc/s3c24xx/s3c24xx_uda1341.c new file mode 100644 index 0000000..fc3c13e --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_uda1341.c @@ -0,0 +1,314 @@ +/* + * Modifications by Christian Pellegrin <chripell(a)evolware.org> + * + * s3c24xx_uda1341.c -- S3C24XX_UDA1341 ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/clk.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/s3c24xx_uda1341.h> + +#include <asm/mach-types.h> +#include <asm/plat-s3c24xx/regs-iis.h> +#include <mach/regs-gpio.h> +#include <mach/regs-gpioj.h> +#include <mach/hardware.h> + +#include "../codecs/uda1341/uda1341.h" +#include "../codecs/uda1341/uda1341_platform_data.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" + +/*#define S3C24XX_UDA1341_DEBUG 1*/ +#ifdef S3C24XX_UDA1341_DEBUG +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x) +#else +#define DBG(x...) +#endif + +static struct clk *xtal; +static struct clk *pclk; + +static unsigned long s3c24xx_uda1341_calc_error(unsigned long rate, + unsigned long clk_rate, + unsigned int div, + unsigned int fs) +{ + long err; + + err = clk_rate / (div * fs); + err -= rate; + if (err < 0) + err = -err; + DBG("rate %lu clk %lu div %u fs %u err %ld\n", + rate, clk_rate, div, fs, err); + return err; +} + +static int s3c24xx_uda1341_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + unsigned int div = 0, cdiv; + int ret = 0; + int clk_source, fs_mode; + unsigned long mpllin_rate = clk_get_rate(xtal); + unsigned long pclk_rate = clk_get_rate(pclk); + unsigned long rate = params_rate(params); + unsigned long err, cerr; + + DBG("mpllin %ld pclk %ld rate %lu\n", mpllin_rate, pclk_rate, rate); + + div = pclk_rate / (256 * rate); + if (div == 0) + div = 1; + if (div > 32) + div = 32; + err = s3c24xx_uda1341_calc_error(rate, pclk_rate, div, 256); + fs_mode = S3C2410_IISMOD_256FS; + clk_source = S3C24XX_CLKSRC_PCLK; + + if (div < 32) { + cdiv = div + 1; + cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 256); + if (cerr < err) { + err = cerr; + div = cdiv; + } + } + + cdiv = pclk_rate / (384 * rate); + if (cdiv == 0) + cdiv = 1; + if (cdiv > 32) + cdiv = 32; + cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 384); + if (cerr < err) { + err = cerr; + div = cdiv; + fs_mode = S3C2410_IISMOD_384FS; + } + + if (cdiv < 32) { + cdiv = cdiv + 1; + cerr = s3c24xx_uda1341_calc_error(rate, pclk_rate, cdiv, 384); + if (cerr < err) { + err = cerr; + div = cdiv; + fs_mode = S3C2410_IISMOD_384FS; + } + } + + cerr = s3c24xx_uda1341_calc_error(rate, mpllin_rate, 1, 256); + if (cerr < err) { + err = cerr; + fs_mode = S3C2410_IISMOD_256FS; + clk_source = S3C24XX_CLKSRC_MPLL; + } + + cerr = s3c24xx_uda1341_calc_error(rate, mpllin_rate, 1, 384); + if (cerr < err) { + err = cerr; + fs_mode = S3C2410_IISMOD_384FS; + clk_source = S3C24XX_CLKSRC_MPLL; + } + + clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; + div = div - 1; + DBG("Will use: %s %s %d sysclk %d err %ld\n", + fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", + clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", + div, clk, err); + + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, clk_source , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + fs_mode); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops s3c24xx_uda1341_ops = { + .hw_params = s3c24xx_uda1341_hw_params, +}; + +static struct snd_soc_dai_link s3c24xx_uda1341_dai_link = { + .name = "UDA1341", + .stream_name = "UDA1341", + .codec_dai = &uda1341_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &s3c24xx_uda1341_ops, +}; + +static struct snd_soc_machine snd_soc_machine_s3c24xx_uda1341 = { + .name = "S3C24XX_UDA1341", + .dai_link = &s3c24xx_uda1341_dai_link, + .num_links = 1, +}; + +static struct s3c24xx_uda1341_platform_data *s3c24xx_uda1341_l3_pins; + +static void setdat(struct uda1341_platform_data *p, int v) +{ + s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_data, v > 0); + s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_data, + S3C2410_GPIO_OUTPUT); +} + +static void setclk(struct uda1341_platform_data *p, int v) +{ + s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_clk, v > 0); + s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_clk, + S3C2410_GPIO_OUTPUT); +} + +static void setmode(struct uda1341_platform_data *p, int v) +{ + s3c2410_gpio_setpin(s3c24xx_uda1341_l3_pins->l3_mode, v > 0); + s3c2410_gpio_cfgpin(s3c24xx_uda1341_l3_pins->l3_mode, + S3C2410_GPIO_OUTPUT); +} + +static void s3c24xx_uda1341_power(int en) +{ + if (s3c24xx_uda1341_l3_pins->power) + s3c24xx_uda1341_l3_pins->power(en); +} + +static struct uda1341_platform_data s3c24xx_uda1341 = { + .setdat = setdat, + .setclk = setclk, + .setmode = setmode, + .data_hold = 1, + .data_setup = 1, + .clock_high = 1, + .mode_hold = 1, + .mode = 1, + .mode_setup = 1, + .power = s3c24xx_uda1341_power, +}; + +static struct snd_soc_device s3c24xx_uda1341_snd_devdata = { + .machine = &snd_soc_machine_s3c24xx_uda1341, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_uda1341, + .codec_data = &s3c24xx_uda1341, +}; + +static struct platform_device *s3c24xx_uda1341_snd_device; + +static int s3c24xx_uda1341_probe(struct platform_device *pdev) +{ + int ret; + + printk(KERN_INFO "S3C24XX_UDA1341 SoC Audio driver\n"); + + s3c24xx_uda1341_l3_pins = pdev->dev.platform_data; + if (s3c24xx_uda1341_l3_pins == NULL) { + printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: " + "unable to find platform data\n"); + return -ENODEV; + } + + s3c24xx_uda1341_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_uda1341_snd_device) { + printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: " + "Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(s3c24xx_uda1341_snd_device, + &s3c24xx_uda1341_snd_devdata); + s3c24xx_uda1341_snd_devdata.dev = &s3c24xx_uda1341_snd_device->dev; + ret = platform_device_add(s3c24xx_uda1341_snd_device); + + if (ret) { + printk(KERN_ERR "S3C24XX_UDA1341 SoC Audio: Unable to add\n"); + platform_device_put(s3c24xx_uda1341_snd_device); + } + + xtal = clk_get(NULL, "xtal"); + pclk = clk_get(NULL, "pclk"); + + return ret; +} + +static int s3c24xx_uda1341_remove(struct platform_device *pdev) +{ + platform_device_unregister(s3c24xx_uda1341_snd_device); + clk_put(xtal); + clk_put(pclk); + return 0; +} + +static struct platform_driver s3c24xx_uda1341_driver = { + .probe = s3c24xx_uda1341_probe, + .remove = s3c24xx_uda1341_remove, + .driver = { + .name = "s3c24xx_uda1341", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c24xx_uda1341_init(void) +{ + return platform_driver_register(&s3c24xx_uda1341_driver); +} + +static void __exit s3c24xx_uda1341_exit(void) +{ + platform_driver_unregister(&s3c24xx_uda1341_driver); +} + + +module_init(s3c24xx_uda1341_init); +module_exit(s3c24xx_uda1341_exit); + +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); +MODULE_DESCRIPTION("S3C24XX_UDA1341 ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); -- 1.4.4.4

