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

[alsa-devel] [PATCH v2 0/8] ASoC: TWL4030: put the codec to off when not active
by Peter Ujfalusi 31 May '10

31 May '10

Hello, Changes since v1: - Patch 3: removed the shifting of offset cancellation value - Added register content reset functionality as suggested by Liam and Mark I have added Mark's Acked-by to patch 1-7, patch 8 is a new one, which shall address the comments regarding to the register content reset. To-do after this series: Revisit the register reset functionality to use burst write. Intro mail from the original series: The following series at the end will let the twl4030 codec to use BIAS_OFF instead of BIAS_STANDBY. The difference in power consumption is about 0.5mA. To achieve this: - The regcache has been reseted to codec default - The codec initialization has been optimized, it is no longer writes all 73 registers at startup, but only modifies few selected one. - The power related code has been cleaned up, and optimized - Support added for machine drivers to select the offset cancellation path - debug support for checking the codec default registers (machine drivers can ask for checking, but shall be disabled in production). - Added codec register reset functionality I guess that's it. The driver has been tested on a custom board with twl5031. It passed all of our internal test cases covering much of the codec features. --- Peter Ujfalusi (8): ASoC: TWL4030: Revisit codec defaults ASoC: TWL4030: Remove wrapper for power down ASoC: TWL4030: Make offset cancellation path configurable ASoC: TWL4030: Optimize the power up sequence ASoC: TWL4030: Helper to check chip default registers ASoC: TWL4030: Correct the ARXR2_APGA_CTL chip default ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use ASoC: TWL4030: Add functionalty to reset the registers sound/soc/codecs/twl4030.c | 291 +++++++++++++++++++++++++------------------- sound/soc/codecs/twl4030.h | 3 + 2 files changed, 169 insertions(+), 125 deletions(-)

3 10

[alsa-devel] How to support software mixing on QUALCOMM 72XX without mmap?
by Berg 31 May '10

31 May '10

Hello,guys. I'm porting ASLA1.0.23 to qualcomm 72xx base on ARM11. The first instance,"aplay test.wav", works now, but the second instance should fail with "device busy" error message. I have tried plug dmix and dshare "aply -D plug:mycard test.wav", but met a fail again for the soc chip doesn't support mmap. Of course,pcm_mycard is defined in /etc/asound.conf. In my pc ,both above cmds work well . Though,the document shows that: NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix for analogue output. Dmix is enabled by default for soundcards which don't support hardware mixing. You still need to set it up for digital outputs. My questions are: 1)Is software mixing implemented in ALSA Driver,or in ALSA LIB? 2)What could I do to make the software mixing work? Is there any special configure which need to adjust? And when testing on pc whth HD audio, I noticed that A)when aplay instances runs WITHOUT dmix, the result of"cat /etc/one_of_my_aplay_pid/maps" should like: ... b331d000-b731e000 rw-s 00000000 00:15 862346 /dev/shm/pulse-shm-2882420837 ... B)when aplay instances runs WITH dmix,the result is: ... b7511000-b7519000 rw-s 00000000 00:0f 4560 /dev/snd/pcmC0D0p b7519000-b751a000 rw-s 81000000 00:0f 4560 /dev/snd/pcmC0D0p b751a000-b751b000 r--s 80000000 00:0f 4560 /dev/snd/pcmC0D0p ... So,the third question: 3) From A) and B),can I make a result that the mmap feacture is not necessary for the buildin software mixing ? Expect your help,please berg

3 2

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for another mainboard
by Daniel Chen 31 May '10

31 May '10

Hi, Attached is a patch against my latest LPIB patches that makes the original reporter's playback and capture audible by default. Please apply. This patch is applicable to all 2.6.3x stable series. Best, -Dan

2 1

[alsa-devel] [PATCH] ALSA: hda: Use mbp31 quirk for an iMac model
by Daniel Chen 31 May '10

31 May '10

Hi, Attached is a patch against sound-2.6 master enabling audible playback and capture for the original reporter's iMac. Please apply. This patch is applicable to all 2.6.3x stable series. Best, -Dan

2 2

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for an ASUS device
by Daniel Chen 31 May '10

31 May '10

Hi, Attached is a patch against sound-2.6 master that enables the original reporter to adjust capture volume through PulseAudio without the daemon crashing. This patch is applicable to all 2.6.3x stable series. Best, -Dan

2 1

[alsa-devel] [PATCH] pcm: Define G723 3-bit and 5-bit formats
by Ben Collins 31 May '10

31 May '10

pcm: Define G723 3-bit and 5-bit formats This defines the 24bps and 40bps (8khz sample rate) G.723 codec formats. They are going to be used once I submit the driver for an mpeg4/g723 compression card. I've updated the signed value to -1 as per Takashi's comments since these are non-linear formats. Signed-off-by: Ben Collins <bcollins(a)bluecherry.net> Acked-by: Takashi Iwai <tiwai(a)suse.de> diff --git a/include/sound/asound.h b/include/sound/asound.h index 0985955..62e0857 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -212,7 +212,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_S18_3BE ((__force snd_pcm_format_t) 41) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3LE ((__force snd_pcm_format_t) 42) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3BE ((__force snd_pcm_format_t) 43) /* in three bytes */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_U18_3BE +#define SNDRV_PCM_FORMAT_G723_24 ((__force snd_pcm_format_t) 44) /* 8 samples in 3 bytes */ +#define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ +#define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ +#define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 8b611a5..37b5c3c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -173,6 +173,10 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_U18_3LE (1ULL << SNDRV_PCM_FORMAT_U18_3LE) #define SNDRV_PCM_FMTBIT_S18_3BE (1ULL << SNDRV_PCM_FORMAT_S18_3BE) #define SNDRV_PCM_FMTBIT_U18_3BE (1ULL << SNDRV_PCM_FORMAT_U18_3BE) +#define SNDRV_PCM_FMTBIT_G723_24 (1ULL << SNDRV_PCM_FORMAT_G723_24) +#define SNDRV_PCM_FMTBIT_G723_24_1B (1ULL << SNDRV_PCM_FORMAT_G723_24_1B) +#define SNDRV_PCM_FMTBIT_G723_40 (1ULL << SNDRV_PCM_FORMAT_G723_40) +#define SNDRV_PCM_FMTBIT_G723_40_1B (1ULL << SNDRV_PCM_FORMAT_G723_40_1B) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index ea2bf82..434af3c 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -128,6 +128,14 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { .width = 4, .phys = 4, .le = -1, .signd = -1, .silence = {}, }, + [SNDRV_PCM_FORMAT_G723_24] = { + .width = 3, .phys = 3, .le = -1, .signd = -1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_G723_40] = { + .width = 5, .phys = 5, .le = -1, .signd = -1, + .silence = {}, + }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, @@ -186,6 +194,14 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { .width = 18, .phys = 24, .le = 0, .signd = 0, .silence = { 0x02, 0x00, 0x00 }, }, + [SNDRV_PCM_FORMAT_G723_24_1B] = { + .width = 3, .phys = 8, .le = -1, .signd = -1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_G723_40_1B] = { + .width = 5, .phys = 8, .le = -1, .signd = -1, + .silence = {}, + }, }; -- Bluecherry: http://www.bluecherrydvr.com/ SwissDisk : http://www.swissdisk.com/ Ubuntu : http://www.ubuntu.com/ My Blog : http://ben-collins.blogspot.com/

2 1

[alsa-devel] [PATCH 1/4] snd-usb-caiaq: Restore 'Control vinyl' input mode on A4DJ
by Mark Hills 31 May '10

31 May '10

This feature was undocumented on early A4DJ units. It is indicated by lighting both the 'line' and 'phono' lamps at the same time. Newer units document this and the newer Windows drivers enable this for all units, so restore the functionality. This patch simplifies the code and changes the mode mapping to match the A8DJ, favouring simpler code and consistency over keeping the existing mapping. Both 'Control vinyl' and 'Phono' input modes enable the hardware preamp. The difference is the input impedance. This reverts commit 9a9527e. Acked-by: Daniel Mack <daniel(a)caiaq.de> Signed-off-by: Mark Hills <mark(a)pogo.org.uk> --- sound/usb/caiaq/control.c | 30 ++---------------------------- 1 files changed, 2 insertions(+), 28 deletions(-) diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index 36ed703..70c3866 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -42,21 +42,12 @@ static int control_info(struct snd_kcontrol *kcontrol, switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): - if (pos == 0) { - /* current input mode of A8DJ */ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 2; - return 0; - } - break; - case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): if (pos == 0) { - /* current input mode of A4DJ */ + /* current input mode of A8DJ and A4DJ */ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; + uinfo->value.integer.max = 2; return 0; } break; @@ -86,14 +77,6 @@ static int control_get(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; - if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) { - /* A4DJ has only one control */ - /* do not expose hardware input mode 0 */ - ucontrol->value.integer.value[0] = dev->control_state[0] - 1; - return 0; - } - if (pos & CNT_INTVAL) ucontrol->value.integer.value[0] = dev->control_state[pos & ~CNT_INTVAL]; @@ -113,15 +96,6 @@ static int control_put(struct snd_kcontrol *kcontrol, unsigned char cmd = EP1_CMD_WRITE_IO; switch (dev->chip.usb_id) { - case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): { - /* A4DJ has only one control */ - /* do not expose hardware input mode 0 */ - dev->control_state[0] = ucontrol->value.integer.value[0] + 1; - snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, - dev->control_state, sizeof(dev->control_state)); - return 1; - } - case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): cmd = EP1_CMD_DIMM_LEDS; break; -- 1.7.1

2 4

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for a Shuttle device
by Daniel Chen 31 May '10

31 May '10

Hi, Attached please find a patch against current sound-2.6 master that fixes audio anomalies and PulseAudio crashing by specifying LPIB method explicitly. Please apply. This patch is applicable to all 2.6.3x stable series. Best, -Dan

2 1

[alsa-devel] [PATCH] ALSA: usb/mixer.c: remove unused variable
by Daniel Mack 31 May '10

31 May '10

This was a development left-over. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/mixer.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index fd3e08d..a35dd74 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1970,8 +1970,6 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, int attribute, int value, int index) { - struct usb_mixer_elem_info *info; - switch (attribute) { case UAC2_CS_CUR: snd_usb_mixer_notify_id(mixer, index); -- 1.7.1

3 3

Re: [alsa-devel] Problem with snd_hda_intel from 05-27-2010 snapshot
by Jaroslav Kysela 30 May '10

30 May '10

On Thu, 27 May 2010, VDR User wrote: > On Thu, May 27, 2010 at 12:37 AM, Jaroslav Kysela <perex(a)perex.cz> wrote: >> Check any suspicious lines in /var/log/messages and follow proc xrun_debug >> settings noted in bug above. I need at least 20 first lines after the >> playback is started. > > By this do you mean compile the drivers with pcm-xrun-debug > --with-debug=verbose --enable-verbose-procfs options? Also, is there > a way to disable all the debug logging in maybe > /etc/modprobe.d/alsa-base.conf? My system is installed on a CF and I > can't have alsa hammering it with writes is they're not needed but the > only way I know to turn the debug logging off is to not compile it in > the first place. I asked for one run, so you can recompile ALSA driver without debugging support after test. I think that the problem might be that the WALLCLK register gives wrong values for your hardware (but the Intel specification says that implementation of this register is MANDATORY). Here is a patch which might show the bug: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 170610e..a1e7c97 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1891,6 +1891,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) int stream; wallclk = azx_readl(chip, WALLCLK) - azx_dev->start_wallclk; + printk("wallclk: %u, %u\n", wallclk, azx_readl(chip, WALLCLK)); if (wallclk < (azx_dev->period_wallclk * 2) / 3) return -1; /* bogus (too early) interrupt */ >> Also, use 'aplay -Dplughw:0' from the alsa-utils package to trigger the >> playback. > > The hdmi device I'm using is 1,3: > > test:~$ aplay -l > **** List of PLAYBACK Hardware Devices **** > card 0: NVidia [HDA NVidia], device 0: ALC883 Analog [ALC883 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: NVidia [HDA NVidia], device 1: ALC883 Digital [ALC883 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: NVidia_1 [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > > So what exactly do I need to do then with aplay? Do I need some file > to play? Sorry, I've never used Linux as a desktop so I'm only > familiar with running VDR from command line. If you use HDMI device, use 'aplay plughw:1,0 <some_wav_file>'. Some_wav_file means a .wav type audio file. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 2

[alsa-devel] How to support software mixing on QUALCOMM 72XX without mmap?
by Berg.Tian 29 May '10

29 May '10

Hello,guys. I'm porting ASLA1.0.23 to qualcomm 72xx base on ARM11. The first instance,"aplay test.wav", works now, but the second instance should fail with "device busy" error message. I have tried plug dmix and dshare "aply -D plug:mycard test.wav", but met a fail again for the soc chip doesn't support mmap. Of course,pcm_mycard is defined in /etc/asound.conf. In my pc ,both above cmds work well . Though,the document shows that: NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix for analogue output. Dmix is enabled by default for soundcards which don't support hardware mixing. You still need to set it up for digital outputs. My questions are: 1)Is software mixing implemented in ALSA Driver,or in ALSA LIB? 2)What could I do to make the software mixing work? Is there any special configure which need to adjust? And when testing on pc whth HD audio, I noticed that A)when aplay instances runs WITHOUT dmix, the result of"cat /etc/one_of_my_aplay_pid/maps" should like: ... b331d000-b731e000 rw-s 00000000 00:15 862346 /dev/shm/pulse-shm-2882420837 ... B)when aplay instances runs WITH dmix,the result is: ... b7511000-b7519000 rw-s 00000000 00:0f 4560 /dev/snd/pcmC0D0p b7519000-b751a000 rw-s 81000000 00:0f 4560 /dev/snd/pcmC0D0p b751a000-b751b000 r--s 80000000 00:0f 4560 /dev/snd/pcmC0D0p ... So,the third question: 3) From A) and B),can I make a result that the mmap feacture is not necessary for the buildin software mixing ? Expect your help,please berg

1 0

Re: [alsa-devel] Mixer volume/decibel mapping
by Clemens Ladisch 29 May '10

29 May '10

Raymond Yau wrote: > 2010/5/21 Clemens Ladisch <clemens(a)ladisch.de> > > DB_LINEAR means that the mixer control value is a multiplier for the > > PCM sample value, as James wrote. A DB_LINEAR mapping has > > 0 == -infinity dB, full scale == 0 dB, and the mapping between mixer > > control values and corresponding dB values it _not_ linear. > > pci/oxygen/oxygen.c:static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); > pci/oxygen/hifier.c:static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); > > Refer to AK4396 datasheet > > Output Volume > > The AK4396 includes channel independent digital output volumes (ATT) with > 256 levels at linear step including MUTE. > These volumes are in front of the DAC and can attenuate the input data from > 0dB to –48dB and mute. > > dB range is -48dB to 0dB only > > ATT7-0: Attenuation Level > ATT = 20 log10 (ATT_DATA / 255) [dB] > FFH : 0dB (Default) > 00H : Mute This is the section of the datasheet on which the above declarations are based. Now if you have some remark or question, you should remember to include it in your mail. Regards, Clemens

2 7

Re: [alsa-devel] [PATCH] ALSA/davinci: Use the resource_size , __devinit and __devexit.
by Sergei Shtylyov 28 May '10

28 May '10

Hello. Wan ZongShun wrote: > Dear all, > Use the resource_size function instead of manually calculating the > resource size. This reduces the chance of introducing off-by-one > errors. > In addition, to use __devinit and __devexit is good too. "Good" isn't a good justification I think. Also, this just asks to be put into two patches instead of one as the patch does two completely different things. > Signed-off-by:Wan ZongShun<mcuos.com(a)gmail.com> WBR, Sergei

1 0

[alsa-devel] [PATCH] ALSA, hda: Add support for another Lenovo ThinkPad Edge in conexant codec
by Andreas Herrmann 28 May '10

28 May '10

From: Andreas Herrmann <andreas.herrmann3(a)amd.com> On a Thinkpad Edge 13 "01972NG" I had the problem that speakers played sound although headphones were plugged in. Using model=ideapad with latest alsa-git kernel fixed this. So adding this quirk to use ideapad for another Thinkpad Edge variant seems sensible. Cc: Jerone Young <jerone.young(a)canonical.com> Signed-off-by: Andreas Herrmann <andreas.herrmann3(a)amd.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) Patch is against today's sound-2.6 git tree. Please apply. Thanks, Andreas diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f4a2bd6..2bf2cb5 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2975,6 +2975,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), -- 1.7.1

2 1

Re: [alsa-devel] [PATCH] ALSA, hda: Add support for Lenovo ThinkPad Edge 13 in conexant codec
by Takashi Iwai 28 May '10

28 May '10

At Thu, 27 May 2010 18:06:56 +0200, Andreas Herrmann wrote: > > From: Andreas Herrmann <andreas.herrmann3(a)amd.com> > > On a Thinkpad Edge 13 I had the problem that speakers played sound > although headphones were plugged. > > Using "model=olpc-xo-1_5" solved this problem. So it seems sensible to > add this quirk. (Using other models, like ideabad did not fix it.) > > Signed-off-by: Andreas Herrmann <andreas.herrmann3(a)amd.com> > --- > sound/pci/hda/patch_conexant.c | 1 + > 1 files changed, 1 insertions(+), 0 deletions(-) > > I don't know whether adding this quirk is the right/complete solution > to support sound on Thinkpad Edge. But at least it solves this > annoying problem. Well, Jerone Young already sent a quirk patch but he took model=ideapd. Could you guys check which one is really better? thanks, Takashi > > Please apply. > > > Thanks, > > Andreas > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index feabb44..f85f072 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -2848,6 +2848,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { > SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), > SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), > SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), > + SND_PCI_QUIRK(0x17aa, 0x21b3, "Thnkpad Edge 13", CXT5066_OLPC_XO_1_5), > SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), > {} > }; > -- > 1.6.4.4 >

3 4

Re: [alsa-devel] [PATCH 3/4] ALSA: usb-audio: parse UAC2 endpoint descriptors correctly
by Takashi Iwai 28 May '10

28 May '10

At Wed, 26 May 2010 13:46:57 -0700, Greg KH wrote: > > On Wed, May 26, 2010 at 06:17:46PM +0200, Daniel Mack wrote: > > On Wed, May 26, 2010 at 06:11:38PM +0200, Daniel Mack wrote: > > > UAC2 devices have their information about pitch control stored in a > > > different field. Parse it, and emulate the bits for a v1 device. > > > > > > A new struct uac2_iso_endpoint_descriptor is added. > > > > > > > > Signed-off-by: Daniel Mack <daniel(a)caiaq.de> > > > --- > > > include/linux/usb/audio-v2.h | 16 ++++++++++++ > > > > Greg, I copied you because of this patch, just to let you know we're > > having fun in your area of responsibility again :) > > Heh, no problem at all. > > Takashi, feel free to add an: > Acked-by: Greg Kroah-Hartman <gregkh(a)suse.de> > to this patch when you take it. OK, I put this to my tree. Will be included in the next pull request in this week. thanks, Takashi

2 2

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for Sony VPCS11V9E
by Daniel Chen 28 May '10