6 17

Re: [alsa-devel] [Intel-gfx] G45 user experience
by Wu Fengguang 12 Nov '08

12 Nov '08

On Tue, Nov 11, 2008 at 05:16:18PM +0800, Fu Michael wrote: > Shane W wrote: >> On Tue, Nov 11, 2008 at 10:32:40AM +0800, Fu Michael wrote: >> >>> Do you have any HDMI sink device that can support 5.1 or 7.1 audio? >>> I'm looking for those kind of devices as well... >>> >> >> I'm not really sure what you mean by HDMI sink but my >> receiver which connects to the G45 does support 5.1 and 7.1 >> HDMI decoding along with TRUEHD, DTSHD and DSD so I can >> certainly test anything that comes down the pipe as far as >> patches go. >> > Shane, would you please tell me the model of this HDMI receiver? Or preferably, could you send us your dmesg? I'm particularly interested in the ELD information inside it. Thanks, Fengguang

1 0

[alsa-devel] alsa-driver 1.0.18 doesn't built on PPC 2.6.27 and 2.6.28
by Elimar Riesebieter 11 Nov '08

11 Nov '08

Hi all, please note the following error: /usr/bin/make -C /usr/src/linux-2.6.27.2 SUBDIRS=/usr/src/modules/alsa-driver CPP="gcc-4.3 -E" CC="gcc-4.3" modules make[4]: Entering directory `/usr/src/linux-2.6.27.2' CC [M] /usr/src/modules/alsa-driver/acore/hwdep.o In file included from /usr/src/linux-2.6.27.2/arch/powerpc/include/asm/system.h:528, from include/linux/list.h:7, from include/linux/module.h:9, from /usr/src/modules/alsa-driver/include/adriver.h:26, from /usr/src/modules/alsa-driver/acore/hwdep.c:1: include/asm-generic/cmpxchg-local.h: In function '__cmpxchg_local_generic': include/asm-generic/cmpxchg-local.h:23: error: implicit declaration of function 'local_irq_save' include/asm-generic/cmpxchg-local.h:44: error: implicit declaration of function 'local_irq_restore' make[6]: *** [/usr/src/modules/alsa-driver/acore/hwdep.o] Fehler 1 make[5]: *** [/usr/src/modules/alsa-driver/acore] Fehler 2 make[4]: *** [_module_/usr/src/modules/alsa-driver] Fehler 2 make[4]: Leaving directory `/usr/src/linux-2.6.27.2' make[3]: *** [compile] Fehler 2 make[3]: Leaving directory `/usr/src/modules/alsa-driver' make[2]: *** [build-stamp] Fehler 2 make[2]: Leaving directory `/usr/src/modules/alsa-driver' make[1]: *** [kdist_image] Fehler 2 make[1]: Leaving directory `/usr/src/modules/alsa-driver' Module /usr/src/modules/alsa-driver failed. Same on 2.6.28-rc3 BTW, alsa-driver 1.0.18 builds fine on PPC 2.6.26. The PPC divers build fine out of vanilla trees (1.6.26/1.0.17 and 2.6.28/1.0.18-rc3) Thanks for cooperation Elimar -- "Talking much about oneself can also be a means to conceal oneself." -Friedrich Nietzsche

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[alsa-devel] arecord read error, input/output error in embedded linux project
by Robert Schmit 11 Nov '08