28 May '10

Hi, Attached is a patch against sound-2.6 master HEAD that resolves stuttering audio on the original reporter's hardware. Please apply. This patch is applicable to all stable trees of 2.6.3x. Best, -Dan

1 0

[alsa-devel] usbaudio: Support for USB audio v2.0 devices
by Daniel Mack 27 May '10

27 May '10

Hi, I've been working on support for USB devices compliant to audio class v2.0 and would like to share what I got. First problem is that they literally changed every single descriptor type by adding or removing fields or changing their width, respectively. Even worse, some enum values were dropped and reassigned so their numerical value has a different meaning now. My first step was to clean up most of the existing descriptor parsers by moving away from anonymous byte array access to real structs so we can see which fields we're actually dereferencing. In a second step, I added definitions for the replacement descriptors and use them at appropriate places. The bInterfaceProtocol field must always be set correctly of course, otherwise we'll parse garbage. Other things that have changed from v1.0 to v2.0 are: * The number of streaming interfaces is now reported by a standard interface association descriptor. The old approach using a proprietary descriptor extension is deprecated. * The number of channels per interface is now stored in the AS_GENERAL descriptor (used to be part of the FORMAT_TYPE descriptor). * The list of supported sample rates is no longer stored in a variable length appendix of the FORMAT_TYPE descriptor but is retrieved from the device using a class specific GET_RANGE command. * Supported sample formats are now reported as 32bit bitmap rather than as a fixed value. For now, this is worked around by choosing just one of them. Eventually, the code should propagate the full ability of the device. * A devices needs to have at least one CLOCK_SOURCE descriptor which denotes a clockID that is needed as argument to the class request command. If can have more than one though and the host driver is in charge to choose the right one. There are also new types for clock selectors and sample rate converters that a device might report. This is all unsupported at the moment. We only parse the first CLOCK_SOURCE and take this one. * Many descriptors (format_type, ...) have changed their layout. Handle this by casting the descriptors to the appropriate structs. With the following patches applied, a v2.0 device is recognized and reported as ALSA device. I can even hear at least some kind of music, but the sample rate setting does not yet seem to be correct. Also, the existing mixer code does not work at all for v2.0 devices. This will be quite some work to support that and I fear the whole thing must be reimplemented for v2.0 specifically. For now, I worked around that by not parsing them at all but bailing out very early. However, I would like to make sure the changes I made so far won't cause any regression for v1.0 devices, and so it would be good to see them applied as a preliminary support. I'll continue working on this and send more updates along. Let me know what you think. Thanks, Daniel

5 33

[PATCH] ALSA: usb-audio: fix feature unit parser for UAC2
by Daniel Mack 27 May '10

27 May '10

Fix a small off-by-one bug which causes the feature unit to announce a wrong number of channels. This leads to illegal requests sent to the firmware eventually. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/mixer.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 97dd176..03ce971 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1126,7 +1126,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } else { struct uac2_feature_unit_descriptor *ftr = _ftr; csize = 4; - channels = (hdr->bLength - 6) / 4; + channels = (hdr->bLength - 6) / 4 - 1; bmaControls = ftr->bmaControls; } -- 1.7.1

1 0

[PATCH] ALSA: usb-audio: fix feature unit parser for UAC2
by Daniel Mack 27 May '10

27 May '10

Fix a small off-by-one bug which causes the feature unit to announce a wrong number of channels. This leads to illegal requests sent to the firmware eventually. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/mixer.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 97dd176..03ce971 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1126,7 +1126,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } else { struct uac2_feature_unit_descriptor *ftr = _ftr; csize = 4; - channels = (hdr->bLength - 6) / 4; + channels = (hdr->bLength - 6) / 4 - 1; bmaControls = ftr->bmaControls; } -- 1.7.1

1 0

[PATCH] ALSA: usb-audio: parse clock topology of UAC2 devices
by Daniel Mack 27 May '10

27 May '10

Audio devices which comply to the UAC2 standard can export complex clock topologies in its descriptors and set up links between them. The entities that are defined are - clock sources, which define the end-leafs. - clock selectors, which act as switch to select one out of many possible clocks sources. - clock multipliers, which have an input clock source, and act as clock source again. They can be used to derive one clock from another. All sample rate changes, clock validity queries and the like must go to clock source elements, while clock selectors and multipliers can be used as terminal clock source. The following patch adds a parser for these elements and functions to iterate over the tree and find the leaf nodes (clock sources). The samplerate set functions were moved to the new clock.c file. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/Makefile | 3 +- sound/usb/card.c | 18 +--- sound/usb/card.h | 1 + sound/usb/clock.c | 311 ++++++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/clock.h | 12 ++ sound/usb/endpoint.c | 57 +++++++++- sound/usb/format.c | 16 ++- sound/usb/pcm.c | 98 +---------------- sound/usb/usbaudio.h | 5 +- 9 files changed, 396 insertions(+), 125 deletions(-) create mode 100644 sound/usb/clock.c create mode 100644 sound/usb/clock.h diff --git a/sound/usb/Makefile b/sound/usb/Makefile index e7ac7f4..1e362bf 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -11,7 +11,8 @@ snd-usb-audio-objs := card.o \ endpoint.o \ urb.o \ pcm.o \ - helper.o + helper.o \ + clock.o snd-usbmidi-lib-objs := midi.o diff --git a/sound/usb/card.c b/sound/usb/card.c index da1346b..7a8ac1d 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -236,7 +236,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } case UAC_VERSION_2: { - struct uac_clock_source_descriptor *cs; struct usb_interface_assoc_descriptor *assoc = usb_ifnum_to_if(dev, ctrlif)->intf_assoc; @@ -245,21 +244,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) return -EINVAL; } - /* FIXME: for now, we expect there is at least one clock source - * descriptor and we always take the first one. - * We should properly support devices with multiple clock sources, - * clock selectors and sample rate conversion units. */ - - cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, UAC2_CLOCK_SOURCE); - - if (!cs) { - snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); - return -EINVAL; - } - - chip->clock_id = cs->bClockID; - for (i = 0; i < assoc->bInterfaceCount; i++) { int intf = assoc->bFirstInterface + i; @@ -481,6 +465,8 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __error; } + chip->ctrl_intf = alts; + if (err > 0) { /* create normal USB audio interfaces */ if (snd_usb_create_streams(chip, ifnum) < 0 || diff --git a/sound/usb/card.h b/sound/usb/card.h index ed92420..1febf2f 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -25,6 +25,7 @@ struct audioformat { unsigned int rate_min, rate_max; /* min/max rates */ unsigned int nr_rates; /* number of rate table entries */ unsigned int *rate_table; /* rate table */ + unsigned char clock; /* associated clock */ }; struct snd_usb_substream; diff --git a/sound/usb/clock.c b/sound/usb/clock.c new file mode 100644 index 0000000..b7aadd6 --- /dev/null +++ b/sound/usb/clock.c @@ -0,0 +1,311 @@ +/* + * Clock domain and sample rate management functions + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/bitops.h> +#include <linux/init.h> +#include <linux/list.h> +#include <linux/slab.h> +#include <linux/string.h> +#include <linux/usb.h> +#include <linux/moduleparam.h> +#include <linux/mutex.h> +#include <linux/usb/audio.h> +#include <linux/usb/audio-v2.h> + +#include <sound/core.h> +#include <sound/info.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> + +#include "usbaudio.h" +#include "card.h" +#include "midi.h" +#include "mixer.h" +#include "proc.h" +#include "quirks.h" +#include "endpoint.h" +#include "helper.h" +#include "debug.h" +#include "pcm.h" +#include "urb.h" +#include "format.h" + +static struct uac_clock_source_descriptor * + snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac_clock_source_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC2_CLOCK_SOURCE))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + +static struct uac_clock_selector_descriptor * + snd_usb_find_clock_selector(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac_clock_selector_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC2_CLOCK_SELECTOR))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + +static struct uac_clock_multiplier_descriptor * + snd_usb_find_clock_multiplier(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac_clock_multiplier_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC2_CLOCK_MULTIPLIER))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + +static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_id) +{ + unsigned char buf; + int ret; + + ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), + UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + UAC2_CX_CLOCK_SELECTOR << 8, selector_id << 8, + &buf, sizeof(buf), 1000); + + if (ret < 0) + return ret; + + return buf; +} + +static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) +{ + int err; + unsigned char data; + struct usb_device *dev = chip->dev; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_CLOCK_VALID << 8, source_id << 8, + &data, sizeof(data), 1000); + + if (err < 0) { + snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n", + __func__, source_id); + return err; + } + + return !!data; +} + +/* Try to find the clock source ID of a given clock entity */ + +static int __uac_clock_find_source(struct snd_usb_audio *chip, + struct usb_host_interface *host_iface, + int entity_id, unsigned long *visited) +{ + struct uac_clock_source_descriptor *source; + struct uac_clock_selector_descriptor *selector; + struct uac_clock_multiplier_descriptor *multiplier; + + entity_id &= 0xff; + + if (test_and_set_bit(entity_id, visited)) { + snd_printk(KERN_WARNING + "%s(): recursive clock topology detected, id %d.\n", + __func__, entity_id); + return -EINVAL; + } + + /* first, see if the ID we're looking for is a clock source already */ + source = snd_usb_find_clock_source(host_iface, entity_id); + if (source) + return source->bClockID; + + selector = snd_usb_find_clock_selector(host_iface, entity_id); + if (selector) { + int ret; + + /* the entity ID we are looking for is a selector. + * find out what it currently selects */ + ret = uac_clock_selector_get_val(chip, selector->bClockID); + if (ret < 0) + return ret; + + if (ret > selector->bNrInPins || ret < 1) { + printk(KERN_ERR + "%s(): selector reported illegal value, id %d, ret %d\n", + __func__, selector->bClockID, ret); + + return -EINVAL; + } + + return __uac_clock_find_source(chip, host_iface, + selector->baCSourceID[ret-1], + visited); + } + + /* FIXME: multipliers only act as pass-thru element for now */ + multiplier = snd_usb_find_clock_multiplier(host_iface, entity_id); + if (multiplier) + return __uac_clock_find_source(chip, host_iface, + multiplier->bCSourceID, visited); + + return -EINVAL; +} + +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct usb_host_interface *host_iface, + int entity_id) +{ + DECLARE_BITMAP(visited, 256); + memset(visited, 0, sizeof(visited)); + return __uac_clock_find_source(chip, host_iface, entity_id, visited); +} + +static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_device *dev = chip->dev; + unsigned int ep; + unsigned char data[3]; + int err, crate; + + ep = get_endpoint(alts, 0)->bEndpointAddress; + + /* if endpoint doesn't have sampling rate control, bail out */ + if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) { + snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n", + dev->devnum, iface, fmt->altsetting); + return 0; + } + + data[0] = rate; + data[1] = rate >> 8; + data[2] = rate >> 16; + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", + dev->devnum, iface, fmt->altsetting, rate, ep); + return err; + } + + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", + dev->devnum, iface, fmt->altsetting, ep); + return 0; /* some devices don't support reading */ + } + + crate = data[0] | (data[1] << 8) | (data[2] << 16); + if (crate != rate) { + snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + // runtime->rate = crate; + } + + return 0; +} + +static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_device *dev = chip->dev; + unsigned char data[4]; + int err, crate; + int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fmt->clock); + + if (clock < 0) + return clock; + + if (!uac_clock_source_is_valid(chip, clock)) { + snd_printk(KERN_ERR "%d:%d:%d: clock source %d is not valid, cannot use\n", + dev->devnum, iface, fmt->altsetting, clock); + return -ENXIO; + } + + data[0] = rate; + data[1] = rate >> 8; + data[2] = rate >> 16; + data[3] = rate >> 24; + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, + UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n", + dev->devnum, iface, fmt->altsetting, rate); + return err; + } + + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + return err; + } + + crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + if (crate != rate) + snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + + return 0; +} + +int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_interface_descriptor *altsd = get_iface_desc(alts); + + switch (altsd->bInterfaceProtocol) { + case UAC_VERSION_1: + return set_sample_rate_v1(chip, iface, alts, fmt, rate); + + case UAC_VERSION_2: + return set_sample_rate_v2(chip, iface, alts, fmt, rate); + } + + return -EINVAL; +} + diff --git a/sound/usb/clock.h b/sound/usb/clock.h new file mode 100644 index 0000000..beb2536 --- /dev/null +++ b/sound/usb/clock.h @@ -0,0 +1,12 @@ +#ifndef __USBAUDIO_CLOCK_H +#define __USBAUDIO_CLOCK_H + +int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate); + +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct usb_host_interface *host_iface, + int entity_id); + +#endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 28ee1ce..9593b91 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -190,6 +190,38 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, return attributes; } +static struct uac2_input_terminal_descriptor * + snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, + int terminal_id) +{ + struct uac2_input_terminal_descriptor *term = NULL; + + while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + term, UAC_INPUT_TERMINAL))) { + if (term->bTerminalID == terminal_id) + return term; + } + + return NULL; +} + +static struct uac2_output_terminal_descriptor * + snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, + int terminal_id) +{ + struct uac2_output_terminal_descriptor *term = NULL; + + while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + term, UAC_OUTPUT_TERMINAL))) { + if (term->bTerminalID == terminal_id) + return term; + } + + return NULL; +} + int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { struct usb_device *dev; @@ -199,7 +231,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) int i, altno, err, stream; int format = 0, num_channels = 0; struct audioformat *fp = NULL; - int num, protocol; + int num, protocol, clock = 0; struct uac_format_type_i_continuous_descriptor *fmt; dev = chip->dev; @@ -263,6 +295,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) } case UAC_VERSION_2: { + struct uac2_input_terminal_descriptor *input_term; + struct uac2_output_terminal_descriptor *output_term; struct uac_as_header_descriptor_v2 *as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); @@ -281,7 +315,25 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) num_channels = as->bNrChannels; format = le32_to_cpu(as->bmFormats); - break; + /* lookup the terminal associated to this interface + * to extract the clock */ + input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (input_term) { + clock = input_term->bCSourceID; + break; + } + + output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (output_term) { + clock = output_term->bCSourceID; + break; + } + + snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n", + dev->devnum, iface_no, altno, as->bTerminalLink); + continue; } default: @@ -338,6 +390,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) * (fp->maxpacksize & 0x7ff); fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no); + fp->clock = clock; /* some quirks for attributes here */ diff --git a/sound/usb/format.c b/sound/usb/format.c index fe29d61..5367cd1 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -29,6 +29,7 @@ #include "quirks.h" #include "helper.h" #include "debug.h" +#include "clock.h" /* * parse the audio format type I descriptor @@ -215,15 +216,17 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; int i, nr_rates, data_size, ret = 0; + int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fp->clock); /* get the number of sample rates first by only fetching 2 bytes */ ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, + UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8, tmp, sizeof(tmp), 1000); if (ret < 0) { - snd_printk(KERN_ERR "unable to retrieve number of sample rates\n"); + snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n", + __func__, clock); goto err; } @@ -237,12 +240,13 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, /* now get the full information */ ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, - data, data_size, 1000); + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8, + data, data_size, 1000); if (ret < 0) { - snd_printk(KERN_ERR "unable to retrieve sample rate range\n"); + snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n", + __func__, clock); ret = -EINVAL; goto err_free; } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 056587d..4568298 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -31,6 +31,7 @@ #include "urb.h" #include "helper.h" #include "pcm.h" +#include "clock.h" /* * return the current pcm pointer. just based on the hwptr_done value. @@ -181,103 +182,6 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, return -EINVAL; } -static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) -{ - struct usb_device *dev = chip->dev; - unsigned int ep; - unsigned char data[3]; - int err, crate; - - ep = get_endpoint(alts, 0)->bEndpointAddress; - /* if endpoint doesn't have sampling rate control, bail out */ - if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) { - snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n", - dev->devnum, iface, fmt->altsetting); - return 0; - } - - data[0] = rate; - data[1] = rate >> 8; - data[2] = rate >> 16; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, - USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", - dev->devnum, iface, fmt->altsetting, rate, ep); - return err; - } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, - USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", - dev->devnum, iface, fmt->altsetting, ep); - return 0; /* some devices don't support reading */ - } - crate = data[0] | (data[1] << 8) | (data[2] << 16); - if (crate != rate) { - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); - // runtime->rate = crate; - } - - return 0; -} - -static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) -{ - struct usb_device *dev = chip->dev; - unsigned char data[4]; - int err, crate; - - data[0] = rate; - data[1] = rate >> 8; - data[2] = rate >> 16; - data[3] = rate >> 24; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n", - dev->devnum, iface, fmt->altsetting, rate); - return err; - } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, - data, sizeof(data), 1000)) < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", - dev->devnum, iface, fmt->altsetting); - return err; - } - crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); - if (crate != rate) - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); - - return 0; -} - -int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) -{ - struct usb_interface_descriptor *altsd = get_iface_desc(alts); - - switch (altsd->bInterfaceProtocol) { - case UAC_VERSION_1: - return set_sample_rate_v1(chip, iface, alts, fmt, rate); - - case UAC_VERSION_2: - return set_sample_rate_v2(chip, iface, alts, fmt, rate); - } - - return -EINVAL; -} - /* * find a matching format and set up the interface */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 06ebf24..24d3319 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -40,9 +40,6 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; - /* for audio class v2 */ - int clock_id; - struct list_head pcm_list; /* list of pcm streams */ int pcm_devs; @@ -53,6 +50,8 @@ struct snd_usb_audio { int setup; /* from the 'device_setup' module param */ int nrpacks; /* from the 'nrpacks' module param */ int async_unlink; /* from the 'async_unlink' module param */ + + struct usb_host_interface *ctrl_intf; /* the audio control interface */ }; /* -- 1.7.1

1 0

Re: [alsa-devel] No mixers on ATI RS780 Azalia
by Clemens Ladisch 27 May '10

27 May '10

Robert Hancock wrote: > On 05/27/2010 06:34 AM, Jan Engelhardt wrote: >> On Thursday 2010-05-27 13:42, Clemens Ladisch wrote: >>> It is possible that an embedded device like this requires some custom >>> initialization. Can you find out if the Windows driver is the standard >>> Microsoft driver or Samsung's? >> >> Looks like the Microsoft one: > > Well, not Samsung, but it's not the Microsoft HDA driver either, it's > using a Realtek driver. > >> == "ATI HDMI Audio" == >> ... >> C:\WINDOWS\system32\drivers\RtHDMI.sys >> Realtek, 5.10.0.5602 built by WinDDK Googling shows that this is the driver that AMD ships for its "ATI HDMI" devices. >> == "Realtek High Definition Audio" == >> ... This also looks like the standard driver. Apparently, this computer uses standard audio hardware; the culprit is probably the Linux radeon driver. Regards, Clemens

1 0

[alsa-devel] [PATCH] imx-ssi.c: fix I2S slave setup
by Eric Bénard 27 May '10

27 May '10

* without this fix, the waveform on the SSI port are not real I2S signal (at least on i.MX27's SSI4) * original Freescale's driver has if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) && ssi_mode->network_mode) { scr &= ~SSI_SCR_I2S_MODE_MASK; scr |= SSI_SCR_I2S_MODE_SLAVE; } So as now the SSI is set in network mode as a default when running in I2S format, it seems reasonable to also set I2S_MODE_SLAVE Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/imx-ssi.c | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 80b4fee..c5e1626 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -83,8 +83,6 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, /* * SSI DAI format configuration. * Should only be called when port is inactive (i.e. SSIEN = 0). - * Note: We don't use the I2S modes but instead manually configure the - * SSI for I2S because the I2S mode is only a register preset. */ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -98,7 +96,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) case SND_SOC_DAIFMT_I2S: /* data on rising edge of bclk, frame low 1clk before data */ strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; - scr |= SSI_SCR_NET; + scr |= SSI_SCR_NET | SSI_SCR_I2S_MODE_SLAVE; break; case SND_SOC_DAIFMT_LEFT_J: /* data on rising edge of bclk, frame high with data */ -- 1.6.3.3

3 4

[alsa-devel] [PATCH] alsa-python: add register_poll to alsaseq
by Hector Martin 27 May '10

27 May '10

Attached patch adds register_poll to the sequencer bindings to support nonblocking sequencer access with the select module. I sent this last month but it looks like it was overlooked. Resending. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc

2 1

[alsa-devel] [PATCH: alsa-lib] Fix a string format ambiguity
by Colin Guthrie 27 May '10

27 May '10

Hi, Simply patch to fix a format-string ambiguity. Meant to give this over before 1.0.23 but forgot. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

2 1

[alsa-devel] [PATCH] Add support for Thinkpad Edge conexant chip
by Jerone Young 27 May '10

27 May '10

This quirks in support for the Thinkpad Edge. Signed-off-by: Jerone Young <jerone.young(a)canonical.com> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index e863649..f4a2bd6 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -2975,6 +2975,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),

2 1

[alsa-devel] [PATCH 1/5] Remove unused io map functions.
by linux＠audioscience.com 27 May '10

27 May '10

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/hpios.c | 23 ----------------------- sound/pci/asihpi/hpios.h | 9 --------- 2 files changed, 0 insertions(+), 32 deletions(-) diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c index de615cf..742ee12 100644 --- a/sound/pci/asihpi/hpios.c +++ b/sound/pci/asihpi/hpios.c @@ -89,26 +89,3 @@ u16 hpios_locked_mem_free(struct consistent_dma_area *p_mem_area) void hpios_locked_mem_free_all(void) { } - -void __iomem *hpios_map_io(struct pci_dev *pci_dev, int idx, - unsigned int length) -{ - HPI_DEBUG_LOG(DEBUG, "mapping %d %s %08llx-%08llx %04llx len 0x%x\n", - idx, pci_dev->resource[idx].name, - (unsigned long long)pci_resource_start(pci_dev, idx), - (unsigned long long)pci_resource_end(pci_dev, idx), - (unsigned long long)pci_resource_flags(pci_dev, idx), length); - - if (!(pci_resource_flags(pci_dev, idx) & IORESOURCE_MEM)) { - HPI_DEBUG_LOG(ERROR, "not an io memory resource\n"); - return NULL; - } - - if (length > pci_resource_len(pci_dev, idx)) { - HPI_DEBUG_LOG(ERROR, "resource too small for requested %d \n", - length); - return NULL; - } - - return ioremap(pci_resource_start(pci_dev, idx), length); -} diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h index a62c3f1..370f39b 100644 --- a/sound/pci/asihpi/hpios.h +++ b/sound/pci/asihpi/hpios.h @@ -166,13 +166,4 @@ struct hpi_adapter { void __iomem *ap_remapped_mem_base[HPI_MAX_ADAPTER_MEM_SPACES]; }; -static inline void hpios_unmap_io(void __iomem *addr, - unsigned long size) -{ - iounmap(addr); -} - -void __iomem *hpios_map_io(struct pci_dev *pci_dev, int idx, - unsigned int length); - #endif -- 1.7.0.4

2 5

[alsa-devel] [PATCH 1/4] ALSA: usb-audio: parse more format descriptors with structs
by Daniel Mack 27 May '10

27 May '10

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/endpoint.c | 11 +++++++---- sound/usb/format.c | 20 ++++++++++---------- sound/usb/format.h | 7 ++++--- 3 files changed, 21 insertions(+), 17 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index ef07a6d..4887342 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -158,8 +158,9 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) int i, altno, err, stream; int format = 0, num_channels = 0; struct audioformat *fp = NULL; - unsigned char *fmt, *csep; + unsigned char *csep; int num, protocol; + struct uac_format_type_i_continuous_descriptor *fmt; dev = chip->dev; @@ -256,8 +257,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) dev->devnum, iface_no, altno); continue; } - if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || - ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { + if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || + ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) { snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; @@ -268,7 +269,9 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) * with the previous one, except for a larger packet size, but * is actually a mislabeled two-channel setting; ignore it. */ - if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 && + if (fmt->bNrChannels == 1 && + fmt->bSubframeSize == 2 && + altno == 2 && num == 3 && fp && fp->altsetting == 1 && fp->channels == 1 && fp->formats == SNDRV_PCM_FMTBIT_S16_LE && protocol == UAC_VERSION_1 && diff --git a/sound/usb/format.c b/sound/usb/format.c index b87cf87..caaa3ef 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -278,12 +278,11 @@ err: * parse the format type I and III descriptors */ static int parse_audio_format_i(struct snd_usb_audio *chip, - struct audioformat *fp, - int format, void *_fmt, + struct audioformat *fp, int format, + struct uac_format_type_i_continuous_descriptor *fmt, struct usb_host_interface *iface) { struct usb_interface_descriptor *altsd = get_iface_desc(iface); - struct uac_format_type_i_discrete_descriptor *fmt = _fmt; int protocol = altsd->bInterfaceProtocol; int pcm_format, ret; @@ -320,7 +319,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, switch (protocol) { case UAC_VERSION_1: fp->channels = fmt->bNrChannels; - ret = parse_audio_format_rates_v1(chip, fp, _fmt, 7); + ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7); break; case UAC_VERSION_2: /* fp->channels is already set in this case */ @@ -392,12 +391,12 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, } int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt, int stream, - struct usb_host_interface *iface) + int format, struct uac_format_type_i_continuous_descriptor *fmt, + int stream, struct usb_host_interface *iface) { int err; - switch (fmt[3]) { + switch (fmt->bFormatType) { case UAC_FORMAT_TYPE_I: case UAC_FORMAT_TYPE_III: err = parse_audio_format_i(chip, fp, format, fmt, iface); @@ -407,10 +406,11 @@ int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *f break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", - chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); + chip->dev->devnum, fp->iface, fp->altsetting, + fmt->bFormatType); return -1; } - fp->fmt_type = fmt[3]; + fp->fmt_type = fmt->bFormatType; if (err < 0) return err; #if 1 @@ -421,7 +421,7 @@ int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *f if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == UAC_FORMAT_TYPE_I && + if (fmt->bFormatType == UAC_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; diff --git a/sound/usb/format.h b/sound/usb/format.h index 8298c4e..387924f 100644 --- a/sound/usb/format.h +++ b/sound/usb/format.h @@ -1,8 +1,9 @@ #ifndef __USBAUDIO_FORMAT_H #define __USBAUDIO_FORMAT_H -int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt, int stream, - struct usb_host_interface *iface); +int snd_usb_parse_audio_format(struct snd_usb_audio *chip, + struct audioformat *fp, int format, + struct uac_format_type_i_continuous_descriptor *fmt, + int stream, struct usb_host_interface *iface); #endif /* __USBAUDIO_FORMAT_H */ -- 1.7.1

2 5

Re: [alsa-devel] [PATCH 17/17] sound: Add missing spin_unlock
by Takashi Iwai 27 May '10

27 May '10

At Wed, 26 May 2010 17:59:27 +0200 (CEST), Julia Lawall wrote: > > From: Julia Lawall <julia(a)diku.dk> > > Add a spin_unlock missing on the error path. > > The semantic match that finds this problem is as follows: > (http://coccinelle.lip6.fr/) > > // <smpl> > @@ > expression E1; > @@ > > * spin_lock(E1,...); > <+... when != E1 > if (...) { > ... when != E1 > * return ...; > } > ...+> > * spin_unlock(E1,...); > // </smpl> > > Signed-off-by: Julia Lawall <julia(a)diku.dk> Thanks, applied now. Takashi > --- > sound/mips/au1x00.c | 1 + > sound/oss/dmasound/dmasound_atari.c | 5 +++-- > 2 files changed, 4 insertions(+), 2 deletions(-) > > diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c > index 3e763d6..446cf97 100644 > --- a/sound/mips/au1x00.c > +++ b/sound/mips/au1x00.c > @@ -516,6 +516,7 @@ get the interrupt driven case to work efficiently */ > break; > if (i == 0x5000) { > printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n"); > + spin_unlock(&au1000->ac97_lock); > return 0; > } > > diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c > index 1f47741..13c2144 100644 > --- a/sound/oss/dmasound/dmasound_atari.c > +++ b/sound/oss/dmasound/dmasound_atari.c > @@ -1277,7 +1277,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy) > * (almost) like on the TT. > */ > write_sq_ignore_int = 0; > - return IRQ_HANDLED; > + goto out; > } > > if (!write_sq.active) { > @@ -1285,7 +1285,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy) > * the sq variables, so better don't do anything here. > */ > WAKE_UP(write_sq.sync_queue); > - return IRQ_HANDLED; > + goto out; > } > > /* Probably ;) one frame is finished. Well, in fact it may be that a > @@ -1322,6 +1322,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy) > /* We are not playing after AtaPlay(), so there > is nothing to play any more. Wake up a process > waiting for audio output to drain. */ > +out: > spin_unlock(&dmasound.lock); > return IRQ_HANDLED; > } >

1 0

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 27 May '10

27 May '10

The following changes since commit ad8332c1302bcb4f80d593fd3eb477be9d7f5604: ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies (2010-05-20 10:28:39 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 This fixes up the build of the i.MX SSI driver, a minor warning fix for SH and fix the dB scales for a bunch of Wolfson CODECs. Guennadi Liakhovetski (1): ASoC: fix uninitialised variable in siu_dai.c Mark Brown (3): ASoC: Fix dB scales for WM835x ASoC: Fix dB scales for WM8400 ASoC: Fix dB scales for WM8990 Stuart Longland (1): ASoC: Update Freescale i.MX SSI driver DMA parameter handling sound/soc/codecs/wm8350.c | 4 ++-- sound/soc/codecs/wm8400.c | 18 +++++++++--------- sound/soc/codecs/wm8990.c | 18 +++++++++--------- sound/soc/imx/imx-pcm-dma-mx2.c | 7 ++++--- sound/soc/sh/siu_dai.c | 2 ++ 5 files changed, 26 insertions(+), 23 deletions(-)

2 1

[alsa-devel] [PATCH 1/2] ASOC: S5PC100: Enable AC97 support
by Jassi Brar 27 May '10

27 May '10

From: Jassi Brar <jassi.brar(a)samsung.com> The S5PC100 has the AC97 controller same as S3C6410. Simply enable the options to build the drivers for S5PC100 also. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 4 ++-- sound/soc/s3c24xx/smdk_wm9713.c | 1 + 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 2a7cc22..aa112d3 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,6 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 || ARCH_S3C64XX + depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 select S3C64XX_DMA if ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to @@ -120,7 +120,7 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES config SND_SOC_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_S3C24XX_SOC && MACH_SMDK6410 + depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100) select SND_SOC_WM9713 select SND_S3C_SOC_AC97 help diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 24fd39f..7dd85e5 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -25,6 +25,7 @@ static struct snd_soc_card smdk; * Default CFG switch settings to use this driver: * * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off + * SMDKC100: Set CFG6 1-3 On, CFG7 1 On */ /* -- 1.6.2.5

3 2

[alsa-devel] Problem with snd_hda_intel from 05-27-2010 snapshot
by VDR User 27 May '10

27 May '10

Hi. I had been mostly successfully using the snd_hda_intel driver from the stable kernel 2.6.34 for HDMI audio with my Nvidia GT220 video card. In an attempt to see if any recent changes to snd_hda_intel would fix a problem I've been having with "pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed" I decided to download the latest snapshot. Unfortunately with these drivers I have no audio. All my settings, and everything have remained the same. The only noticable difference I've found is /sys/module/snd_hda_intel/parameters/probe_only is 0,0,0,0,0,0,0,0 in the snapshot drivers, and N,N,N,N,N,N,N,N in the 2.6.34 kernel drivers (which are compiled into the kernel). Any help? Thanks, Derek

2 2

[alsa-devel] [PATCH 2/2] ASOC: S5PV210: Enable AC97 support
by Jassi Brar 27 May '10

27 May '10

From: Jassi Brar <jassi.brar(a)samsung.com> The S5PV210 and S5PC110 has the AC97 controller same as S3C6410. Simply enable the options to build the drivers for S5PC110 and S5PV210 also. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 4 ++-- sound/soc/s3c24xx/smdk_wm9713.c | 2 ++ 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index aa112d3..292d817 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,6 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 + depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 select S3C64XX_DMA if ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to @@ -120,7 +120,7 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES config SND_SOC_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100) + depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_S3C_SOC_AC97 help diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 7dd85e5..5527b9e 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -26,6 +26,8 @@ static struct snd_soc_card smdk; * * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off * SMDKC100: Set CFG6 1-3 On, CFG7 1 On + * SMDKC110: Set CFGB10 1-2 Off, CFGB12 1-3 On + * SMDKV210: Set CFGB10 1-2 Off, CFGB12 1-3 On */ /* -- 1.6.2.5

1 0

[alsa-devel] [RFC PATCH v2 0/3] ep93xx i2s audio
by Ryan Mallon 27 May '10

27 May '10

The following patch series adds support for i2s audio on the ep93xx, with the Snapper CL15 used as an example implementation. I have created psuedo clocks for sclk and lrclk as suggested by Hartley/Mark. This moves the sdiv/lrdiv calculation into ep93xx_i2s_hw_parms. The ep93xx documentation implies that lrclk is gated for unused bits (ie if lrdiv > word length). If this is true for all ep93xx i2s setups then the lrclk/sclk's could be replaced by a single audio rate clock. I have fixed the enabling/disabling of the i2s peripheral, which corrects the problem I was having with playback sometimes resulting in noise. I have also made a number of changes suggested from the first round of patches. Ryan Mallon (3): ep93xx i2s audio driver ep93xx i2s core support ep93xx i2s audio snapper cl15 support arch/arm/mach-ep93xx/clock.c | 69 ++++- arch/arm/mach-ep93xx/core.c | 31 ++ arch/arm/mach-ep93xx/include/mach/ep93xx-regs.h | 10 + arch/arm/mach-ep93xx/include/mach/platform.h | 1 + arch/arm/mach-ep93xx/snappercl15.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/ep93xx/Kconfig | 18 + sound/soc/ep93xx/Makefile | 11 + sound/soc/ep93xx/ep93xx-i2s.c | 489 +++++++++++++++++++++++ sound/soc/ep93xx/ep93xx-i2s.h | 25 ++ sound/soc/ep93xx/ep93xx-pcm.c | 323 +++++++++++++++ sound/soc/ep93xx/ep93xx-pcm.h | 22 + sound/soc/ep93xx/snappercl15.c | 143 +++++++ 14 files changed, 1144 insertions(+), 1 deletions(-) create mode 100644 sound/soc/ep93xx/Kconfig create mode 100644 sound/soc/ep93xx/Makefile create mode 100644 sound/soc/ep93xx/ep93xx-i2s.c create mode 100644 sound/soc/ep93xx/ep93xx-i2s.h create mode 100644 sound/soc/ep93xx/ep93xx-pcm.c create mode 100644 sound/soc/ep93xx/ep93xx-pcm.h create mode 100644 sound/soc/ep93xx/snappercl15.c

4 6

[alsa-devel] [patch 0/5] kirkwood openrd client audio support - v3
by apatard＠mandriva.com 27 May '10

27 May '10

Hi, It's a new try to get merged kirkwood openrd client support. The changes are : - set window configuration according to dma address - namespaced all cs42l51 #define - modified cs42l51_dac_names array according to what proposed Mark Brown - use SOC_DOUBLE_TLV for mic boost Arnaud

3 9

[PATCH 3/4] eukrea_mbimx27: add audio codec
by Eric Bénard 26 May '10

26 May '10

Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- arch/arm/mach-mx2/eukrea_mbimx27-baseboard.c | 44 +++++++++++++++++++++++++- 1 files changed, 43 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-mx2/eukrea_mbimx27-baseboard.c b/arch/arm/mach-mx2/eukrea_mbimx27-baseboard.c index 34bdc26..471dbd2 100644 --- a/arch/arm/mach-mx2/eukrea_mbimx27-baseboard.c +++ b/arch/arm/mach-mx2/eukrea_mbimx27-baseboard.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2009 Eric Benard - eric(a)eukrea.com + * Copyright (C) 2009-2010 Eric Benard - eric(a)eukrea.com * * Based on pcm970-baseboard.c which is : * Copyright (C) 2008 Juergen Beisert (kernel(a)pengutronix.de) @@ -37,6 +37,8 @@ #include <mach/mmc.h> #include <mach/imx-uart.h> #include <mach/spi.h> +#include <mach/ssi.h> +#include <mach/audmux.h> #include "devices.h" @@ -93,6 +95,13 @@ static int eukrea_mbimx27_pins[] = { PD29_PF_CSPI1_SCLK, PD30_PF_CSPI1_MISO, PD31_PF_CSPI1_MOSI, + /* SSI4 */ +#if defined(CONFIG_SND_SOC_EUKREA_TLV320) + PC16_PF_SSI4_FS, + PC17_PF_SSI4_RXD | GPIO_PUEN, + PC18_PF_SSI4_TXD | GPIO_PUEN, + PC19_PF_SSI4_CLK, +#endif }; static const uint32_t eukrea_mbimx27_keymap[] = { @@ -287,6 +296,12 @@ static struct spi_board_info eukrea_mbimx27_spi_board_info[] __initdata = { }, }; +static struct i2c_board_info eukrea_mbimx27_i2c_devices[] = { + { + I2C_BOARD_INFO("tlv320aic23", 0x1a), + }, +}; + static int eukrea_mbimx27_spi_cs[] = {GPIO_PORTD | 28}; static struct spi_imx_master eukrea_mbimx27_spi_0_data = { @@ -303,6 +318,10 @@ static struct imxmmc_platform_data sdhc_pdata = { .dat3_card_detect = 1, }; +struct imx_ssi_platform_data eukrea_mbimx27_ssi_pdata = { + .flags = IMX_SSI_DMA, +}; + /* * system init for baseboard usage. Will be called by cpuimx27 init. * @@ -314,6 +333,24 @@ void __init eukrea_mbimx27_baseboard_init(void) mxc_gpio_setup_multiple_pins(eukrea_mbimx27_pins, ARRAY_SIZE(eukrea_mbimx27_pins), "MBIMX27"); +#if defined(CONFIG_SND_SOC_EUKREA_TLV320) + /* SSI unit master I2S codec connected to SSI_PINS_4*/ + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_TFSDIR | + MXC_AUDMUX_V1_PCR_TCLKDIR | + MXC_AUDMUX_V1_PCR_RFSDIR | + MXC_AUDMUX_V1_PCR_RCLKDIR | + MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + MXC_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) + ); + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + ); +#endif + mxc_register_device(&mxc_uart_device1, &uart_pdata[0]); mxc_register_device(&mxc_uart_device2, &uart_pdata[1]); #if !defined(MACH_EUKREA_CPUIMX27_USEUART4) @@ -323,6 +360,11 @@ void __init eukrea_mbimx27_baseboard_init(void) mxc_register_device(&mxc_fb_device, &eukrea_mbimx27_fb_data); mxc_register_device(&mxc_sdhc_device0, &sdhc_pdata); + i2c_register_board_info(0, eukrea_mbimx27_i2c_devices, + ARRAY_SIZE(eukrea_mbimx27_i2c_devices)); + + mxc_register_device(&imx_ssi_device0, &eukrea_mbimx27_ssi_pdata); + #if defined(CONFIG_TOUCHSCREEN_ADS7846) \ || defined(CONFIG_TOUCHSCREEN_ADS7846_MODULE) /* ADS7846 Touchscreen controller init */ -- 1.6.3.3 --------------010207020109090402010001--