11 Nov '08

I am working on resurrecting an embedded linux project started by someone else. The project is using an AT91SAM9260 processor on an AT91SAM9260-EK development board and I am currently trying to get audio playback and capture working through an external W6811 codec chip. The previous developer had the audio working both ways but unfortunately I can only get playback to work but not capture. During a capture operation, I can verify on a scope that the proper clocks and frame syncs are being delivered to the W6811 chip and furthermore I can see data on the PCM output line of the chip (which is of course connected to an input pin on the AT91SAM9260 processor). So I'm almost certain that the problem is not hardware. A machine/codec driver for the W6811 chip was written by the previous developer and is known to be correct and properly included in the kernel. The ALSA version is 1.0.14 and is compiled into a 2.6.22.1 kernel. I am currently using the default alsa.conf file and am not using ~/.asoundrc or /etc/asound.conf. I am testing the audio using aplay and arecord. Specifically, arecord is failing, after 10 seconds, as follows: /usr/bin $ arecord -traw -fS8 -r8000 ulaw2.raw Recording raw data 'ulaw2.raw' : Signed 8 bit, Rate 8000 Hz, Mono arecord: pcm_read:1346: read error: Input/output error An strace of the arecord execution provides the following at the point of error. The ioctl() call is what hangs for 10 seconds, then the error message is printed and the program immediately exits. open("test_sample.raw", O_WRONLY|O_CREAT|O_LARGEFILE, 0644) = 3 ioctl(4, 0x800c4151, 0xbef15c24) = -1 EIO (Input/output error) write(2, "arecord", 7arecord) = 7 write(2, ": ", 2: ) = 2 write(2, "pcm_read", 8pcm_read) = 8 write(2, ":", 1:) = 1 write(2, "1346", 41346) = 4 write(2, ": ", 2: ) = 2 write(2, "read error: ", 12read error: ) = 12 write(2, "Input/output error", 18Input/output error) = 18 write(2, "\n", 1 ) = 1 exit(1) = ? Process 844 detached Is it possible that there is an extra step towards properly configuring a capture channel/stream that would explain why playback works but capture does not?? If anybody has any ideas, I would be very grateful as I have been stuck on this for days now. I'm not sure what information would be helpful, but I will try to describe some details of my sound configuration here. THANKS!! /dev/snd $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: GRHEKW6811 [GRH_EK_W6811], device 0: W6811 W6811-PCM-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 /dev/snd $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: GRHEKW6811 [GRH_EK_W6811], device 0: W6811 W6811-PCM-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 /dev/snd $ ls -l crw-rw-rw- 1 root root 116, 0 Dec 31 17:27 controlC0 crw-rw-rw- 1 root root 116, 24 Dec 31 17:29 pcmC0D0c crw-rw-rw- 1 root root 116, 16 Dec 31 17:27 pcmC0D0p crw-rw-rw- 1 root root 116, 33 Dec 31 17:51 timer /proc/asound $ ls GRHEKW6811 cards pcm version card0 devices timers /proc/asound $ cat devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer /proc/asound $ cat pcm 00-00: W6811 W6811-PCM-0 : : playback 1 : capture 1 /proc/asound $ cat cards 0 [GRHEKW6811 ]: W6811 - GRH_EK_W6811 GRH_EK_W6811 (W6811) /proc/asound $ cat timers G0: system timer : 10000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE

1 0

[PATCH] ALSA: add /sys/class/sound/card#/id (r/w) and card#/number (r/o) files
by Jaroslav Kysela 11 Nov '08

11 Nov '08

For udev, we need a way to rename soundcard names. The soundcard numbers (indexes) are hardwired but we have a text identification which can be changed at run-time. The ALSA user space tools already allow using of this text identification. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/init.c | 66 ++++++++++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 65 insertions(+), 1 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index b47ff8b..1426abd 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -533,6 +533,64 @@ static void choose_default_id(struct snd_card *card) } } +#ifndef CONFIG_SYSFS_DEPRECATED +static ssize_t +card_id_show_attr(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_card *card = dev_get_drvdata(dev); + return snprintf(buf, PAGE_SIZE, "%s\n", card ? card->id : "(null)"); +} + +static ssize_t +card_id_store_attr(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_card *card = dev_get_drvdata(dev); + char buf1[sizeof(card->id)]; + size_t copy = count > sizeof(card->id) - 1 ? + sizeof(card->id) - 1 : count; + size_t idx; + int c; + + for (idx = 0; idx < copy; idx++) { + c = buf[idx]; + if (!isalnum(c) && c != '_' && c != '-') + return -EINVAL; + } + memcpy(buf1, buf, copy); + buf1[copy] = '\0'; + mutex_lock(&snd_card_mutex); + if (!snd_info_check_reserved_words(buf1)) { + __exist: + mutex_unlock(&snd_card_mutex); + return -EEXIST; + } + for (idx = 0; idx < snd_ecards_limit; idx++) { + if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) + goto __exist; + } + strcpy(card->id, buf1); + mutex_unlock(&snd_card_mutex); + + return count; +} + +static struct device_attribute card_id_attrs = + __ATTR(id, S_IRUGO | S_IWUSR, card_id_show_attr, card_id_store_attr); + +static ssize_t +card_number_show_attr(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_card *card = dev_get_drvdata(dev); + return snprintf(buf, PAGE_SIZE, "%i\n", card ? card->number : -1); +} + +static struct device_attribute card_number_attrs = + __ATTR(number, S_IRUGO, card_number_show_attr, NULL); +#endif /* CONFIG_SYSFS_DEPRECATED */ + /** * snd_card_register - register the soundcard * @card: soundcard structure @@ -553,7 +611,7 @@ int snd_card_register(struct snd_card *card) #ifndef CONFIG_SYSFS_DEPRECATED if (!card->card_dev) { card->card_dev = device_create(sound_class, card->dev, - MKDEV(0, 0), NULL, + MKDEV(0, 0), card, "card%i", card->number); if (IS_ERR(card->card_dev)) card->card_dev = NULL; @@ -576,6 +634,12 @@ int snd_card_register(struct snd_card *card) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER); #endif +#ifndef CONFIG_SYSFS_DEPRECATED + if (card->card_dev) { + device_create_file(card->card_dev, &card_id_attrs); + device_create_file(card->card_dev, &card_number_attrs); + } +#endif return 0; } -- 1.6.0.1.217.gb4928 ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

1 0

[alsa-devel] [RFC][PATCH] pxa2xx-ac97: switch AC unit to correct state before probing
by Dmitry Baryshkov 11 Nov '08