1 0

[PATCH 2/4] soc/imx: add eukrea-tlv320
by Eric Bénard 26 May '10

26 May '10

Add the necessary files to support the TLV320AIC23B wired in I2S on our i.MX platforms. Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 8 +++ sound/soc/imx/Makefile | 2 + sound/soc/imx/eukrea-tlv320.c | 131 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 141 insertions(+), 0 deletions(-) create mode 100644 sound/soc/imx/eukrea-tlv320.c diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 7174b4c..3e27b16 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -11,3 +11,11 @@ config SND_IMX_SOC config SND_MXC_SOC_SSI tristate +config SND_SOC_EUKREA_TLV320 + bool "Eukrea TLV320" + depends on MACH_EUKREA_MBIMX27_BASEBOARD + select SND_IMX_SOC + select SND_SOC_TLV320AIC23 + help + Enable I2S based access to the TLV320AIC23B codec attached + to the SSI4 interface diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index 9f8bb92..28d4b1e 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -10,3 +10,5 @@ obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o # i.MX Machine Support snd-soc-phycore-ac97-objs := phycore-ac97.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o +snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o +obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c new file mode 100644 index 0000000..e368787 --- /dev/null +++ b/sound/soc/imx/eukrea-tlv320.c @@ -0,0 +1,131 @@ +/* + * eukrea-tlv320.c -- SoC audio for eukrea_cpuimxXX in I2S mode + * + * Copyright 2010 Eric Bénard, Eukréa Electromatique <eric(a)eukrea.com> + * + * based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * which is Copyright 2009 Simtec Electronics + * and on sound/soc/imx/phycore-ac97.c which is + * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer(a)pengutronix.de> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> + +#include "../codecs/tlv320aic23.h" +#include "imx-ssi.h" + +#define CODEC_CLOCK 12000000 + +static int eukrea_tlv320_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 the CODEC as the bus clock master, I2S */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + /* Set the CODEC as the bus clock master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret) { + pr_err("%s: failed setting codec sysclk\n", __func__); + return ret; + } + + return 0; +} + +static struct snd_soc_ops eukrea_tlv320_snd_ops = { + .hw_params = eukrea_tlv320_hw_params, +}; + +static struct snd_soc_dai_link eukrea_tlv320_dai = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .codec_dai = &tlv320aic23_dai, + .ops = &eukrea_tlv320_snd_ops, +}; + +static struct snd_soc_card eukrea_tlv320 = { + .name = "cpuimx-audio", + .platform = &imx_soc_platform, + .dai_link = &eukrea_tlv320_dai, + .num_links = 1, +}; + +static struct snd_soc_device eukrea_tlv320_snd_devdata = { + .card = &eukrea_tlv320, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *eukrea_tlv320_snd_device; + +static int __init eukrea_tlv320_init(void) +{ + int ret; + + if (!machine_is_eukrea_cpuimx27()) + /* return happy. We might run on a totally different machine */ + return 0; + + eukrea_tlv320_snd_device = platform_device_alloc("soc-audio", -1); + if (!eukrea_tlv320_snd_device) + return -ENOMEM; + + eukrea_tlv320_dai.cpu_dai = &imx_ssi_pcm_dai[0]; + + platform_set_drvdata(eukrea_tlv320_snd_device, &eukrea_tlv320_snd_devdata); + eukrea_tlv320_snd_devdata.dev = &eukrea_tlv320_snd_device->dev; + ret = platform_device_add(eukrea_tlv320_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(eukrea_tlv320_snd_device); + } + + return ret; +} + +static void __exit eukrea_tlv320_exit(void) +{ + platform_device_unregister(eukrea_tlv320_snd_device); +} + +module_init(eukrea_tlv320_init); +module_exit(eukrea_tlv320_exit); + +MODULE_AUTHOR("Eric Bénard <eric(a)eukrea.com>"); +MODULE_DESCRIPTION("CPUIMX ALSA SoC driver"); +MODULE_LICENSE("GPL"); -- 1.6.3.3 --------------010207020109090402010001 Content-Type: text/x-patch; name="0003-eukrea_mbimx27-add-audio-codec.patch" Content-Transfer-Encoding: 8bit Content-Disposition: attachment; filename="0003-eukrea_mbimx27-add-audio-codec.patch"

1 0

Re: [alsa-devel] CX20582 (Pebble) and Thinkpad x100e...
by Tobin Davis 26 May '10

26 May '10

I'm forwarding this on to the alsa-devel mailing list as I am not personally maintaining this driver at this time. If you could also download alsa-info.sh from http://alsa-project.org, run it, and post the results to the mailing list that would be great. -- Tobin Davis "Look! There! Evil!.. pure and simple, total evil from the Eighth Dimension!" -- Buckaroo Banzai On Mon, 2010-05-24 at 22:20 +0300, Hakan Kızılhan wrote: > Pototskiy Akex > Takashi Iwai > Tobin Davis > > Hi, > > I am Hakan from Turkey. If no drawback, i want to ask for a hardware problem. > > I have AMD based Thinkpad x100e and this machine have Conexant CX20582 > (Pebble) sound card. > Using Ubuntu 10.04 and microphones doesn't work. I have readed following bug: > > https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/549289 > > options snd-hda-intel model="olpc-xo-1_5" enables output > jack(headphones) but microphones not. > i dont capture sound anyway. > > i saw "sound/pci/hda/patch_conexant.c" related patches for different models. > > So, is it posible to patch this file for Thinkpad x100e model? > May i send hardware specific information for patch? And what should i send? > > Thanks a lot... > > ____________ > Hakan Kizilhan

2 1

Re: [alsa-devel] [patch 4/5] orion/kirkwood: Add i2s support
by Arnaud Patard 26 May '10

26 May '10

Nicolas Pitre <nico(a)fluxnic.net> writes: Hi, > On Tue, 25 May 2010, apatard(a)mandriva.com wrote: > >> This patch enables support for the i2s controller available on orion/kirkwood platforms >> >> Signed-off-by: Arnaud Patard <apatard(a)mandriva.com> > > The actual Orion processor family has no I2S/audio support. It would be > better if all the audio related stuff was renamed to "kirkwood" and > "KIRKWOOD_*" instead. I've been told there was some orion with audio. As I've never looked at the orion family in details, I assumed this is right and I've changed the naming accordingly. I can change again but it would be nice to have an agreement here. Who's/what's right ? orion can or can not have i2s audio ? Arnaud

1 0

[alsa-devel] [PATCH 0/7] ASoC: TWL4030: put the codec to off when not active
by Peter Ujfalusi 26 May '10

26 May '10

Hello, The following series at the end will let the twl4030 codec to use BIAS_OFF instead of BIAS_STANDBY. The difference in power consumption is about 0.5mA. To achieve this: - The regcache has been reseted to codec default - The codec initialization has been optimized, it is no longer writes all 73 registers at startup, but only modifies few selected one. - The power related code has been cleaned up, and optimized - Support added for machine drivers to select the offset cancellation path - debug support for checking the codec default registers (machine drivers can ask for checking, but shall be disabled in production). I guess that's it. The driver has been tested on a custom board with twl5031. It passed all of our internal test cases covering much of the codec features. --- Peter Ujfalusi (7): ASoC: TWL4030: Revisit codec defaults ASoC: TWL4030: Remove wrapper for power down ASoC: TWL4030: Make offset cancellation path configurable ASoC: TWL4030: Optimize the power up sequence ASoC: TWL4030: Helper to check chip default registers ASoC: TWL4030: Correct the ARXR2_APGA_CTL chip default ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use sound/soc/codecs/twl4030.c | 286 ++++++++++++++++++++++++-------------------- sound/soc/codecs/twl4030.h | 2 + 2 files changed, 157 insertions(+), 131 deletions(-)

3 16

Re: [alsa-devel] [patch 4/5] orion/kirkwood: Add i2s support
by saeed bishara 26 May '10

26 May '10

On Wed, May 26, 2010 at 3:31 AM, Nicolas Pitre <nico(a)fluxnic.net> wrote: > On Tue, 25 May 2010, apatard(a)mandriva.com wrote: > >> This patch enables support for the i2s controller available on orion/kirkwood platforms >> >> Signed-off-by: Arnaud Patard <apatard(a)mandriva.com> > > The actual Orion processor family has no I2S/audio support. It would be > better if all the audio related stuff was renamed to "kirkwood" and > "KIRKWOOD_*" instead. This i2s was first included in orion90L (6183). saeed

1 0

Re: [alsa-devel] [patch 0/5] kirkwood openrd client audio support - v3
by Mark Brown 26 May '10

26 May '10

On Tue, May 25, 2010 at 09:10:54PM -0400, Nicolas Pitre wrote: > I don't mind if you wish to carry it through the ASOC tree. The bulk of > it is ASOC related anyway. However I think that "orion" names should be > "kirkwood" throughout instead, as pure Orion chips have no audio > interfaces. Can I add an ack from you for the patches with the name change, then?

1 0

[alsa-devel] [PATCH]alsa:patch_realtek.c iMac9,1 sound fixes.
by Justin P. Mattock 25 May '10

25 May '10

What seemed to be a simple little task of getting the microphone to work, ended up becoming a bigger task than I had set out. (oh-well, good learning experience). Anyways below you will find an updated patch to the iMac9,1 machine which fixes some issues that I found.(keep in mind this is still a work in progress). First issue: With the original patch, I've noticed by unmuting the mic (and even having it muted), there is a distorted("Noise") coming from the internal speakers, even when the headphones are plugged in. What my finding's revealed is: /* Mic (rear) pin: input vref at 80% */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, _From the original patch. Looking at codec#0 0x18/0x1a is listed as: 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: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90100141: [Fixed] Speaker at Int N/A Conn = Unknown, Color = Unknown DefAssociation = 0x4, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x41: OUT VREF_50 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 seems this Node is listed as: [Fixed] Speaker while 0x15 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: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x018b3020: [Jack] Line In at Ext Rear Conn = Comb, Color = Blue DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x01: VREF_50 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 is [Jack] Line In at Ext Rear. (looking at the other apple products as examples I came up with the fix below). Second issue: alc885_mbp_4ch_modes The original patch does a good job with the HP pin automute function, but from what I noticed is I would have to manually change the channel form 2 to 4 after plugging the headphones in. And not to mention having odd moments to where I was jamming out with the headphones on, then later realized I had sound blasting out of the speakers as well. My findings revealed that changing alc885_mbp_4ch_modes to alc885_mba21_ch_modes and setting - spec->autocfg.speaker_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x18; gets the automute function when the headphones plugged in working flawlessly(and the no need to manually change the channel number afterwards). Third issue: alc885_imac91_mixer There probably doesnt need to be anything changed with this (esspecially if your one to like lots of sliders),but my findings revealed that mac osx only has a master on the top right, another switch on itunes, and then a slider for the mic. So the changes I did below try and mimic osx as much as possible (only thing I had an issue with is just having one mute switch on the master, instead of having two(still investigating)). fourth issue: alc882_capture_source I endeded up creating alc889A_imac91_capture_source() only because looking at alc882_capture_source I see that the mic is set to 0x1 while this works, I also noticed that adding 0x1 and 0x01 and testing that 0x1 somehow stops working, and 0x01 works(so I figured 0x01 was more of the alpha of the numbers(still need to figure out where that valuse is)). In any case the microphone does work with the original, and with the below patch, but both still record not as clean(lots of "Noise", which I would like to look into too). Note: using alsamixer -Va reveals the capture switches. Overall the original patch is good, the below patch does (hopefully)address some issues, if it's too much of a change I can go back, and just adjust the verb pins on the original and just call it that..(main thing over here is I don't have the "Noise" coming out of the speakers(and music sounds good)... cheers.. Signed-off-by: Justin P. Mattock <justinmattock(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 84 +++++++++++++++++++--------------------- 1 files changed, 40 insertions(+), 44 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 886d8e4..d8adea3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6937,6 +6937,14 @@ static struct hda_input_mux alc889A_mb31_capture_source = { }, }; +static struct hda_input_mux alc889A_imac91_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x01 }, + { "Line", 0x2 }, /* Not sure! */ + }, +}; + /* * 2ch mode */ @@ -7398,15 +7406,8 @@ static struct snd_kcontrol_new alc885_macmini3_mixer[] = { }; static struct snd_kcontrol_new alc885_imac91_mixer[] = { - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_MUTE ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), { } /* end */ }; @@ -7907,61 +7908,56 @@ static struct hda_verb alc885_mbp3_init_verbs[] = { /* iMac 9,1 */ static struct hda_verb alc885_imac91_init_verbs[] = { - /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* HP Pin: output 0 (0x0c) */ + /* Internal Speaker Pin (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP Pin: Rear */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - /* Internal Speakers: output 0 (0x0d) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC880_HP_EVENT | AC_USRSP_EN)}, + /* Line in Rear */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Mic (rear) pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Front Mic pin: input vref at 80% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin: use output 1 when in LineOut mode */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + /* Rear mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Input mixer2 */ + /* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Input mixer3 */ + /* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* ADC1: mute amp left and right */ + /* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC2: mute amp left and right */ + /* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC3: mute amp left and right */ + /* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - { } }; @@ -8030,7 +8026,7 @@ static void alc885_imac91_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; } @@ -9539,14 +9535,14 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc885_imac24_init_hook, }, [ALC885_IMAC91] = { - .mixers = { alc885_imac91_mixer, alc882_chmode_mixer }, + .mixers = {alc885_imac91_mixer}, .init_verbs = { alc885_imac91_init_verbs, alc880_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mbp_4ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes), - .input_mux = &alc882_capture_source, + .channel_mode = alc885_mba21_ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), + .input_mux = &alc889A_imac91_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, .unsol_event = alc_automute_amp_unsol_event, -- 1.7.1.rc1.21.gf3bd6

2 2

[alsa-devel] [PATCH] ALSA: usb/caiaq: fix Traktor Kontrol X1 ABS_HAT2X axis
by Daniel Mack 25 May '10

25 May '10

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/input.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index 8bbfbfd..dcb6207 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -171,7 +171,7 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, input_report_abs(input_dev, ABS_HAT0Y, (buf[4] << 8) | buf[5]); input_report_abs(input_dev, ABS_HAT1X, (buf[12] << 8) | buf[13]); input_report_abs(input_dev, ABS_HAT1Y, (buf[2] << 8) | buf[3]); - input_report_abs(input_dev, ABS_HAT2X, (buf[15] << 8) | buf[15]); + input_report_abs(input_dev, ABS_HAT2X, (buf[14] << 8) | buf[15]); input_report_abs(input_dev, ABS_HAT2Y, (buf[0] << 8) | buf[1]); input_report_abs(input_dev, ABS_HAT3X, (buf[10] << 8) | buf[11]); input_report_abs(input_dev, ABS_HAT3Y, (buf[6] << 8) | buf[7]); -- 1.7.1

2 2

[alsa-devel] [PATCH] ALSA: hda: Fix model quirk for Dell M1730
by Daniel Chen 25 May '10

25 May '10

Hi, Attached please find a patch for sound-2.6 master HEAD that fixes capture out-of-the-box for Dell M1730. This patch is applicable to all 2.6.32, 2.6.33, and 2.6.34 stable trees. Best, -Dan

2 1

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for Toshiba A100-259
by Daniel Chen 25 May '10

25 May '10

Hi, Attached please find a patch for sound-2.6 based on my previous patch (ALSA: hda: Use LPIB for Acer Aspire 5110) specifying the LPIB quirk for an additional SSID to make capture work correctly in an installed distribution. This patch is applicable to 2.6.32, 2.6.33, and 2.6.34 stable trees (some with fuzz). Best, -Dan

3 2

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for Acer Aspire 5110
by Daniel Chen 25 May '10

25 May '10

Hi, Attached please find a patch against sound-2.6 master HEAD specifying the LPIB quirk for an additional SSID to make capture work correctly in an installed distribution. This patch is applicable to 2.6.32, 2.6.33, and 2.6.34 stable trees (some with fuzz). Best, -Dan

2 1

[alsa-devel] [PATCH] aw2-alsa.c: use pci_ids.h defines and fix checkpatch.pl noise
by H Hartley Sweeten 25 May '10

25 May '10

Use the VENDOR/DEVICE ids provided in pci_ids.h instead of creating local ids of the same values. Also, fix the following checkpatch.pl warnings: WARNING: Use #include <linux/io.h> instead of <asm/io.h> WARNING: unnecessary whitespace before a quoted newline Signed-off-by: H Hartley Sweeten <hsweeten(a)visionengravers.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> --- diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 67921f9..c150022 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -26,7 +26,7 @@ #include <linux/slab.h> #include <linux/interrupt.h> #include <linux/delay.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> @@ -44,9 +44,6 @@ MODULE_LICENSE("GPL"); /********************************* * DEFINES ********************************/ -#define PCI_VENDOR_ID_SAA7146 0x1131 -#define PCI_DEVICE_ID_SAA7146 0x7146 - #define CTL_ROUTE_ANALOG 0 #define CTL_ROUTE_DIGITAL 1 @@ -165,7 +162,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = { - {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0, + {PCI_VENDOR_ID_PHILIPS, PCI_DEVICE_ID_PHILIPS_SAA7146, 0, 0, 0, 0, 0}, {0} }; @@ -419,7 +416,7 @@ static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - snd_printdd(KERN_DEBUG "aw2: Playback_open \n"); + snd_printdd(KERN_DEBUG "aw2: Playback_open\n"); runtime->hw = snd_aw2_playback_hw; return 0; } @@ -435,7 +432,7 @@ static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - snd_printdd(KERN_DEBUG "aw2: Capture_open \n"); + snd_printdd(KERN_DEBUG "aw2: Capture_open\n"); runtime->hw = snd_aw2_capture_hw; return 0; }

2 1

Re: [alsa-devel] snd_intel_hda : broken microphone
by mutenmaster＠tiscali.it 25 May '10

25 May '10

ok you can find attached my codec0 output data. I hope this helps. if you or anyone else need any other piece of info, please ask. ---- Messaggio originale---- Apparently (I assume) this is not for all via sound chips so it's best to expalin what your setup is. The output from lspci does not tell the story on the codec at times. The best output is from /proc/asound/card0/codec#0 On Wed, 2010-05-19 at 11:33 +0200, mutenmaster(a)tiscali.it wrote: > hello people, > > i am really determined in letting you know, for the > third time, > that my front mic input does not work. i do not know if the > rear > one works, i suspect not, but it's not important, because > before > the following commit, the front mic input worked: > > $ sudo git bisect > bad > f5271101faf1655d862849f42518c2a88ef394fb > is the first bad commit > > commit f5271101faf1655d862849f42518c2a88ef394fb > > Author: Lydia Wang > <lydiawang(a)viatech.com.cn> > Date: Sat Oct 10 19:07: > 35 2009 +0800 > > > ALSA HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE > control type > > Enter > low power state if AA-Path volume is muted. > > > Signed-off-by: > Lydia Wang <lydiawang@viatech. com.cn> > > Signed-off-by: Logan Li > <loganli(a)viatech.com.cn> > Signed-off-by: > Takashi Iwai <tiwai@suse. > de> > > :040000 040000 > ab1ae14ff67f0b08848faeb3ad68def7f3e727f0 > > 0bfbfde11fe32e87e8edbdb3c4d03c2d30370393 M sound > > i'm copypasting > the lspci ouput, so that you can have a general idea > of what kind of > machine i have. please let me know if there is > any other info you need > to solve this bug. > > more details on the problem: > doing > $ arecord -f cd > test.wav > > produces a file of 44 bytes. it's as if no microphone is > attached > to the jack. this behaviour has been evolving from the commit > > i showed you forth. at the beginning, a file with actual data was > > produced, but no audio came out of the speakers when reproducing > it. > going on with the commits, the file became 44 bytes long. > > yours, > > vasilis > > -- > > $ lspci -vvv > > 00:00.0 Host bridge: Advanced Micro Devices > [AMD] RS780 Host Bridge Alternate > Subsystem: ASUSTeK Computer Inc. > Device 83a2 > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- > VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- > Status: Cap+ > 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ > >SERR- <PERR- INTx- > Latency: 0 > Capabilities: [c4] HyperTransport: > Slave or Primary Interface > Command: BaseUnitID=0 UnitCnt=12 MastHost- > DefDir- DUL- > Link Control 0: CFlE- CST- CFE- <LkFail- Init+ EOC- TXO- > <CRCErr=0 IsocEn+ LSEn- ExtCTL- 64b- > Link Config 0: MLWI=16bit > DwFcIn- MLWO=16bit DwFcOut- LWI=16bit DwFcInEn- LWO=16bit DwFcOutEn- > > Link Control 1: CFlE- CST- CFE- <LkFail+ Init- EOC+ TXO+ <CRCErr=0 > IsocEn- LSEn- ExtCTL- 64b- > Link Config 1: MLWI=8bit DwFcIn- MLWO=8bit > DwFcOut- LWI=8bit DwFcInEn- LWO=8bit DwFcOutEn- > Revision ID: 3.00 > > Link Frequency 0: [b] > Link Error 0: <Prot- <Ovfl- <EOC- CTLTm- > Link > Frequency Capability 0: 200MHz+ 300MHz- 400MHz+ 500MHz- 600MHz+ 800MHz+ > 1.0GHz+ 1.2GHz- 1.4GHz- 1.6 GHz- Vend- > Feature Capability: IsocFC- > LDTSTOP+ CRCTM- ECTLT- 64bA- UIDRD- > Link Frequency 1: 200MHz > Link > Error 1: <Prot- <Ovfl- <EOC- CTLTm- > Link Frequency Capability 1: > 200MHz- 300MHz- 400MHz- 500MHz- 600MHz- 800MHz- 1.0GHz- 1.2GHz- 1.4GHz- > 1.6 GHz- Vend- > Error Handling: PFlE- OFlE- PFE- OFE- EOCFE- RFE- > CRCFE- SERRFE- CF- RE- PNFE- ONFE- EOCNFE- RNFE- CRCNFE- SERRNFE- > > Prefetchable memory behind bridge Upper: 00-00 > Bus Number: 00 > > Capabilities: [54] HyperTransport: UnitID Clumping > Capabilities: [40] > HyperTransport: Retry Mode > Capabilities: [9c] HyperTransport: #1a > > Capabilities: [f8] HyperTransport: #1c > > 00:02.0 PCI bridge: Advanced > Micro Devices [AMD] RS780 PCI to PCI bridge (ext gfx port 0) (prog-if > 00 [Normal decode]) > 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 > Bus: > primary=00, secondary=01, subordinate=01, sec-latency=0 > I/O behind > bridge: 0000d000- 0000dfff > Memory behind bridge: fea00000-feafffff > > Prefetchable memory behind bridge: 00000000d0000000- 00000000dfffffff > > Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- > <MAbort- <SERR- <PERR- > BridgeCtl: Parity+ SERR+ NoISA- VGA+ MAbort- > >Reset- FastB2B- > PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- > > Capabilities: [50] Power Management version 3 > Flags: PMEClk- DSI- D1- > D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+, D3cold+) > Status: D0 PME- > Enable- DSel=0 DScale=0 PME- > Capabilities: [58] Express (v2) Root Port > (Slot+), MSI 00 > DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency > L0s <64ns, L1 <1us > ExtTag+ RBE+ FLReset- > DevCtl: Report errors: > Correctable- Non-Fatal- Fatal- Unsupported- > RlxdOrd+ ExtTag- > PhantFunc- AuxPwr- NoSnoop+ > MaxPayload 128 bytes, MaxReadReq 128 > bytes > DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- > TransPend- > LnkCap: Port #0, Speed 5GT/s, Width x16, ASPM L0s L1, > Latency L0 <64ns, L1 <1us > ClockPM- Suprise- LLActRep+ BwNot+ > > LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk+ > > ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- > LnkSta: Speed 2.5 GT/s, > Width x16, TrErr- Train- SlotClk+ DLActive+ BWMgmt+ ABWMgmt- > SltCap: > AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug- Surpise- > Slot # 2, > PowerLimit 75.000000; Interlock- NoCompl+ > SltCtl: Enable: AttnBtn- > PwrFlt- MRL- PresDet- CmdCplt- HPIrq- LinkChg- > Control: AttnInd > Unknown, PwrInd Unknown, Power- Interlock- > SltSta: Status: AttnBtn- > PowerFlt- MRL- CmdCplt- PresDet+ Interlock- > Changed: MRL- PresDet+ > LinkState+ > RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- > PMEIntEna- CRSVisible- > RootCap: CRSVisible- > RootSta: PME ReqID > 0000, PMEStatus- PMEPending- > DevCap2: Completion Timeout: Not > Supported, TimeoutDis- ARIFwd- > DevCtl2: Completion Timeout: 50us to > 50ms, TimeoutDis- ARIFwd- > LnkCtl2: Target Link Speed: 5GT/s, > EnterCompliance- SpeedDis-, Selectable De- emphasis: -3.5dB > Transmit > Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- > > Compliance De- emphasis: -6dB > LnkSta2: Current De-emphasis Level: > -3.5dB > Capabilities: [a0] Message Signalled Interrupts: Mask- 64bit- > Count=1/1 Enable+ > Address: fee0f00c Data: 4159 > Capabilities: [b0] > Subsystem: ASUSTeK Computer Inc. Device 83a2 > Capabilities: [b8] > HyperTransport: MSI Mapping Enable+ Fixed+ > Capabilities: [100] Vendor > Specific Information <?> > Capabilities: [110] Virtual Channel <?> > > Kernel driver in use: pcieport-driver > > 00:0a.0 PCI bridge: Advanced > Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 5) (prog-if 00 > [Normal decode]) > 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 > Bus: > primary=00, secondary=02, subordinate=02, sec- latency=0 > I/O behind > bridge: 0000e000-0000efff > Memory behind bridge: feb00000-febfffff > > Prefetchable memory behind bridge: 00000000fdf00000-00000000fdffffff > > Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- > <MAbort- <SERR- <PERR- > BridgeCtl: Parity+ SERR+ NoISA+ VGA- MAbort- > >Reset- FastB2B- > PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- > > Capabilities: [50] Power Management version 3 > Flags: PMEClk- DSI- D1- > D2- AuxCurrent=0mA PME(D0+,D1- ,D2-,D3hot+,D3cold+) > Status: D0 PME- > Enable- DSel=0 DScale=0 PME- > Capabilities: [58] Express (v2) Root Port > (Slot+), MSI 00 > DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency > L0s <64ns, L1 <1us > ExtTag+ RBE+ FLReset- > DevCtl: Report errors: > Correctable- Non-Fatal- Fatal- Unsupported- > RlxdOrd+ ExtTag- > PhantFunc- AuxPwr- NoSnoop+ > MaxPayload 128 bytes, MaxReadReq 128 > bytes > DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- > TransPend- > LnkCap: Port #1, Speed 5GT/s, Width x1, ASPM L0s L1, > Latency L0 <64ns, L1 <1us > ClockPM- Suprise- LLActRep+ BwNot+ > > LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk+ > > ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- > LnkSta: Speed 2.5GT/s, > Width x1, TrErr- Train- SlotClk+ DLActive+ BWMgmt+ ABWMgmt- > SltCap: > AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug- Surpise- > Slot # a, > PowerLimit 25.000000; Interlock- NoCompl+ > SltCtl: Enable: AttnBtn- > PwrFlt- MRL- PresDet- CmdCplt- HPIrq- LinkChg- > Control: AttnInd > Unknown, PwrInd Unknown, Power- Interlock- > SltSta: Status: AttnBtn- > PowerFlt- MRL- CmdCplt- PresDet+ Interlock- > Changed: MRL- PresDet+ > LinkState+ > RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- > PMEIntEna- CRSVisible- > RootCap: CRSVisible- > RootSta: PME ReqID > 0000, PMEStatus- PMEPending- > DevCap2: Completion Timeout: Not > Supported, TimeoutDis- ARIFwd- > DevCtl2: Completion Timeout: 50us to > 50ms, TimeoutDis- ARIFwd- > LnkCtl2: Target Link Speed: 2.5GT/s, > EnterCompliance- SpeedDis-, Selectable De-emphasis: -6dB > Transmit > Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- > > Compliance De- emphasis: -6dB > LnkSta2: Current De-emphasis Level: > -6dB > Capabilities: [a0] Message Signalled Interrupts: Mask- 64bit- > Count=1/1 Enable+ > Address: fee0f00c Data: 4161 > Capabilities: [b0] > Subsystem: ASUSTeK Computer Inc. Device 83a2 > Capabilities: [b8] > HyperTransport: MSI Mapping Enable+ Fixed+ > Capabilities: [100] Vendor > Specific Information <?> > Capabilities: [110] Virtual Channel <?> > > Kernel driver in use: pcieport-driver > > 00:11.0 SATA controller: ATI > Technologies Inc SB700/SB800 SATA Controller [AHCI mode] (prog-if 01 > [AHCI 1.0]) > Subsystem: ASUSTeK Computer Inc. Device 8389 > Control: > I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- > SERR+ FastB2B- DisINTx- > Status: Cap+ 66MHz+ UDF- FastB2B- ParErr- > DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: > 64, Cache Line Size: 64 bytes > Interrupt: pin A routed to IRQ 22 > > Region 0: I/O ports at c000 [size=8] > Region 1: I/O ports at b000 > [size=4] > Region 2: I/O ports at a000 [size=8] > Region 3: I/O ports at > 9000 [size=4] > Region 4: I/O ports at 8000 [size=16] > Region 5: Memory > at fe9ffc00 (32-bit, non-prefetchable) [size=1K] > Capabilities: [60] > Power Management version 2 > Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA > PME(D0-,D1-,D2-,D3hot-,D3cold-) > Status: D0 PME- Enable- DSel=0 > DScale=0 PME- > Capabilities: [70] SATA HBA <?> > Kernel driver in use: > ahci > > 00:12.0 USB Controller: ATI Technologies Inc SB700/SB800 USB > OHCI0 Controller (prog-if 10 [OHCI]) > Subsystem: ASUSTeK Computer Inc. > Device 8389 > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- > VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- > Status: Cap- > 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- > >SERR- <PERR- INTx- > Latency: 64, Cache Line Size: 64 bytes > Interrupt: > pin A routed to IRQ 16 > Region 0: Memory at fe9fe000 (32-bit, non- > prefetchable) [size=4K] > Kernel driver in use: ohci_hcd > Kernel > modules: ohci- hcd > > 00:12.1 USB Controller: ATI Technologies Inc SB700 > USB OHCI1 Controller (prog-if 10 [OHCI]) > Subsystem: ASUSTeK Computer > Inc. Device 8389 > Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ > VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- > Status: Cap- > 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- > >SERR- <PERR- INTx- > Latency: 64, Cache Line Size: 64 bytes > Interrupt: > pin A routed to IRQ 16 > Region 0: Memory at fe9fd000 (32-bit, non- > prefetchable) [size=4K] > Kernel driver in use: ohci_hcd > Kernel > modules: ohci-hcd > > 00:12.2 USB Controller: ATI Technologies Inc > SB700/SB800 USB EHCI Controller (prog-if 20 [EHCI]) > Subsystem: ASUSTeK > Computer Inc. Device 8389 > Control: I/O- Mem+ BusMaster+ SpecCycle- > MemWINV+ VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- > Status: > Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- > <MAbort- >SERR- <PERR- INTx- > Latency: 64, Cache Line Size: 64 bytes > > Interrupt: pin B routed to IRQ 17 > Region 0: Memory at fe9ff800 (32- > bit, non-prefetchable) [size=256] > Capabilities: [c0] Power Management > version 2 > Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+, > D3hot+,D3cold-) > Status: D0 PME-Enable- DSel=0 DScale=0 PME- > Bridge: > PM- B3+ > Capabilities: [e4] Debug port: BAR=1 offset=00e0 > Kernel > driver in use: ehci_hcd > Kernel modules: ehci-hcd > > 00:13.0 USB > Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller (prog- > if 10 [OHCI]) > Subsystem: ASUSTeK Computer Inc. Device 8389 > Control: > I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- > SERR+ FastB2B- DisINTx- > Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- > DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: > 64, Cache Line Size: 64 bytes > Interrupt: pin A routed to IRQ 18 > > Region 0: Memory at fe9fc000 (32-bit, non-prefetchable) [size=4K] > > Kernel driver in use: ohci_hcd > Kernel modules: ohci-hcd > > 00:13.1 USB > Controller: ATI Technologies Inc SB700 USB OHCI1 Controller (prog-if 10 > [OHCI]) > Subsystem: ASUSTeK Computer Inc. Device 8389 > Control: I/O+ > Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR+ > FastB2B- DisINTx- > Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- > DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: > 64, Cache Line Size: 64 bytes > Interrupt: pin A routed to IRQ 18 > > Region 0: Memory at fe9fb000 (32-bit, non- prefetchable) [size=4K] > > Kernel driver in use: ohci_hcd > Kernel modules: ohci-hcd > > 00:13.2 USB > Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller (prog- > if 20 [EHCI]) > Subsystem: ASUSTeK Computer Inc. Device 8389 > Control: > I/O- Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- > SERR+ FastB2B- DisINTx- > Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- > DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: > 64, Cache Line Size: 64 bytes > Interrupt: pin B routed to IRQ 19 > > Region 0: Memory at fe9ff400 (32-bit, non-prefetchable) [size=256] > > Capabilities: [c0] Power Management version 2 > Flags: PMEClk- DSI- D1+ > D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) > Status: D0 PME- > Enable- DSel=0 DScale=0 PME- > Bridge: PM- B3+ > Capabilities: [e4] > Debug port: BAR=1 offset=00e0 > Kernel driver in use: ehci_hcd > Kernel > modules: ehci-hcd > > 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus > Controller (rev 3c) > Subsystem: ASUSTeK Computer Inc. Device 8389 > > Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- > Stepping- SERR- FastB2B- DisINTx+ > Status: Cap+ 66MHz+ UDF- FastB2B- > ParErr- DEVSEL=medium >TAbort- <TAbort+ <MAbort- >SERR- <PERR- INTx- > > Capabilities: [b0] HyperTransport: MSI Mapping Enable- Fixed+ > > 00:14.1 > IDE interface: ATI Technologies Inc SB700/SB800 IDE Controller (prog-if > 8a [Master SecP PriP]) > Subsystem: ASUSTeK Computer Inc. Device 8389 > > Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- > Stepping- SERR- FastB2B- DisINTx- > Status: Cap+ 66MHz+ UDF- FastB2B- > ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > > Latency: 64 > Interrupt: pin A routed to IRQ 16 > Region 0: I/O ports at > 01f0 [size=8] > Region 1: I/O ports at 03f4 [size=1] > Region 2: I/O > ports at 0170 [size=8] > Region 3: I/O ports at 0374 [size=1] > Region 4: > I/O ports at ff00 [size=16] > Capabilities: [70] Message Signalled > Interrupts: Mask- 64bit- Count=1/1 Enable- > Address: 00000000 Data: > 0000 > Kernel driver in use: pata_atiixp > > 00:14.2 Audio device: ATI > Technologies Inc SBx00 Azalia (Intel HDA) > Subsystem: ASUSTeK Computer > Inc. Device 836c > 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: 64, Cache Line Size: 64 bytes > Interrupt: > pin ? routed to IRQ 16 > Region 0: Memory at fe9f4000 (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+) > Status: D0 PME-Enable- DSel=0 DScale=0 PME- > Kernel driver > in use: HDA Intel > Kernel modules: snd-hda-intel > > 00:14.3 ISA bridge: > ATI Technologies Inc SB700/SB800 LPC host controller > Subsystem: > ASUSTeK Computer Inc. Device 8389 > Control: I/O+ Mem+ BusMaster+ > SpecCycle+ MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- > > Status: Cap- 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- > <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: 0 > > 00:14.4 PCI bridge: > ATI Technologies Inc SBx00 PCI to PCI Bridge (prog-if 01 [Subtractive > decode]) > Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- > ParErr- Stepping- SERR+ FastB2B- DisINTx- > Status: Cap- 66MHz+ UDF- > FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- > INTx- > Latency: 64 > Bus: primary=00, secondary=03, subordinate=03, sec- > latency=64 > I/O behind bridge: 0000f000- 00000fff > Memory behind bridge: > fff00000-000fffff > Prefetchable memory behind bridge: fff00000-000fffff > > Secondary status: 66MHz- FastB2B+ ParErr- DEVSEL=medium >TAbort- > <TAbort- <MAbort+ <SERR- <PERR- > BridgeCtl: Parity+ SERR+ NoISA+ VGA- > MAbort- >Reset- FastB2B- > PriDiscTmr- SecDiscTmr- DiscTmrStat- > DiscTmrSERREn- > > 00:14.5 USB Controller: ATI Technologies Inc > SB700/SB800 USB OHCI2 Controller (prog-if 10 [OHCI]) > Subsystem: > ASUSTeK Computer Inc. Device 8389 > Control: I/O- Mem+ BusMaster+ > SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- > > Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- > <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: 64, Cache Line Size: 64 > bytes > Interrupt: pin C routed to IRQ 18 > Region 0: Memory at fe9fa000 > (32-bit, non-prefetchable) [size=4K] > Kernel driver in use: ohci_hcd > > Kernel modules: ohci-hcd > > 00:18.0 Host bridge: Advanced Micro Devices > [AMD] Family 10h [Opteron, Athlon64, Sempron] HyperTransport > Configuration > 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- > Capabilities: [80] HyperTransport: Host or > Secondary Interface > Command: WarmRst+ DblEnd- DevNum=0 ChainSide- > HostHide+ Slave- <EOCErr- DUL- > Link Control: CFlE- CST- CFE- <LkFail- > Init+ EOC- TXO- <CRCErr=0 IsocEn+ LSEn+ ExtCTL- 64b- > Link Config: > MLWI=16bit DwFcIn- MLWO=16bit DwFcOut- LWI=16bit DwFcInEn- LWO=16bit > DwFcOutEn- > Revision ID: 3.00 > Link Frequency: [b] > Link Error: > <Prot- <Ovfl- <EOC- CTLTm- > Link Frequency Capability: 200MHz+ 300MHz- > 400MHz+ 500MHz- 600MHz+ 800MHz+ 1.0GHz+ 1.2GHz+ 1.4 GHz- 1.6GHz- Vend- > > Feature Capability: IsocFC+ LDTSTOP+ CRCTM- ECTLT- 64bA+ UIDRD- ExtRS- > UCnfE- > > 00:18.1 Host bridge: Advanced Micro Devices [AMD] Family 10h > [Opteron, Athlon64, Sempron] Address Map > 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- > > 00: 18.2 Host bridge: Advanced > Micro Devices [AMD] Family 10h [Opteron, Athlon64, Sempron] DRAM > Controller > 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- > Kernel modules: amd64_edac_mod > > 00:18.3 Host bridge: Advanced > Micro Devices [AMD] Family 10h [Opteron, Athlon64, Sempron] > Miscellaneous Control > 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- > Capabilities: [f0] Secure device <?> > > 00: > 18.4 Host bridge: Advanced Micro Devices [AMD] Family 10h [Opteron, > Athlon64, Sempron] Link Control > 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- > > 01:00.0 VGA compatible > controller: ATI Technologies Inc Device 954f (prog-if 00 [VGA > controller]) > Subsystem: ASUSTeK Computer Inc. Device 02a8 > 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 18 > Region 0: > Memory at d0000000 (64-bit, prefetchable) [size=256M] > Region 2: Memory > at feaf0000 (64-bit, non-prefetchable) [size=64K] > Region 4: I/O ports > at d000 [size=256] > Expansion ROM at feac0000 [disabled] [size=128K] > > Capabilities: [50] Power Management version 3 > Flags: PMEClk- DSI- D1+ > D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) > Status: D0 PME- > Enable- DSel=0 DScale=0 PME- > Capabilities: [58] Express (v2) Legacy > Endpoint, MSI 00 > DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency > L0s <4us, L1 unlimited > ExtTag+ AttnBtn- AttnInd- PwrInd- RBE+ > FLReset- > DevCtl: Report errors: Correctable- Non-Fatal- Fatal- > Unsupported- > RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop+ > > MaxPayload 128 bytes, MaxReadReq 128 bytes > DevSta: CorrErr- > UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- > LnkCap: Port #0, > Speed 2.5GT/s, Width x16, ASPM L0s L1, Latency L0 <64ns, L1 <1us > > ClockPM- Suprise- LLActRep- BwNot- > LnkCtl: ASPM Disabled; RCB 64 > bytes Disabled- Retrain- CommClk+ > ExtSynch- ClockPM- AutWidDis- > BWInt- AutBWInt- > LnkSta: Speed 2.5GT/s, Width x16, TrErr- Train- > SlotClk+ DLActive- BWMgmt- ABWMgmt- > DevCap2: Completion Timeout: Not > Supported, TimeoutDis- > DevCtl2: Completion Timeout: 50us to 50ms, > TimeoutDis- > LnkCtl2: Target Link Speed: 2.5GT/s, EnterCompliance- > SpeedDis-, Selectable De-emphasis: -6dB > Transmit Margin: Normal > Operating Range, EnterModifiedCompliance- ComplianceSOS- > Compliance > De-emphasis: -6dB > LnkSta2: Current De-emphasis Level: -6dB > > Capabilities: [a0] Message Signalled Interrupts: Mask- 64bit+ Count=1/1 > Enable- > Address: 0000000000000000 Data: 0000 > Capabilities: [100] > Vendor Specific Information <?> > > 01:00.1 Audio device: ATI Technologies > Inc Device aa38 > Subsystem: ASUSTeK Computer Inc. Device aa38 > 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 B routed to IRQ 28 > Region 0: > Memory at feaec000 (64-bit, non- prefetchable) [size=16K] > Capabilities: > [50] Power Management version 3 > Flags: PMEClk- DSI- D1+ D2+ > AuxCurrent=0mA PME(D0-,D1- , D2-,D3hot-,D3cold-) > Status: D0 PME-Enable- > DSel=0 DScale=0 PME- > Capabilities: [58] Express (v2) Legacy Endpoint, > MSI 00 > DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <4us, > L1 unlimited > ExtTag+ AttnBtn- AttnInd- PwrInd- RBE+ FLReset- > > DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- > > RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop+ > MaxPayload 128 bytes, > MaxReadReq 128 bytes > DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- > AuxPwr- TransPend- > LnkCap: Port #0, Speed 2.5GT/s, Width x16, ASPM > L0s L1, Latency L0 <64ns, L1 <1us > ClockPM- Suprise- LLActRep- BwNot- > > LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk+ > > ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- > LnkSta: Speed 2.5GT/s, > Width x16, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- > DevCap2: > Completion Timeout: Not Supported, TimeoutDis- > DevCtl2: Completion > Timeout: 50us to 50ms, TimeoutDis- > LnkCtl2: Target Link Speed: 2.5 > GT/s, EnterCompliance- SpeedDis-, Selectable De- emphasis: -6dB > > Transmit Margin: Normal Operating Range, EnterModifiedCompliance- > ComplianceSOS- > Compliance De- emphasis: -6dB > LnkSta2: Current De- > emphasis Level: -6dB > Capabilities: [a0] Message Signalled Interrupts: > Mask- 64bit+ Count=1/1 Enable+ > Address: 00000000fee0f00c Data: 4189 > > Capabilities: [100] Vendor Specific Information <?> > Kernel driver in > use: HDA Intel > Kernel modules: snd-hda-intel > > 02:00.0 Ethernet > controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express > Gigabit Ethernet controller (rev 03) > Subsystem: ASUSTeK Computer Inc. > Device 83a3 > 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 27 > Region 0: I/O ports at e800 [size=256] > Region > 2: Memory at fdfff000 (64- bit, prefetchable) [size=4K] > Region 4: > Memory at fdff8000 (64- bit, prefetchable) [size=16K] > Expansion ROM at > febf0000 [disabled] [size=64K] > Capabilities: [40] Power Management > version 3 > Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=375mA PME(D0+, D1+, > D2+,D3hot+,D3cold+) > Status: D0 PME- Enable- DSel=0 DScale=0 PME- > > Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Count=1/1 > Enable+ > Address: 00000000fee0f00c Data: 4179 > Capabilities: [70] > Express (v2) Endpoint, MSI 01 > DevCap: MaxPayload 256 bytes, PhantFunc > 0, Latency L0s <512ns, L1 <64us > ExtTag- AttnBtn- AttnInd- PwrInd- > RBE+ FLReset- > DevCtl: Report errors: Correctable- Non-Fatal- Fatal- > Unsupported- > RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- > > MaxPayload 128 bytes, MaxReadReq 4096 bytes > DevSta: CorrErr+ > UncorrErr- FatalErr- UnsuppReq+ AuxPwr+ TransPend- > LnkCap: Port #0, > Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <512ns, L1 <64us > > ClockPM+ Suprise- LLActRep- BwNot- > LnkCtl: ASPM Disabled; RCB 64 > bytes Disabled- Retrain- CommClk+ > ExtSynch- ClockPM- AutWidDis- > BWInt- AutBWInt- > LnkSta: Speed 2.5 GT/s, Width x1, TrErr- Train- > SlotClk+ DLActive- BWMgmt- ABWMgmt- > DevCap2: Completion Timeout: Not > Supported, TimeoutDis+ > DevCtl2: Completion Timeout: 50us to 50ms, > TimeoutDis- > LnkCtl2: Target Link Speed: 2.5GT/s, EnterCompliance- > SpeedDis-, Selectable De-emphasis: -6dB > Transmit Margin: Normal > Operating Range, EnterModifiedCompliance- ComplianceSOS- > Compliance > De- emphasis: -6dB > LnkSta2: Current De-emphasis Level: -6dB > > Capabilities: [ac] MSI-X: Enable- Mask- TabSize=4 > Vector table: BAR=4 > offset=00000000 > PBA: BAR=4 offset=00000800 > Capabilities: [cc] Vital > Product Data <?> > Capabilities: [100] Advanced Error Reporting > UESta: > DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- > UnsupReq- ACSVoil- > UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- > UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSVoil- > UESvrt: DLP+ SDES+ > TLP- FCP+ CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- > ACSVoil- > CESta: RxErr- BadTLP- BadDLLP- Rollover- Timeout- > NonFatalErr+ > CESta: RxErr- BadTLP- BadDLLP- Rollover- Timeout- > NonFatalErr+ > AERCap: First Error Pointer: 00, GenCap+ CGenEn- ChkCap+ > ChkEn- > Capabilities: [140] Virtual Channel <?> > Capabilities: [160] > Device Serial Number 00- 00-00-00-00-00-00-00 > Kernel driver in use: > r8169 > > > > > Risparmia con Tutto Incluso Light: telefono + adsl 8 mega a soli 14,95 € al mese. > > Gratis la Sim Tiscali Mobile con 25 euro di traffico! > > L'offerta è valida solo se attivi entro il 20/05/10 > > http: //abbonati.tiscali. it/telefono-adsl/prodotti/tc/tuttoincluso_light/?WT. mc_id=01fw > _______________________________________________ Alsa- devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa- project. org/mailman/listinfo/alsa-devel Trova Casa sul portale immobiliare n°1 in Italia http://tiscali.casa.it Trova Casa sul portale immobiliare n°1 in Italia http://tiscali.casa.it