11 Nov '08

Hi, I've stumbled upon a problem with my tosa unit. The bootloader leaves ac97 in some partially enabled state, so when request_irq enables the AC97 irq, I see a unending storm of irq requests. I've come with the following patch (against 2.6.27, if the idea is OK, I'll rebase against 2.6.28-rc). From: Dmitry Baryshkov <dbaryshkov(a)gmail.com> If AC97 unit is in partially enabled state, early request_irq can trigger IRQ storm or even full hang up. Workaround this by forcibly switching ACLINK off at the start of the probe. Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/pxa2xx-ac97.c | 30 +++++++++++++++++------------- 1 files changed, 17 insertions(+), 13 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index d94a495..259c6e1 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -315,10 +315,6 @@ static int pxa2xx_ac97_probe(struct platform_device *pdev, { int ret; - ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL); - if (ret < 0) - goto err; - pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); @@ -331,28 +327,36 @@ static int pxa2xx_ac97_probe(struct platform_device *pdev, if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); ac97conf_clk = NULL; - goto err_irq; + goto err_conf; } #endif + GCR = GCR_ACLINK_OFF; + ac97_clk = clk_get(&pdev->dev, "AC97CLK"); if (IS_ERR(ac97_clk)) { ret = PTR_ERR(ac97_clk); ac97_clk = NULL; - goto err_irq; + goto err_clk; } clk_enable(ac97_clk); + + ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL); + if (ret < 0) + goto err_irq; + return 0; - err_irq: +err_irq: GCR |= GCR_ACLINK_OFF; + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; +err_clk: #ifdef CONFIG_PXA27x - if (ac97conf_clk) { - clk_put(ac97conf_clk); - ac97conf_clk = NULL; - } + clk_put(ac97conf_clk); + ac97conf_clk = NULL; +err_conf: #endif - free_irq(IRQ_AC97, NULL); - err: return ret; } -- 1.5.6.5 -- With best wishes Dmitry

2 3

[alsa-devel] [PATCH] Make the HP EliteBook 8530p use AD1884A model 'laptop' by default.
by Travis Place 11 Nov '08

11 Nov '08

Changelog: Added a QUIRK to patch_analog.c for the HP Elitebook 8530p (IDs 0x103c:0x30e7) to use AD1884A model 'laptop' by default. Playback and Capture confirmed working. Signed-off-by: Travis Place

2 1

Re: [alsa-devel] [Intel-gfx] G45 user experience
by Wu Fengguang 11 Nov '08

11 Nov '08

On Mon, Nov 10, 2008 at 08:15:01PM -0800, Shane W wrote: > On Tue, Nov 11, 2008 at 10:32:40AM +0800, Fu Michael wrote: > > Do you have any HDMI sink device that can support 5.1 or 7.1 audio? I'm > > looking for those kind of devices as well... > > I'm not really sure what you mean by HDMI sink but my > receiver which connects to the G45 does support 5.1 and 7.1 > HDMI decoding along with TRUEHD, DTSHD and DSD so I can > certainly test anything that comes down the pipe as far as > patches go. Wow, thank you very much! > By the way, had a bit more time to test out these HDMI sync > xorg patches. Under mplayer, after playing audio for about > 10 or so minutes with a video stream, the audio starts to > drop out and then stops altogether. Stopping and starting Do you mean the volume goes down gradually? > mplayer doesn't fix it, stopping mplayer and playing a wav > via aplay gives silence. Stopping Xorg does correct it and > once Xorg is restarted, it'll run with video for another 10 > or so minutes and stop again. This is using Saturday's GIT OK, then restarting Xorg fixed it temporarily. I wonder if restarting the monitor restores the sound as well? > on ALSA, Xorg and Intel with the IALSA Intel hdmi patch, > and the HDMI patches for Xorg and the Intel driver applied. Thank you, Fengguang

1 0

[alsa-devel] HDA-Intel - Patch to support detecting HD Audio devices with PCI class code
by Yang, Libin 11 Nov '08

11 Nov '08

Sorry, adding the title... Thanks & Regards, Libin > -----Original Message----- > From: Yang, Libin > Sent: Tuesday, November 11, 2008 10:43 AM > To: 'Takashi Iwai' > Cc: 'alsa-devel(a)alsa-project.org' > Subject: > > Hi Takashi, > > This patch is to support detecting new AMD HD Audio devices with HDA PCI > class code. Would you please review it? Thanks a lot! > -------------------------------------------------------------------- > > Signed-off-by: Libin Yang <libin.yang(a)amd.com> > > --- alsa-driver-1.0.18.orig/alsa-kernel/pci/hda/hda_intel.c 2008-10-29 > 20:41:35.000000000 +0800 > +++ alsa-driver-1.0.18/alsa-kernel/pci/hda/hda_intel.c 2008-11-11 > 18:33:14.000000000 +0800 > @@ -291,6 +291,7 @@ > /* Define VIA HD Audio Device ID*/ > #define VIA_HDAC_DEVICE_ID 0x3288 > > +#define PCI_CLASS_MULTIMEDIA_HDA 0x040300 > > /* > */ > @@ -410,6 +411,7 @@ > AZX_DRIVER_ULI, > AZX_DRIVER_NVIDIA, > AZX_DRIVER_TERA, > + AZX_DRIVER_AMD_AUTO, > AZX_NUM_DRIVERS, /* keep this as last entry */ > }; > > @@ -423,6 +425,7 @@ > [AZX_DRIVER_ULI] = "HDA ULI M5461", > [AZX_DRIVER_NVIDIA] = "HDA NVidia", > [AZX_DRIVER_TERA] = "HDA Teradici", > + [AZX_DRIVER_AMD_AUTO] = "HDA AMD", > }; > > /* > @@ -1198,6 +1201,7 @@ > static unsigned int azx_default_codecs[AZX_NUM_DRIVERS] __devinitdata = { > [AZX_DRIVER_ICH] = 3, > [AZX_DRIVER_ATI] = 3, > + [AZX_DRIVER_AMD_AUTO] = 3, > }; > > static int __devinit azx_codec_create(struct azx *chip, const char *model, > @@ -2146,6 +2150,7 @@ > chip->playback_streams = ULI_NUM_PLAYBACK; > chip->capture_streams = ULI_NUM_CAPTURE; > break; > + case AZX_DRIVER_AMD_AUTO: > case AZX_DRIVER_ATIHDMI: > chip->playback_streams = ATIHDMI_NUM_PLAYBACK; > chip->capture_streams = ATIHDMI_NUM_CAPTURE; > @@ -2373,6 +2378,9 @@ > { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, > /* Teradici */ > { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, > + /* AMD Generic, PCI class code and Vendor ID for HD Audio */ > + { PCI_VENDOR_ID_ATI, PCI_ANY_ID, PCI_ANY_ID, PCI_ANY_ID, > + PCI_CLASS_MULTIMEDIA_HDA, 0xffffff, AZX_DRIVER_AMD_AUTO }, > { 0, } > }; > MODULE_DEVICE_TABLE(pci, azx_ids); > > Thanks & Regards, > Libin