1 0

[alsa-devel] Type safe Haskell interface to ALSA sequencer
by Henning Thielemann 25 May '10

25 May '10

Dear ALSA developers, I read the ALSA-Sequencer-API-Doxygen documentation, various ALSA-MIDI-How-Tos and their examples and succeeded to work with ALSA sequencers in Haskell. However, I'm still uncertain about the meaning of the system components and how they interact. It is extensively documented, what works, but not, what does not work. Whenever I do a mistake, ALSA leaves me with an unspecific "operation not permitted" or "invalid argument" error code. Thus I try to encode conditions in Haskell types. For instance once I got an "Invalid argument" error code and only after some investigation, I found that I opened the sequencer in Output mode but later called event_input. (Actually I only wanted to output something, but used event_input for synchronization using Echo events. Unfortunately the intention of Echo messages is even not explained in the Doxygen stuff.) In order to prevent another occurrence of this problem, I defined distinct Haskell types for sequencers depending on the input mode. Now, event_input is only allowed on sequencers that were opened in Output or Duplex mode. However I wondered, why I could create writeable ports in a output-only sequencer. Does this make sense? Would it be reasonable to forbid writeable ports in output sequencers and readable ports in input sequencers by Haskell types? Is there a difference between cap_read|cap_write and cap_duplex? I also wondered why there can be several ports in one sequencer, since multiple clients can connect to one port, and event_input and event_output have no port argument. Further I wondered why queues are not bound to a specific port. Can queues be shared amongst ports? Is it sensible to have multiple queues in one sequencer? Why has sync_output_queue no queue argument? The more I think about it, it seems to me that a port is more an identifier, a part of an address, than a mailbox. I could use ports as tags, that tell which part of an application to control. A queue looks like a mail company and I can use the same sender and receiver addresses across different companies. If this is true, then I think it would be worth to mention this in the API documentation. In another example I called sync_output_queue in order to wait for all events being sent, but it aborted with "Operation not permitted", whereas I have seen example code where it works. What are the preconditions to calling sync_output_queue? I really like to see a picture of how sequencers, clients, ports, port_infos, queues, timers, events are related.

2 1

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: add support for Akai MPD16
by Krzysztof Foltman 25 May '10

25 May '10

The decoding/encoding is based on own reverse-engineering. Both control and data ports are handled. Writing to control port supports SysEx events only, as this is the only type of messages that MPD16 recognizes. Signed-off-by: Krzysztof Foltman <wdev(a)foltman.com> diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 2c1558c..d669feb 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -434,7 +434,7 @@ static void snd_usbmidi_maudio_broken_running_status_input( u8 cin = buffer[i] & 0x0f; struct usbmidi_in_port *port = &ep->ports[cable]; int length; - + length = snd_usbmidi_cin_length[cin]; if (cin == 0xf && buffer[i + 1] >= 0xf8) ; /* realtime msg: no running status change */ @@ -628,13 +628,13 @@ static struct usb_protocol_ops snd_usbmidi_standard_ops = { static struct usb_protocol_ops snd_usbmidi_midiman_ops = { .input = snd_usbmidi_midiman_input, - .output = snd_usbmidi_standard_output, + .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_midiman_packet, }; static struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = { .input = snd_usbmidi_maudio_broken_running_status_input, - .output = snd_usbmidi_standard_output, + .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; @@ -645,6 +645,110 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = { }; /* + * AKAI MPD16 protocol: + * + * For control port (endpoint 1): + * ============================== + * One or more chunks consisting of first byte of (0x10 | msg_len) and then a + * SysEx message (msg_len=9 bytes long): + * + * F0 47 62 60 aa bb cc xx F7 (xx = (aa + bb + cc) & 0x7F) + * + * aa bb cc + * -------- + * 10 01 7E edit mode on? + * 10 01 00 edit mode off? + * 10 04 7F select bank 1 + * 10 04 00 select bank 0 + * 11 pp nn set pad pp to send note nn + * 12 pp ss set sensitivity on pad pp to ss + * 13 00 cc set slider CC number to cc + * 14 00 cc set MIDI channel to cc + * 15 01 pp request note number for pad pp (returns 20 pp nn) + * 15 02 pp request sensitivity for pad pp (returns 21 pp ss) + * 15 03 00 request slider CC number (returns 22 00 cc) + * 15 05 00 request MIDI channel number (returns 23 00 cc) + * + * For data port (endpoint 2): + * =========================== + * One or more chunks consisting of first byte of (0x20 | msg_len) and then a + * MIDI message (msg_len bytes long) + * + * Messages sent: + * 21 FE (active sense) + * 23 90 xx xx (note on) + * 23 Ax xx xx (polyphonic pressure) + * 23 Bx xx xx (control change) + */ +static void snd_usbmidi_akai_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) +{ + unsigned int pos = 0; + unsigned int len = (unsigned int)buffer_length; + while (pos < len) { + int port = (buffer[pos] >> 4) - 1; + unsigned int msg_len = buffer[pos] & 0x0f; + pos++; + if (pos + msg_len <= len && (port == 0 || port == 1)) + snd_usbmidi_input_data(ep, 0, &buffer[pos], msg_len); + pos += msg_len; + } +} + +static void snd_usbmidi_akai_output(struct snd_usb_midi_out_endpoint *ep, + struct urb *urb) +{ + uint8_t *msg; + uint8_t *plen = &ep->ports[0].data[0]; /* length so far */ + int count; + uint8_t tmp; + + if (!ep->ports[0].active) + return; + + msg = urb->transfer_buffer + urb->transfer_buffer_length; + + /* only try adding more data when there's space for at least 1 SysEx */ + while (urb->transfer_buffer_length + 16 < ep->max_transfer) { + count = snd_rawmidi_transmit(ep->ports[0].substream, &tmp, 1); + if (count < 1) { + ep->ports[0].active = 0; + return; + } + if (ep->ports[0].state == STATE_UNKNOWN) { + /* only accept SysEx'es, skip everything else */ + if (tmp != 0xF0) + continue; + /* start accumulating SysEx data */ + *plen = 0x00; + ep->ports[0].state = STATE_SYSEX_0; + } + if (*plen > 14) { + /* SysEx too long, drop it and start over */ + ep->ports[0].state = STATE_UNKNOWN; + continue; + } + msg[++(*plen)] = tmp; + if (tmp == 0xF7) { + /* add total size (w/hdr) to length-to-transfer */ + urb->transfer_buffer_length += 1 + *plen; + /* set port number nibble in the header */ + msg[0] = 0x10 | *plen; + msg += 1 + *plen; + + /* go back to waiting for SysEx */ + ep->ports[0].state = STATE_UNKNOWN; + continue; + } + } +} + +static struct usb_protocol_ops snd_usbmidi_akai_ops = { + .input = snd_usbmidi_akai_input, + .output = snd_usbmidi_akai_output, +}; + +/* * Novation USB MIDI protocol: number of data bytes is in the first byte * (when receiving) (+1!) or in the second byte (when sending); data begins * at the third byte. @@ -1434,6 +1538,9 @@ static struct port_info { EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), + /* Akai MPD16 */ + CONTROL_PORT(0x09e8, 0x0062, 0, "%s Control"), + EXTERNAL_PORT(0x09e8, 0x0062, 1, "%s MIDI"), /* Access Music Virus TI */ EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"), PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0, @@ -1707,7 +1814,7 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi, snd_usbmidi_switch_roland_altsetting(umidi); if (endpoint[0].out_ep || endpoint[0].in_ep) - return 0; + return 0; intf = umidi->iface; if (!intf || intf->num_altsetting < 1) @@ -1745,7 +1852,7 @@ static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi* umidi, struct snd_usb_midi_endpoint_info* endpoints) { int err, i; - + err = snd_usbmidi_detect_endpoints(umidi, endpoints, MIDI_MAX_ENDPOINTS); for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { if (endpoints[i].out_ep) @@ -2035,6 +2142,12 @@ int snd_usbmidi_create(struct snd_card *card, umidi->usb_protocol_ops = &snd_usbmidi_cme_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_AKAI: + umidi->usb_protocol_ops = &snd_usbmidi_akai_ops; + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + /* endpoint 1 is input-only */ + endpoints[1].out_cables = 0; + break; default: snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); err = -ENXIO; diff --git a/sound/usb/midi.h b/sound/usb/midi.h index 2089ec9..2fca80b 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -37,6 +37,8 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_MIDI_CME, data is NULL */ +/* for QUIRK_MIDI_AKAI, data is NULL */ + int snd_usbmidi_create(struct snd_card *card, struct usb_interface *iface, struct list_head *midi_list, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 91ddef3..f8797f6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1973,6 +1973,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* AKAI devices */ +{ + USB_DEVICE(0x09e8, 0x0062), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "AKAI", + .product_name = "MPD16", + .ifnum = 0, + .type = QUIRK_MIDI_AKAI, + } +}, + /* TerraTec devices */ { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 136e5b4..b45e54c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -289,6 +289,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, + [QUIRK_MIDI_AKAI] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index d679e72..06ebf24 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -74,6 +74,7 @@ enum quirk_type { QUIRK_MIDI_FASTLANE, QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, + QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, -- 1.7.0.4