1 0

Re: [alsa-devel] Bugs on aspire one A150
by Maxim Levitsky 10 Nov '08

10 Nov '08

Bob Copeland wrote: > On Fri, Nov 7, 2008 at 10:26 PM, Maxim Levitsky <maximlevitsky(a)gmail.com> wrote: >> ** 2 - wireless: not to mention the fact that ath5k wasn't installed by >> default in ubuntu... >> wireless more or less works, but kernel log is full of backtraces. > > As you've seen, these should at least be gone now... > >> Was able to connect to my WPA2 access point. >> Sometimes wireless fails completely, especially after suspend to ram. > > Did you get noise calibration failures in dmesg at this point? Felix posted > a patch recently that handles them better. Yep, but they seems to be gone in latest -git too. Wireless work fine now, it seems like my iwl3945. > >> Advanced features like monitor/injection work, but when I changed the card's >> mac address it stopped working. >> I also noticed that if I then start airodump, then wireless works with new >> mac. Why it doesn't accept new mac?, can this be fixed? > >> ** 4 - wireless led doesn't work. >> ath5k devs, can you fix this? > > Currently all the LED code we have in there is for setting a particular gpio > for particular laptops (and they differ between models). We only have quirks > for IBM and HP, none for Acer. Also, the legacy-hal seems to have a different > strategy for 2425 chips. I can try to hack up some code for you to test. > I have seen on the web that madwifi did support the led with some gpio settings sysctl -w dev.wifi0.ledpin=3 sysctl -w dev.wifi0.softled=1 from https://help.ubuntu.com/community/AspireOne Best regards, Maxim Levitsky

1 0

[alsa-devel] MS-7350 Realtek ALC888
by Philipp Matthias Hahn 10 Nov '08

10 Nov '08

Hello! I have an MSI MS-7350 motherborad with an onboard Realtek ALC888 HDA codec. I need to specify model=6stack-dig to get the internal CD connector to show up, else automatic BIOS scanning doesn't find it. Might be that you want to add the appropriate DMI/PCI quirk stuff, so other users get the benefit of automatic detection. $ lspci -s 0:10.1 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) $ lspci -s 0:10.1 -n -vvv 00:10.1 0403: 10de:026c (rev a2) Subsystem: 1462:7350 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 (500ns min, 1250ns max) Interrupt: pin B routed to IRQ 22 Region 0: Memory at f9ff8000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel BYtE Philipp -- / / (_)__ __ ____ __ Philipp Hahn / /__/ / _ \/ // /\ \/ / /____/_/_//_/\_,_/ /_/\_\ pmhahn(a)titan.lahn.de

1 0

[alsa-devel] [PATCH] ALSA SOC driver for s3c24xx: 8 bit sound fix
by Christian Pellegrin 10 Nov '08

10 Nov '08

fixes playing/recording of 8 bit audio files. Generated on 20081108 against v2.6.27 Signed-off-by: Christian Pellegrin <chripell(a)fsfe.org> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ba4476b..c18977b 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -261,10 +261,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: + iismod &= ~S3C2410_IISMOD_16BIT; + ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; + ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->dma_size = 2; break; + default: + return -EINVAL; } writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); -- 1.4.4.4

4 4

[alsa-devel] [PATCH] gusextreme: Fix build errors
by Ville Syrjala 10 Nov '08

10 Nov '08

gusextreme depends on opl3 support. Add the approriate select to Kconfig. Also remove the unnecessary hwdep select. Relevant build errors: ERROR: "snd_opl3_hwdep_new" [sound/isa/gus/snd-gusextreme.ko] undefined! ERROR: "snd_opl3_create" [sound/isa/gus/snd-gusextreme.ko] undefined! Signed-off-by: Ville Syrjala <syrjala(a)sci.fi> --- sound/isa/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 660beb4..ce0aa04 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -211,7 +211,7 @@ config SND_GUSCLASSIC config SND_GUSEXTREME tristate "Gravis UltraSound Extreme" - select SND_HWDEP + select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help -- 1.5.6.4

2 1

[alsa-devel] [PATCH] Renamed the struct soc_enum field .mask to .count. The field refers to the number of enumerable entries instead of being a bit mask. This change makes the interface a more clear.
by Marc Singer 10 Nov '08

10 Nov '08

Signed-off-by: Marc Singer <elf(a)buici.com> --- include/sound/soc-codec.h | 14 +++++++------- sound/soc/soc-core.c | 20 ++++++++++---------- sound/soc/soc-dapm.c | 14 +++++++------- 3 files changed, 24 insertions(+), 24 deletions(-) diff --git a/include/sound/soc-codec.h b/include/sound/soc-codec.h index d8595ef..d59015a 100644 --- a/include/sound/soc-codec.h +++ b/include/sound/soc-codec.h @@ -79,13 +79,13 @@ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (reg_left) | ((shift) << 8) | \ ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } -#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ +#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xcount, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .texts = xtexts } -#define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) \ - SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xtexts) -#define SOC_ENUM_SINGLE_EXT(xmask, xtexts) \ -{ .mask = xmask, .texts = xtexts } + .count = xcount, .texts = xtexts } +#define SOC_ENUM_SINGLE(xreg, xshift, xcount, xtexts) \ + SOC_ENUM_DOUBLE(xreg, xshift, xshift, xcount, xtexts) +#define SOC_ENUM_SINGLE_EXT(xcount, xtexts) \ +{ .count = xcount, .texts = xtexts } #define SOC_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ .info = snd_soc_info_enum_double, \ @@ -180,7 +180,7 @@ struct soc_enum { unsigned short reg2; unsigned char shift_l; unsigned char shift_r; - unsigned int mask; + unsigned int count; /* Count of options */ const char **texts; void *dapm; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 706f98d..b8efb4b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1262,10 +1262,10 @@ int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = e->shift_l == e->shift_r ? 1 : 2; - uinfo->value.enumerated.items = e->mask; + uinfo->value.enumerated.items = e->count; - if (uinfo->value.enumerated.item > e->mask - 1) - uinfo->value.enumerated.item = e->mask - 1; + if (uinfo->value.enumerated.item > e->count - 1) + uinfo->value.enumerated.item = e->count - 1; strcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item]); return 0; @@ -1288,7 +1288,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->count; bitmask <<= 1) ; val = snd_soc_read(codec, e->reg); ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); @@ -1317,14 +1317,14 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, unsigned short val; unsigned short mask, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->count; bitmask <<= 1) ; - if (ucontrol->value.enumerated.item[0] > e->mask - 1) + if (ucontrol->value.enumerated.item[0] > e->count - 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; mask = (bitmask - 1) << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->mask - 1) + if (ucontrol->value.enumerated.item[1] > e->count - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; mask |= (bitmask - 1) << e->shift_r; @@ -1351,10 +1351,10 @@ int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = e->mask; + uinfo->value.enumerated.items = e->count; - if (uinfo->value.enumerated.item > e->mask - 1) - uinfo->value.enumerated.item = e->mask - 1; + if (uinfo->value.enumerated.item > e->count - 1) + uinfo->value.enumerated.item = e->count - 1; strcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item]); return 0; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f203693..5b83329 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -119,13 +119,13 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value; int val, item, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->count; bitmask <<= 1) ; val = snd_soc_read(w->codec, e->reg); item = (val >> e->shift_l) & (bitmask - 1); p->connect = 0; - for (i = 0; i < e->mask; i++) { + for (i = 0; i < e->count; i++) { if (!(strcmp(p->name, e->texts[i])) && item == i) p->connect = 1; } @@ -162,7 +162,7 @@ static int dapm_connect_mux(struct snd_soc_card *soc_card, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int i; - for (i = 0; i < e->mask; i++) { + for (i = 0; i < e->count; i++) { if (!(strcmp(control_name, e->texts[i]))) { list_add(&path->list, &soc_card->dapm_paths); list_add(&path->list_sink, &dest->sources); @@ -1197,7 +1197,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->count; bitmask <<= 1) ; val = snd_soc_read(widget->codec, e->reg); ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); @@ -1227,15 +1227,15 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, unsigned short mask, bitmask; int ret = 0; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->count; bitmask <<= 1) ; - if (ucontrol->value.enumerated.item[0] > e->mask - 1) + if (ucontrol->value.enumerated.item[0] > e->count - 1) return -EINVAL; mux = ucontrol->value.enumerated.item[0]; val = mux << e->shift_l; mask = (bitmask - 1) << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->mask - 1) + if (ucontrol->value.enumerated.item[1] > e->count - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; mask |= (bitmask - 1) << e->shift_r; -- 1.5.6.5

2 1

[alsa-devel] Non-functional Altec Lansing FX5051 USB-soundcard + speakers
by Jeremy Nickurak 10 Nov '08

10 Nov '08

About 14 months ago I purchased a nice new ~$300 set of altec lansing 5.1 speakers, which include a USB-sound card. Since that time they've largely gone unused, or at least not properly used. I brought this up on the mailing list at the time, but didn't get much response, so I thought I'd poke my head up again and see if anybody has any insight. First off, the set works correctly, out-of-the-box under windows, without having to install any special drivers or software at all. Under linux, plugging them in instantly freezes the usb sound card up, and locks it at 100% volume, in such a way that there's absolutely no way to turn the volume down: not from alsamixer, and not from the speaker's hardware control dial. Sound does play, but again, it's unbearably loud, to the point where I don't do it for risk of damaging my speakers and/or ears. I filed the bug away on the bug-tracker, but haven't heard much out of it since. The bug contains my lsusb -v information, the "contents" output from amixer for this card, and some preliminary results showing some ability to tweak the volume with amixer, but not reliably. Any suggestions are welcome. I'm happy to experiment with this set more if it can help, run experimental code, etc. Heck, I'd go at the thing with a screwdriver for chipset information if I know what I was looking for or thought it could help.

2 1

[alsa-devel] [PATCH] ALSA: HDSP: check for io box before uploading firmware
by Tim Blechmann 10 Nov '08

10 Nov '08

currently the hdsp driver tries to upload the firmware, even if the io box is not connected. this patch adds a check for the io box before trying to upload the firmware. thus instead of messages complaining about the fifo status and firmware loading failure, the driver gives a message that no multiface or digiface is connected. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/rme9652/hdsp.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index d723543..cf9eb41 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5045,6 +5045,9 @@ static int __devinit snd_hdsp_create(struct snd_card *card, /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ ssleep(2); + if ((err = hdsp_check_for_iobox(hdsp)) < 0) + return err; + if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { #ifdef HDSP_FW_LOADER if ((err = hdsp_request_fw_loader(hdsp)) < 0) @@ -5057,7 +5060,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, /* init is complete, we return */ return 0; #endif - /* no iobox connected, we defer initialization */ + /* we defer initialization */ snd_printk(KERN_INFO "Hammerfall-DSP: card initialization pending : waiting for firmware\n"); if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0) return err; -- 1.5.6.3

3 2

[alsa-devel] Stuttering 100mA and 500mA MaxPower usb audio investigation on bus powered and external powererd usb hubs
by Jelle de Jong 10 Nov '08

10 Nov '08

Hello everybody, I have been testing several scenarios to find some glues why some usb audio devices did not fully work and gave stuttering sounds I found the following: usb devices define a max power attribute during there enumeration process. I have some usb devices that have this at 500mA and some that have this on 100mA. I also have some usb hubs that have external extra power supply and some that are only a hub with bus power. Here it comes: Make sure the 4 port usb hub is full with for example: - usb keyboard (100mA) - usb mouse (100mA) - usb audio (100mA or 500mA) - usb stick (200mA/500mA) - usb hub with external power (500mA MaxPower) (AC adapter delivers 1A) - usb hub without external power (100mA MaxPower) Make sure to use audio with several tones low tones (most dvds will do) When connecting a 100mA usb audio devices on a hub with external power it will give stuttering sound!! (most usb audo devices are 100mA and most hubs are external powered due to more then one usb stick) When connecting a 500mA usb audio device on a hub with external power it will work perfectly. When connecting a 100mA or a 500mA usb audio device on a hub without external power it will work perfectly. I have the idea there is something wrong with the power distribution system when using external powered usb hubs... Any ideas how to correctly address this problem/bug? Best regards, Jelle