3 5

[alsa-devel] MMAP in ALSA
by Varadarajan, Srikanth 25 May '10

25 May '10

Hi All, I am new to the list and ALSA. For the hardware I am working with, we expose it to ALSA as one card capable of playing one PCM stream. If we have one application using the hardware through ALSA, other applications are unable to get access to it. So, we want to have PulseAudio on top and use the pulse virtual device to render audio through ALSA. Is this a recommended way of having multiple applications access one audio HW simultaneously? Is there any other way of handling multiple applications get access to the same hardware? Also, why are mmap and munmap operations required in ALSA? Right now, we do not have mmap and munmap operations exposed. Though the hardware constraints have "SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID" defined. Looks like, if we have to PulseAudio on top, mmap operations are required in ALSA. If there is a close enough ALSA implementation in the current Linux kernel that I can lookup, please point me to that. Any help/suggestions are welcome. Thanks, Srikanth

3 2

[alsa-devel] Query about code organization for kernel driver.
by Anksh Agarwal 25 May '10

25 May '10

Hi Everyone, I am currently working on a custom ARM based board and have succeeded in running alsa on my target. In this reference I have some basic queries and would be great if someone could give me pointers.This would help me in rorganizing the code as per acceptance from the group. since we planning to support a list of devices I wanted to keep the interface independent of playback/record session.this helps in switching to different devices in middle of playback. At any point a single device would be enabled. So i have another driver snd_device.c which uses sysfs interface to get device index or name to enable/disable device. The reason for this was that we could reuse a lot of code from the alsa-plugin(maemo). But then will it be right to put the device enable/disable code under sound/soc/xxxx/snd_device.c path, or can we split and place it seperately under sound/driver/xxxx/snd_device.c or any other suggested directory? i am not aware how to split the audio path enable/disable code from the soc driver and which would be ideal directory to place them, since there is hardly any interaction between playback and device enable/disable routines? I think the device enable/disable code outside sound/ directory wont be a good idea as its to be specifically used by alsa playback/record only. please suggest as this would help me a lot. Also let me know if any more info would help to get to a better conclusion. Really appreciate all the help. Thanks, Anksh

2 1

[alsa-devel] [PATCH] ASoC: fix uninitialised variable in siu_dai.c
by Guennadi Liakhovetski 24 May '10

24 May '10

Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- Strangely, the compiler doesn't warn about this one, but I really don't see how this can be correct. Am I being blind? sound/soc/sh/siu_dai.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index d86ee1b..eeed5ed 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -588,6 +588,8 @@ static int siu_dai_prepare(struct snd_pcm_substream *substream, ret = siu_dai_spbstart(port_info); if (ret < 0) goto fail; + } else { + ret = 0; } port_info->play_cap |= self; -- 1.6.2.4

3 2

[alsa-devel] [PATCH] ASoC: Update Freescale i.MX SSI driver
by Stuart Longland 24 May '10

24 May '10

Hi all, Attached is a patch that corrects the i.MX SSI driver so that it compiles against the latest ASoC tree. I'm able to compile the driver with this patch, but haven't done any testing beyond this point... the update is quite naïve, based on changes I had seen to other drivers. I hope to give it a try on Monday, but in the meantime, here's the patch for others to try. -- Stuart Longland (aka Redhatter, VK4MSL) .'''. Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : . . . . . . . . . . . . . . . . . . . . . . .'.' http://dev.gentoo.org/~redhatter :.' I haven't lost my mind... ...it's backed up on a tape somewhere.

3 2

Re: [alsa-devel] [PATCH v3] NUC900/audio: Add nuc900 audio driver
by Mark Brown 24 May '10

24 May '10

On Sat, May 22, 2010 at 09:46:31PM +0800, Wan ZongShun wrote: > Dear Liam & Mark, > > This patch fixed some issues according to your comments. > Here and now, I adjusted the driver and called it v3 for resubmitting. Since your patch has already been applied please submit incremental updates rather than a revised version of the full patch.

2 1

[alsa-devel] [PATCH 024/199] Documentation/sound/alsa/emu10k1-jack.txt: Checkpatch cleanup
by Andrea Gelmini 23 May '10

23 May '10

Documentation/sound/alsa/emu10k1-jack.txt:3: ERROR: trailing whitespace Documentation/sound/alsa/emu10k1-jack.txt:39: ERROR: trailing whitespace Documentation/sound/alsa/emu10k1-jack.txt:58: ERROR: trailing whitespace Signed-off-by: Andrea Gelmini <andrea.gelmini(a)gelma.net> --- Documentation/sound/alsa/emu10k1-jack.txt | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/emu10k1-jack.txt b/Documentation/sound/alsa/emu10k1-jack.txt index 751d450..3cc6548 100644 --- a/Documentation/sound/alsa/emu10k1-jack.txt +++ b/Documentation/sound/alsa/emu10k1-jack.txt @@ -1,6 +1,6 @@ This document is a guide to using the emu10k1 based devices with JACK for low latency, multichannel recording functionality. All of my recent work to allow -Linux users to use the full capabilities of their hardware has been inspired +Linux users to use the full capabilities of their hardware has been inspired by the kX Project. Without their work I never would have discovered the true power of this hardware. @@ -36,7 +36,7 @@ The 16 input ports are connected to the 16 physical inputs. Contrary to popular belief, all emu10k1 cards are multichannel cards. Which of these input channels have physical inputs connected to them depends on the card model. Trial and error is highly recommended; the pinout diagrams -for the card have been reverse engineered by some enterprising kX users and are +for the card have been reverse engineered by some enterprising kX users and are available on the internet. Meterbridge is helpful here, and the kX forums are packed with useful information. @@ -55,7 +55,7 @@ JACK Epilog FXBUS2(nr) -------------------------------------------- capture_1 asio14 FXBUS2(0xe) capture_2 asio15 FXBUS2(0xf) -capture_3 asio0 FXBUS2(0x0) +capture_3 asio0 FXBUS2(0x0) ~capture_4 Center EXTOUT(0x11) // mapped to by Center ~capture_5 LFE EXTOUT(0x12) // mapped to by LFE capture_6 asio3 FXBUS2(0x3) -- 1.7.1.251.gf80a2

1 1

[alsa-devel] [PATCH 092/199] Documentation/sound/oss/ALS: Checkpatch cleanup
by Andrea Gelmini 23 May '10

23 May '10

Documentation/sound/oss/ALS:15: ERROR: trailing whitespace Documentation/sound/oss/ALS:40: ERROR: trailing whitespace Documentation/sound/oss/ALS:55: ERROR: trailing whitespace Documentation/sound/oss/ALS:56: ERROR: trailing whitespace Signed-off-by: Andrea Gelmini <andrea.gelmini(a)gelma.net> --- Documentation/sound/oss/ALS | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/oss/ALS b/Documentation/sound/oss/ALS index d01ffbf..282a2e1 100644 --- a/Documentation/sound/oss/ALS +++ b/Documentation/sound/oss/ALS @@ -12,7 +12,7 @@ SB16 driver). To use an ALS sound card under Linux, enable the following options as modules in the sound configuration section of the kernel config: - 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support - - FM synthesizer (YM3812/OPL-3) support + - FM synthesizer (YM3812/OPL-3) support - standalone MPU401 support may be required for some cards; for the ALS-007, when using isapnptools, it is required Since the ALS-007/100/200 are PnP cards, ISAPnP support should probably be @@ -37,7 +37,7 @@ requires they be different. For "sb", "io", "irq" and "dma" should be set to the same values used to configure the audio section of the card with isapnp. "dma16" should be explicitly set to "-1" for an ALS-007 since this card does not have a 16 bit dma channel; if not specified the kernel will -default to using channel 5 anyway which will cause audio not to work. +default to using channel 5 anyway which will cause audio not to work. "mpu_io" should be set to 0. The "io" parameter of the "opl3" module should also agree with the setting used by isapnp. To get the MPU401 interface working on an ALS-007 card, the "mpu401" module will be required since this @@ -52,8 +52,8 @@ The resulting sound driver will provide the following capabilities: - Software selection of record source (line in, CD, FM, mic, master) - Record and playback of midi data via the external MPU-401 - Playback of midi data using inbuilt FM synthesizer - - Control of the ALS-007 mixer via any OSS-compatible mixer programs. - Controls available are Master (L&R), Line in (L&R), CD (L&R), + - Control of the ALS-007 mixer via any OSS-compatible mixer programs. + Controls available are Master (L&R), Line in (L&R), CD (L&R), DSP/PCM/audio out (L&R), FM (L&R) and Mic in (mono). Jonathan Woithe -- 1.7.1.251.gf80a2

1 13

[alsa-devel] snd_intel_hda : broken microphone
by mutenmaster＠tiscali.it 23 May '10

23 May '10

hello people, i am really determined in letting you know, for the third time, that my front mic input does not work. i do not know if the rear one works, i suspect not, but it's not important, because before the following commit, the front mic input worked: $ sudo git bisect bad f5271101faf1655d862849f42518c2a88ef394fb is the first bad commit commit f5271101faf1655d862849f42518c2a88ef394fb Author: Lydia Wang <lydiawang(a)viatech.com.cn> Date: Sat Oct 10 19:07: 35 2009 +0800 ALSA HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type Enter low power state if AA-Path volume is muted. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse. de> :040000 040000 ab1ae14ff67f0b08848faeb3ad68def7f3e727f0 0bfbfde11fe32e87e8edbdb3c4d03c2d30370393 M sound i'm copypasting the lspci ouput, so that you can have a general idea of what kind of machine i have. please let me know if there is any other info you need to solve this bug. more details on the problem: doing $ arecord -f cd test.wav produces a file of 44 bytes. it's as if no microphone is attached to the jack. this behaviour has been evolving from the commit i showed you forth. at the beginning, a file with actual data was produced, but no audio came out of the speakers when reproducing it. going on with the commits, the file became 44 bytes long. yours, vasilis -- $ lspci -vvv 00:00.0 Host bridge: Advanced Micro Devices [AMD] RS780 Host Bridge Alternate Subsystem: ASUSTeK Computer Inc. Device 83a2 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ >SERR- <PERR- INTx- Latency: 0 Capabilities: [c4] HyperTransport: Slave or Primary Interface Command: BaseUnitID=0 UnitCnt=12 MastHost- DefDir- DUL- Link Control 0: CFlE- CST- CFE- <LkFail- Init+ EOC- TXO- <CRCErr=0 IsocEn+ LSEn- ExtCTL- 64b- Link Config 0: MLWI=16bit DwFcIn- MLWO=16bit DwFcOut- LWI=16bit DwFcInEn- LWO=16bit DwFcOutEn- Link Control 1: CFlE- CST- CFE- <LkFail+ Init- EOC+ TXO+ <CRCErr=0 IsocEn- LSEn- ExtCTL- 64b- Link Config 1: MLWI=8bit DwFcIn- MLWO=8bit DwFcOut- LWI=8bit DwFcInEn- LWO=8bit DwFcOutEn- Revision ID: 3.00 Link Frequency 0: [b] Link Error 0: <Prot- <Ovfl- <EOC- CTLTm- Link Frequency Capability 0: 200MHz+ 300MHz- 400MHz+ 500MHz- 600MHz+ 800MHz+ 1.0GHz+ 1.2GHz- 1.4GHz- 1.6GHz- Vend- Feature Capability: IsocFC- LDTSTOP+ CRCTM- ECTLT- 64bA- UIDRD- Link Frequency 1: 200MHz Link Error 1: <Prot- <Ovfl- <EOC- CTLTm- Link Frequency Capability 1: 200MHz- 300MHz- 400MHz- 500MHz- 600MHz- 800MHz- 1.0GHz- 1.2GHz- 1.4GHz- 1.6GHz- Vend- Error Handling: PFlE- OFlE- PFE- OFE- EOCFE- RFE- CRCFE- SERRFE- CF- RE- PNFE- ONFE- EOCNFE- RNFE- CRCNFE- SERRNFE- Prefetchable memory behind bridge Upper: 00-00 Bus Number: 00 Capabilities: [54] HyperTransport: UnitID Clumping Capabilities: [40] HyperTransport: Retry Mode Capabilities: [9c] HyperTransport: #1a Capabilities: [f8] HyperTransport: #1c 00:02.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (ext gfx port 0) (prog-if 00 [Normal decode]) 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 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 I/O behind bridge: 0000d000-0000dfff Memory behind bridge: fea00000-feafffff Prefetchable memory behind bridge: 00000000d0000000-00000000dfffffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- <SERR- <PERR- BridgeCtl: Parity+ SERR+ NoISA- VGA+ MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME- Enable- DSel=0 DScale=0 PME- Capabilities: [58] Express (v2) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag+ RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 5GT/s, Width x16, ASPM L0s L1, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep+ BwNot+ LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x16, TrErr- Train- SlotClk+ DLActive+ BWMgmt+ ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug- Surpise- Slot # 2, PowerLimit 75.000000; Interlock- NoCompl+ SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet- CmdCplt- HPIrq- LinkChg- Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet+ Interlock- Changed: MRL- PresDet+ LinkState+ RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna- CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- DevCap2: Completion Timeout: Not Supported, TimeoutDis- ARIFwd- DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis- ARIFwd- LnkCtl2: Target Link Speed: 5GT/s, EnterCompliance- SpeedDis-, Selectable De-emphasis: -3.5dB Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -3.5dB Capabilities: [a0] Message Signalled Interrupts: Mask- 64bit- Count=1/1 Enable+ Address: fee0f00c Data: 4159 Capabilities: [b0] Subsystem: ASUSTeK Computer Inc. Device 83a2 Capabilities: [b8] HyperTransport: MSI Mapping Enable+ Fixed+ Capabilities: [100] Vendor Specific Information <?> Capabilities: [110] Virtual Channel <?> Kernel driver in use: pcieport-driver 00:0a.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 5) (prog-if 00 [Normal decode]) 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 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 I/O behind bridge: 0000e000-0000efff Memory behind bridge: feb00000-febfffff Prefetchable memory behind bridge: 00000000fdf00000-00000000fdffffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- <SERR- <PERR- BridgeCtl: Parity+ SERR+ NoISA+ VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME- Enable- DSel=0 DScale=0 PME- Capabilities: [58] Express (v2) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag+ RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #1, Speed 5GT/s, Width x1, ASPM L0s L1, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep+ BwNot+ LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive+ BWMgmt+ ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug- Surpise- Slot # a, PowerLimit 25.000000; Interlock- NoCompl+ SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet- CmdCplt- HPIrq- LinkChg- Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet+ Interlock- Changed: MRL- PresDet+ LinkState+ RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna- CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- DevCap2: Completion Timeout: Not Supported, TimeoutDis- ARIFwd- DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis- ARIFwd- LnkCtl2: Target Link Speed: 2.5GT/s, EnterCompliance- SpeedDis-, Selectable De-emphasis: -6dB Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -6dB Capabilities: [a0] Message Signalled Interrupts: Mask- 64bit- Count=1/1 Enable+ Address: fee0f00c Data: 4161 Capabilities: [b0] Subsystem: ASUSTeK Computer Inc. Device 83a2 Capabilities: [b8] HyperTransport: MSI Mapping Enable+ Fixed+ Capabilities: [100] Vendor Specific Information <?> Capabilities: [110] Virtual Channel <?> Kernel driver in use: pcieport-driver 00:11.0 SATA controller: ATI Technologies Inc SB700/SB800 SATA Controller [AHCI mode] (prog-if 01 [AHCI 1.0]) Subsystem: ASUSTeK Computer Inc. Device 8389 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: I/O ports at c000 [size=8] Region 1: I/O ports at b000 [size=4] Region 2: I/O ports at a000 [size=8] Region 3: I/O ports at 9000 [size=4] Region 4: I/O ports at 8000 [size=16] Region 5: Memory at fe9ffc00 (32-bit, non-prefetchable) [size=1K] Capabilities: [60] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [70] SATA HBA <?> Kernel driver in use: ahci 00:12.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller (prog-if 10 [OHCI]) Subsystem: ASUSTeK Computer Inc. Device 8389 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 16 Region 0: Memory at fe9fe000 (32-bit, non- prefetchable) [size=4K] Kernel driver in use: ohci_hcd Kernel modules: ohci-hcd 00:12.1 USB Controller: ATI Technologies Inc SB700 USB OHCI1 Controller (prog-if 10 [OHCI]) Subsystem: ASUSTeK Computer Inc. Device 8389 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 16 Region 0: Memory at fe9fd000 (32-bit, non- prefetchable) [size=4K] Kernel driver in use: ohci_hcd Kernel modules: ohci-hcd 00:12.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller (prog-if 20 [EHCI]) Subsystem: ASUSTeK Computer Inc. Device 8389 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 17 Region 0: Memory at fe9ff800 (32- bit, non-prefetchable) [size=256] Capabilities: [c0] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+, D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Bridge: PM- B3+ Capabilities: [e4] Debug port: BAR=1 offset=00e0 Kernel driver in use: ehci_hcd Kernel modules: ehci-hcd 00:13.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller (prog- if 10 [OHCI]) Subsystem: ASUSTeK Computer Inc. Device 8389 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 18 Region 0: Memory at fe9fc000 (32-bit, non-prefetchable) [size=4K] Kernel driver in use: ohci_hcd Kernel modules: ohci-hcd 00:13.1 USB Controller: ATI Technologies Inc SB700 USB OHCI1 Controller (prog-if 10 [OHCI]) Subsystem: ASUSTeK Computer Inc. Device 8389 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 18 Region 0: Memory at fe9fb000 (32-bit, non-prefetchable) [size=4K] Kernel driver in use: ohci_hcd Kernel modules: ohci-hcd 00:13.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller (prog- if 20 [EHCI]) Subsystem: ASUSTeK Computer Inc. Device 8389 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 19 Region 0: Memory at fe9ff400 (32-bit, non-prefetchable) [size=256] Capabilities: [c0] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME- Enable- DSel=0 DScale=0 PME- Bridge: PM- B3+ Capabilities: [e4] Debug port: BAR=1 offset=00e0 Kernel driver in use: ehci_hcd Kernel modules: ehci-hcd 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 3c) Subsystem: ASUSTeK Computer Inc. Device 8389 Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort+ <MAbort- >SERR- <PERR- INTx- Capabilities: [b0] HyperTransport: MSI Mapping Enable- Fixed+ 00:14.1 IDE interface: ATI Technologies Inc SB700/SB800 IDE Controller (prog-if 8a [Master SecP PriP]) Subsystem: ASUSTeK Computer Inc. Device 8389 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 Interrupt: pin A routed to IRQ 16 Region 0: I/O ports at 01f0 [size=8] Region 1: I/O ports at 03f4 [size=1] Region 2: I/O ports at 0170 [size=8] Region 3: I/O ports at 0374 [size=1] Region 4: I/O ports at ff00 [size=16] Capabilities: [70] Message Signalled Interrupts: Mask- 64bit- Count=1/1 Enable- Address: 00000000 Data: 0000 Kernel driver in use: pata_atiixp 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) Subsystem: ASUSTeK Computer Inc. Device 836c 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: 64, Cache Line Size: 64 bytes Interrupt: pin ? routed to IRQ 16 Region 0: Memory at fe9f4000 (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+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel 00:14.3 ISA bridge: ATI Technologies Inc SB700/SB800 LPC host controller Subsystem: ASUSTeK Computer Inc. Device 8389 Control: I/O+ Mem+ BusMaster+ SpecCycle+ MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge (prog-if 01 [Subtractive decode]) Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 Bus: primary=00, secondary=03, subordinate=03, sec- latency=64 I/O behind bridge: 0000f000-00000fff Memory behind bridge: fff00000-000fffff Prefetchable memory behind bridge: fff00000-000fffff Secondary status: 66MHz- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity+ SERR+ NoISA+ VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- 00:14.5 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI2 Controller (prog-if 10 [OHCI]) Subsystem: ASUSTeK Computer Inc. Device 8389 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin C routed to IRQ 18 Region 0: Memory at fe9fa000 (32-bit, non-prefetchable) [size=4K] Kernel driver in use: ohci_hcd Kernel modules: ohci-hcd 00:18.0 Host bridge: Advanced Micro Devices [AMD] Family 10h [Opteron, Athlon64, Sempron] HyperTransport Configuration 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- Capabilities: [80] HyperTransport: Host or Secondary Interface Command: WarmRst+ DblEnd- DevNum=0 ChainSide- HostHide+ Slave- <EOCErr- DUL- Link Control: CFlE- CST- CFE- <LkFail- Init+ EOC- TXO- <CRCErr=0 IsocEn+ LSEn+ ExtCTL- 64b- Link Config: MLWI=16bit DwFcIn- MLWO=16bit DwFcOut- LWI=16bit DwFcInEn- LWO=16bit DwFcOutEn- Revision ID: 3.00 Link Frequency: [b] Link Error: <Prot- <Ovfl- <EOC- CTLTm- Link Frequency Capability: 200MHz+ 300MHz- 400MHz+ 500MHz- 600MHz+ 800MHz+ 1.0GHz+ 1.2GHz+ 1.4GHz- 1.6GHz- Vend- Feature Capability: IsocFC+ LDTSTOP+ CRCTM- ECTLT- 64bA+ UIDRD- ExtRS- UCnfE- 00:18.1 Host bridge: Advanced Micro Devices [AMD] Family 10h [Opteron, Athlon64, Sempron] Address Map 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- 00:18.2 Host bridge: Advanced Micro Devices [AMD] Family 10h [Opteron, Athlon64, Sempron] DRAM Controller 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- Kernel modules: amd64_edac_mod 00:18.3 Host bridge: Advanced Micro Devices [AMD] Family 10h [Opteron, Athlon64, Sempron] Miscellaneous Control 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- Capabilities: [f0] Secure device <?> 00: 18.4 Host bridge: Advanced Micro Devices [AMD] Family 10h [Opteron, Athlon64, Sempron] Link Control 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- 01:00.0 VGA compatible controller: ATI Technologies Inc Device 954f (prog-if 00 [VGA controller]) Subsystem: ASUSTeK Computer Inc. Device 02a8 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 18 Region 0: Memory at d0000000 (64-bit, prefetchable) [size=256M] Region 2: Memory at feaf0000 (64-bit, non-prefetchable) [size=64K] Region 4: I/O ports at d000 [size=256] Expansion ROM at feac0000 [disabled] [size=128K] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME- Enable- DSel=0 DScale=0 PME- Capabilities: [58] Express (v2) Legacy Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <4us, L1 unlimited ExtTag+ AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x16, ASPM L0s L1, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x16, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- DevCap2: Completion Timeout: Not Supported, TimeoutDis- DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis- LnkCtl2: Target Link Speed: 2.5GT/s, EnterCompliance- SpeedDis-, Selectable De-emphasis: -6dB Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -6dB Capabilities: [a0] Message Signalled Interrupts: Mask- 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [100] Vendor Specific Information <?> 01:00.1 Audio device: ATI Technologies Inc Device aa38 Subsystem: ASUSTeK Computer Inc. Device aa38 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 B routed to IRQ 28 Region 0: Memory at feaec000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [58] Express (v2) Legacy Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <4us, L1 unlimited ExtTag+ AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x16, ASPM L0s L1, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x16, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- DevCap2: Completion Timeout: Not Supported, TimeoutDis- DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis- LnkCtl2: Target Link Speed: 2.5 GT/s, EnterCompliance- SpeedDis-, Selectable De-emphasis: -6dB Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De- emphasis Level: -6dB Capabilities: [a0] Message Signalled Interrupts: Mask- 64bit+ Count=1/1 Enable+ Address: 00000000fee0f00c Data: 4189 Capabilities: [100] Vendor Specific Information <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel 02:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 03) Subsystem: ASUSTeK Computer Inc. Device 83a3 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 27 Region 0: I/O ports at e800 [size=256] Region 2: Memory at fdfff000 (64-bit, prefetchable) [size=4K] Region 4: Memory at fdff8000 (64-bit, prefetchable) [size=16K] Expansion ROM at febf0000 [disabled] [size=64K] Capabilities: [40] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=375mA PME(D0+,D1+, D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Count=1/1 Enable+ Address: 00000000fee0f00c Data: 4179 Capabilities: [70] Express (v2) Endpoint, MSI 01 DevCap: MaxPayload 256 bytes, PhantFunc 0, Latency L0s <512ns, L1 <64us ExtTag- AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 4096 bytes DevSta: CorrErr+ UncorrErr- FatalErr- UnsuppReq+ AuxPwr+ TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <512ns, L1 <64us ClockPM+ Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- DevCap2: Completion Timeout: Not Supported, TimeoutDis+ DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis- LnkCtl2: Target Link Speed: 2.5GT/s, EnterCompliance- SpeedDis-, Selectable De-emphasis: -6dB Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -6dB Capabilities: [ac] MSI-X: Enable- Mask- TabSize=4 Vector table: BAR=4 offset=00000000 PBA: BAR=4 offset=00000800 Capabilities: [cc] Vital Product Data <?> Capabilities: [100] Advanced Error Reporting UESta: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSVoil- UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSVoil- UESvrt: DLP+ SDES+ TLP- FCP+ CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- ACSVoil- CESta: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ CESta: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ AERCap: First Error Pointer: 00, GenCap+ CGenEn- ChkCap+ ChkEn- Capabilities: [140] Virtual Channel <?> Capabilities: [160] Device Serial Number 00-00-00-00-00-00-00-00 Kernel driver in use: r8169 Risparmia con Tutto Incluso Light: telefono + adsl 8 mega a soli 14,95 € al mese. Gratis la Sim Tiscali Mobile con 25 euro di traffico! L'offerta è valida solo se attivi entro il 20/05/10 http://abbonati.tiscali.it/telefono-adsl/prodotti/tc/tuttoincluso_light/?WT…