2 2

[alsa-devel] Backported sbxfi driver (UNTESTED!)
by Romain 10 Nov '08

10 Nov '08

Hi, I've tested the sbxfi driver and it works great on my setup. Had little problems before getting it running : i had to comment out the snd_printk stuff, otherwise the module wouldn't load (snd_printk* symbols not found, in dmesg) This was probably not the cleanest way to deal but it's just debug messages and i'm new wrt kernel driver hacking. I also had to disable my onboard audio in BIOS, otherwise the X-Fi would not get picked up. i'm using Fedora 9 with Linux 2.6.26.6-79.fc9.i686 kernel (standard supply from fedora repos),and pulseaudio I'm using with standard 2.1 jack output, i do not yet own home cinema speakers so i couldn't test all the cabling options. Thanks for this great work, Best Regards, Romain.

1 0

[alsa-devel] [PATCH] ALSA: hda - Add another HP model (6730s) for AD1884A.
by Michel Marti 09 Nov '08

09 Nov '08

Signed-off-by: Michel Marti <mma(a)objectxp.com> --- sound/pci/hda/patch_analog.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d3fd432..ee62e31 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3861,6 +3861,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), {} }; -- 1.5.6.5

2 2

[alsa-devel] [PATCH] ALSA: hdsp: check for iobox and upload firmware during ioctl
by Tim Blechmann 09 Nov '08

09 Nov '08

currently, the error message when trying to run hdspmixer or hdspconf if the breakout box is not connected is somehow misleading, since it asks the user to upload the firmware. this patch adds a test, whether the breakout box is connected and tries to upload the firmware in the case, that it is not present, e.g. because of power failures of the breakout box. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/rme9652/hdsp.c | 17 +++++++++++++---- 1 files changed, 13 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cf9eb41..5bd0be6 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4548,11 +4548,18 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne { struct hdsp *hdsp = (struct hdsp *)hw->private_data; void __user *argp = (void __user *)arg; + int err; switch (cmd) { case SNDRV_HDSP_IOCTL_GET_PEAK_RMS: { struct hdsp_peak_rms __user *peak_rms = (struct hdsp_peak_rms __user *)arg; + if ((err = hdsp_check_for_iobox (hdsp)) < 0) + return err; + + if ((err = hdsp_check_for_firmware(hdsp, 1)) < 0) + return err; + if (!(hdsp->state & HDSP_FirmwareLoaded)) { snd_printk(KERN_ERR "Hammerfall-DSP: firmware needs to be uploaded to the card.\n"); return -EINVAL; @@ -4572,10 +4579,12 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne unsigned long flags; int i; - if (!(hdsp->state & HDSP_FirmwareLoaded)) { - snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n"); - return -EINVAL; - } + if ((err = hdsp_check_for_iobox (hdsp)) < 0) + return err; + + if ((err = hdsp_check_for_firmware(hdsp, 1)) < 0) + return err; + spin_lock_irqsave(&hdsp->lock, flags); info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp); info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp); -- 1.5.6.3

2 1

[alsa-devel] Intel G45 HDMI audio - missing samples
by Shane W 08 Nov '08

08 Nov '08

Hi, One issue with Intel audio over HDMI on a DG45-ID motherboard. When playing a short sample via HDMI, it misses the first quarter second or so of the audio. This is particularly noticable when using text to speech as the first word is lost. I can get around this by having a seperate program hold the audio device open while using text to speech, that seems to get the whole sample but it's a cludgy workaround. The same issue does not exist on the optical port. Best, Shane

1 0

[alsa-devel] Intel HDA Headphones vs. Speaker Mute
by Richard Spindler 07 Nov '08

07 Nov '08

Hi, the builtin sound card in my laptop does not behave as it should, when running it with the ALSA system. While there are separate mixer controls for headphones and speakers, the Master control levels both, speakers and headphones simultaneously, while the headphone control does not act at all. Also in my opinion the speakers should mute automatically as soon as I plug in the headphones, this does not happen too. I am running Ubuntu 8.04.1: $ uname -a Linux monostatos 2.6.24-19-generic #1 SMP Wed Aug 20 22:56:21 UTC 2008 i686 GNU/Linux $ lspci -nn |grep Audio 00:1b.0 Audio device [0403]: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller [8086:27d8] (rev 02) If you need any more information about that machine, or if there is anything I can do to help fix that problem, do not hesitate to ask. Cheers -Richard PS.: I also published a lengthy rant, and some ideas about that problem at http://propirate.net/oracle/archives/2008/11/05/alsa-headaches-erm-headphon…

2 1

[alsa-devel] Hints for usb audio patch
by LCID Fire 07 Nov '08

07 Nov '08

Hi. I have a patch lying around for some month now and would need some hints how to correctly implement it. 1. What's the correct way to save the state of a special feature? There is a switch which turns on a certain feature in the usb mixer. Looking at the struct usb_mixer_interface I implemented it similar to "audigy2nx_leds" - by adding a new u8 (and the necessary functions). 2. Should one rather set ignore_ctl_error to 1 (which I currently do) or sort the errors out by adding a handling for a specific device? 3. For the switch to active I need an interrupt message for which I implemented snd_usb_interrupt_trans since I did not find any other convenience function to do this.

2 1

[alsa-devel] [PATCH 0/4] ASoC updates
by Mark Brown 07 Nov '08

07 Nov '08

The following changes since commit 0ee4663617fb0f78cec4cc6558a096ccbd8c3ffc: Takashi Iwai (1): ALSA: ASoC - Remove unnecessary inclusion of linux/version.h are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai David Anders (1): ASoC: Add new parameter to s3c24xx_pcm_enqueue Marek Vasut (1): ASoC: Add Palm/PXA27x unified ASoC audio driver Mark Brown (1): ASoC: Remove core version number Troy Kisky (1): ASoC: TLV320AIC23B Support more sample rates arch/arm/mach-pxa/include/mach/palmasoc.h | 13 ++ include/sound/soc.h | 2 - sound/soc/codecs/tlv320aic23.c | 227 +++++++++++++++++++----- sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/palm27x.c | 269 +++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c24xx-pcm.c | 12 +- sound/soc/soc-core.c | 1 - 8 files changed, 483 insertions(+), 52 deletions(-) create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h create mode 100644 sound/soc/pxa/palm27x.c

5 14

[alsa-devel] snd_ctl compiler warnings
by Alan Horstmann 07 Nov '08

07 Nov '08

The code I am working with at present includes unremarkable lines such as: snd_ctl_elem_value_t *line_in_vol; snd_ctl_elem_value_alloca(&line_in_vol); Building this with a cross compiler for arm based on gcc 4.2.1 gives this warning for the 2nd line: : warning: the address of 'line_in_vol' will always evaluate as 'true' but an older compiler on the desktop does not show this. Would this warning be expected in these circumstances, or is the compiler being over zealous? Am I missing something or is there a straightforward way to avoid such warnings? (Code runs OK). Thanks Alan

2 4

[alsa-devel] [PATCH] ALSA: have snd_BUG_ON() always refer to arguments
by Mike Frysinger 07 Nov '08

07 Nov '08

The snd_BUG_ON() macro should always expand its argument even if we're forcing it to false. This kills off unused warnings that did not exist before the snd_assert() -> snd_BUG_ON() conversion. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- include/sound/core.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 35424a9..87de9fc 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -387,7 +387,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) #define snd_printd(fmt, args...) /* nothing */ #define snd_BUG() /* nothing */ -#define snd_BUG_ON(cond) ({/*(void)(cond);*/ 0;}) /* always false */ +#define snd_BUG_ON(cond) ((cond) && 0) /* always false */ #endif /* CONFIG_SND_DEBUG */ -- 1.6.0.3

3 14

Re: [alsa-devel] supported sound cards
by Edward Terry 07 Nov '08

07 Nov '08

Clemens Ladisch wrote: > The only chip where hardware mixing is supported is Creative's Emu10k1 > (snd-emu10k1 driver), used on the SB Live!, most Audigy and certain > low-end X-Fi cards. (Cards with the 'real' X-Fi chip do _not_ work > well in Linux.) > > Other supported chips are ICE1724 (M-Audio Delta 1010(LT), DiO 2496, > 66, 44, 410, Audiophile 24/96; Digigram VX442; TerraTec EWX 24/96, > EWS 88MT/D, DMX 6Fire, Phase 88; Hoontech SoundTrack DSP 24/Value/ > Media7.1; Event EZ8; Lionstracs Mediastation, Terrasoniq TS 88) and > VT1720/24 (AMP AUDIO2000; M-Audio Revolution 5.1, 7.1, Audiophile 192; > TerraTec Aureon 5.1 Sky, 7.1 Space/Universe, Phase 22/28; > Onkyo SE-90PCI, SE-200PCI; AudioTrak Prodigy 192, 7.1 (HIFI/LT/XT), > HD2; Hercules Fortissimo IV; ESI Juli@; Pontis MS300; EGO-SYS > WaveTerminal 192M). Thanks. I'll look over these cards. > The most high-end supported cards are probably the Asus Xonar cards. > The D2/D2X have even slightly better audio quality than the best X-Fi, > and, as the obviously most important feature, colorfully illuminated > jacks. The various Dolby features are done in software in the Windows > driver and are not supported in Linux. If you need a PCI-E card, the > Xonar DX or D2X are your only choice. What about the Asus Xonar HDAV1.3? John Rigg wrote: >> The most high-end supported cards are probably the Asus Xonar cards. > > `High end' means different things in different situations of course. > RME HDSP series are probably the `highest end' cards supported, but > they need external converters, so they're probably unsuitable here. The Xonar cards do look interesting, but I'd also like to carry one higher-end card. I'm looking for a single-slot PCI Express solution that works as a standalone sound card, i.e. without any mandatory external devices (though optional ones are fine). Thanks for the information. Edward

3 2

[alsa-devel] Intel DG45ID motherboard and weird pinouts
by Jan Arie Gutter 07 Nov '08

07 Nov '08

Hi, I'm running Gentoo stable with kernel 2.6.27 (I think it comes with alsa 1.0.17) on my brand-new Intel DG45ID motherboard (hda-intel, therefore). I'm trying to get analogue out on all the correct channels, but it seems that something is slightly wierd: if I unmute headphones only, the front channel works, otherwise it seems that the front channel is split all over the place (I can provide more specific details if requested). I tried alsa-driver-1.0.18rc3 as well (keeping alsa-lib and alsa-utils the old version), but the headphones control disappeared: I could not find any equivalent settings to enable front speakers. Now, to the questions: 1) Can I help fix the pinout information in ALSA? I'm willing to experiment (especially if I can do it against 1.0.17). 2) Is there a doc somewhere that describes a step-by-step method to add a quirk to the codec (IDT 92HD73E1X5, I presume) for my board? 3) Would the problem be fixed if I update alsa-libs and alsa-utils to 1.0.18rc3, or is the problem only in the hda-intel kernel driver? I've got moderate kernel-hacking skills: I can compile, patch, debug minor stuff, etc, I've got ready access to the hardware and I'm willing to experiment! I DON'T have experience with ALSA or audio drivers, and since the PC is at home, I can only hack after hours. Any help would be greatly appreciated! Thanks! Jan Gutter -- "-and those damned stupid barbarians with their damned stupid swords will win after all..." -- Larry Niven (The Magic Goes Away)

2 2

[alsa-devel] (no subject)
by Wu Fengguang 07 Nov '08

07 Nov '08

1 1