2 1

Re: [alsa-devel] [PATCH] ALSA: hda: Use LPIB for Toshiba A100-259
by Daniel Chen 22 May '10

22 May '10

(Sorry for top posting due to MUA) Convention is to sort ascending; 0x1106 proceeds 0x1043. On May 22, 2010 3:27 PM, "Paul Menzel" <paulepanter(a)users.sourceforge.net> wrote: Dear Daniel, Am Samstag, den 22.05.2010, 13:25 -0400 schrieb Daniel Chen: […] > - SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), > SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), > + SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), > + SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB), Is not »m« before »p« in the alphabet? Thanks, Paul _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[PATCH] ASoC: Update Freescale i.MX SSI driver DMA parameter handling
by Stuart Longland 22 May '10

22 May '10

This updates the i.MX SSI driver to make it compatible with the ASoC tree following the move of DMA parameters from the DAI to the audio substream object. Signed-off-by: Stuart Longland <redhatter(a)gentoo.org> --- sound/soc/imx/imx-pcm-dma-mx2.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 2b31ac6..05f19c9 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -73,7 +73,8 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err) { struct snd_pcm_substream *substream = data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_dma_params *dma_params = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; int ret; @@ -102,7 +103,7 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) struct imx_pcm_runtime_data *iprtd = runtime->private_data; int ret; - dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream); + dma_params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH); if (iprtd->dma < 0) { @@ -212,7 +213,7 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) struct imx_pcm_runtime_data *iprtd = runtime->private_data; int err; - dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream); + dma_params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); iprtd->substream = substream; iprtd->buf = (unsigned int *)substream->dma_buffer.area; -- 1.6.4 --2fHTh5uZTiUOsy+g--

1 0

[alsa-devel] [PATCH] ad193x: add set_sysclk entry to support different clock input
by Barry Song 21 May '10

21 May '10

Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/ad193x.c | 41 ++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/ad193x.h | 5 +++++ 2 files changed, 45 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index c8ca114..1def75e 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -24,6 +24,7 @@ /* codec private data */ struct ad193x_priv { + unsigned int sysclk; struct snd_soc_codec codec; u8 reg_cache[AD193X_NUM_REGS]; }; @@ -251,15 +252,32 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int ad193x_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 ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + switch (freq) { + case 12288000: + case 18432000: + case 24576000: + case 36864000: + ad193x->sysclk = freq; + return 0; + } + return -EINVAL; +} + static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - int word_len = 0, reg = 0; + int word_len = 0, reg = 0, master_rate = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); /* bit size */ switch (params_format(params)) { @@ -275,6 +293,25 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, break; } + switch (ad193x->sysclk) { + case 12288000: + master_rate = AD193X_PLL_INPUT_256; + break; + case 18432000: + master_rate = AD193X_PLL_INPUT_384; + break; + case 24576000: + master_rate = AD193X_PLL_INPUT_512; + break; + case 36864000: + master_rate = AD193X_PLL_INPUT_768; + break; + } + + reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0); + reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate; + snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); + reg = snd_soc_read(codec, AD193X_DAC_CTRL2); reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len; snd_soc_write(codec, AD193X_DAC_CTRL2, reg); @@ -348,6 +385,7 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type) /* pll input: mclki/xi */ snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04); + ad193x->sysclk = 12288000; ret = snd_soc_register_codec(codec); if (ret != 0) { @@ -383,6 +421,7 @@ static struct snd_soc_dai_ops ad193x_dai_ops = { .hw_params = ad193x_hw_params, .digital_mute = ad193x_mute, .set_tdm_slot = ad193x_set_tdm_slot, + .set_sysclk = ad193x_set_dai_sysclk, .set_fmt = ad193x_set_dai_fmt, }; diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index a03c880..654ba64 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -11,6 +11,11 @@ #define AD193X_PLL_CLK_CTRL0 0x800 #define AD193X_PLL_POWERDOWN 0x01 +#define AD193X_PLL_INPUT_MASK (~0x6) +#define AD193X_PLL_INPUT_256 (0 << 1) +#define AD193X_PLL_INPUT_384 (1 << 1) +#define AD193X_PLL_INPUT_512 (2 << 1) +#define AD193X_PLL_INPUT_768 (3 << 1) #define AD193X_PLL_CLK_CTRL1 0x801 #define AD193X_DAC_CTRL0 0x802 #define AD193X_DAC_POWERDOWN 0x01 -- 1.5.6.3

3 2

[alsa-devel] [PATCH 0/2] Correct widget and PLL handling
by Jorge Eduardo Candelaria 21 May '10

21 May '10

Fix an issue when using 19.2 Mhz as clock input for TWL6040. Also, Drivers in TWL6040 can be modeled after PGA components of DAPM. This allows for better power management and also ensures that the DACs are enabled before the Drivers, reducing this way a pop-noise heard when powering up components.

3 5

[alsa-devel] [ice1712] Word Clock with Delta1010 E
by Bernhard Walle 21 May '10

21 May '10

Hi, we have a problem with the word clock input of the Delta 1010 E. At first: I modified the source code of envy24control to accept the Delta 1010 E as subdevice so that it allows to set the word clock. I'll post the patch after it works with the driver. However, the problem also exists when I use plain alsamixer or amixer. With the unmodified driver, I'm able to turn word clock on but whenever I record something, the arecord program (or Audacity) doesn't get any samples. With OSS (which we're using now but we're trying to switch to ALSA) my colleague Markus Prosch made a patch with lots of try & error that made the word clock working: http://opensound.hg.sourceforge.net/hgweb/opensound/opensound/rev/6a4bad6e7… When I try to "port" that patch to ALSA, the behaviour changes: The recording works, but the sample rate is too low so that it sounds quite strange. In any case: In the same hardware setup, it works flawlessly with OSS. Also, we had no problem with the older Delta 1010. It seems to be an incompatible change (or bug ;)) in revision E. Are there any devs here that know the hardware better and could help me in debugging that issue? Without any deeper sound driver knowledge and without any data sheet, that's quite hard for me. Thanks for your help! Regards, Bernhard PS: Please keep me in Cc since I'm not on the list.

2 2

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: add support for Akai MPD16
by Krzysztof Foltman 20 May '10

20 May '10

The decoding/encoding is based on own reverse-engineering. Both control and data ports are handled. Writing to control port supports SysEx events only, as this is the only type of messages that MPD16 recognizes. Signed-off-by: Krzysztof Foltman <wdev(a)foltman.com> diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 2c1558c..9c23d89 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -645,6 +645,105 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = { }; /* + * AKAI MPD16 protocol: + * + * For control port (endpoint 1): + * ============================== + * One or more chunks consisting of first byte of (0x10 | msg_len) and then a + * SysEx message (msg_len=9 bytes long). + * + * For data port (endpoint 2): + * =========================== + * One or more chunks consisting of first byte of (0x20 | msg_len) and then a + * MIDI message (msg_len bytes long) + * + * Messages sent: Active Sense, Note On, Poly Pressure, Control Change. + */ +static void snd_usbmidi_akai_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) +{ + unsigned int pos = 0; + unsigned int len = (unsigned int)buffer_length; + while (pos < len) { + unsigned int port = (buffer[pos] >> 4) - 1; + unsigned int msg_len = buffer[pos] & 0x0f; + pos++; + if (pos + msg_len <= len && port < 2) + snd_usbmidi_input_data(ep, 0, &buffer[pos], msg_len); + pos += msg_len; + } +} + +#define MAX_AKAI_SYSEX_LEN 9 + +static void snd_usbmidi_akai_output(struct snd_usb_midi_out_endpoint *ep, + struct urb *urb) +{ + uint8_t *msg; + int pos, end, count, buf_end; + uint8_t tmp[MAX_AKAI_SYSEX_LEN]; + struct snd_rawmidi_substream *substream = ep->ports[0].substream; + + if (!ep->ports[0].active) + return; + + msg = urb->transfer_buffer + urb->transfer_buffer_length; + buf_end = ep->max_transfer - MAX_AKAI_SYSEX_LEN - 1; + + /* only try adding more data when there's space for at least 1 SysEx */ + while (urb->transfer_buffer_length < buf_end) { + count = snd_rawmidi_transmit_peek(substream, + tmp, MAX_AKAI_SYSEX_LEN); + if (!count) { + ep->ports[0].active = 0; + return; + } + /* try to skip non-SysEx data */ + for (pos = 0; pos < count && tmp[pos] != 0xF0; pos++) + ; + + if (pos > 0) { + snd_rawmidi_transmit_ack(substream, pos); + continue; + } + + /* look for the start or end marker */ + for (end = 1; end < count && tmp[end] < 0xF0; end++) + ; + + /* next SysEx started before the end of current one */ + if (end < count && tmp[end] == 0xF0) { + /* it's incomplete - drop it */ + snd_rawmidi_transmit_ack(substream, end); + continue; + } + /* SysEx complete */ + if (end < count && tmp[end] == 0xF7) { + /* queue it, ack it, and get the next one */ + count = end + 1; + msg[0] = 0x10 | count; + memcpy(&msg[1], tmp, count); + snd_rawmidi_transmit_ack(substream, count); + urb->transfer_buffer_length += count + 1; + msg += count + 1; + continue; + } + /* less than 9 bytes and no end byte - wait for more */ + if (count < MAX_AKAI_SYSEX_LEN) { + ep->ports[0].active = 0; + return; + } + /* 9 bytes and no end marker in sight - malformed, skip it */ + snd_rawmidi_transmit_ack(substream, count); + } +} + +static struct usb_protocol_ops snd_usbmidi_akai_ops = { + .input = snd_usbmidi_akai_input, + .output = snd_usbmidi_akai_output, +}; + +/* * Novation USB MIDI protocol: number of data bytes is in the first byte * (when receiving) (+1!) or in the second byte (when sending); data begins * at the third byte. @@ -1434,6 +1533,11 @@ static struct port_info { EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), + /* Akai MPD16 */ + CONTROL_PORT(0x09e8, 0x0062, 0, "%s Control"), + PORT_INFO(0x09e8, 0x0062, 1, "%s MIDI", 0, + SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | + SNDRV_SEQ_PORT_TYPE_HARDWARE), /* Access Music Virus TI */ EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"), PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0, @@ -2035,6 +2139,12 @@ int snd_usbmidi_create(struct snd_card *card, umidi->usb_protocol_ops = &snd_usbmidi_cme_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_AKAI: + umidi->usb_protocol_ops = &snd_usbmidi_akai_ops; + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + /* endpoint 1 is input-only */ + endpoints[1].out_cables = 0; + break; default: snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); err = -ENXIO; diff --git a/sound/usb/midi.h b/sound/usb/midi.h index 2089ec9..2fca80b 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -37,6 +37,8 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_MIDI_CME, data is NULL */ +/* for QUIRK_MIDI_AKAI, data is NULL */ + int snd_usbmidi_create(struct snd_card *card, struct usb_interface *iface, struct list_head *midi_list, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 91ddef3..f8797f6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1973,6 +1973,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* AKAI devices */ +{ + USB_DEVICE(0x09e8, 0x0062), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "AKAI", + .product_name = "MPD16", + .ifnum = 0, + .type = QUIRK_MIDI_AKAI, + } +}, + /* TerraTec devices */ { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 136e5b4..b45e54c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -289,6 +289,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, + [QUIRK_MIDI_AKAI] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index d679e72..06ebf24 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -74,6 +74,7 @@ enum quirk_type { QUIRK_MIDI_FASTLANE, QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, + QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, -- 1.7.0.4

1 0

[alsa-devel] [PATCH] ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
by Jarkko Nikula 20 May '10

20 May '10

Codec output pin should be defined with SND_SOC_DAPM_OUTPUT as otherwise external widgets doesn't alter the output state. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- I noticed this with a widget and audio map below where state of "Headphone Jack" wasn't changing the codec output. SND_SOC_DAPM_HP("Headphone Jack", NULL), {"Headphone Jack", NULL, "TPA6130A2 Headphone Left"}, {"Headphone Jack", NULL, "TPA6130A2 Headphone Right"}, {"TPA6130A2 Left", NULL, "LLOUT"}, {"TPA6130A2 Right", NULL, "RLOUT"} --- sound/soc/codecs/tpa6130a2.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 100a747..99b70e5 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -359,8 +359,8 @@ static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { 0, 0, tpa6130a2_supply_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* Outputs */ - SND_SOC_DAPM_HP("TPA6130A2 Headphone Left", NULL), - SND_SOC_DAPM_HP("TPA6130A2 Headphone Right", NULL), + SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Left"), + SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Right"), }; static const struct snd_soc_dapm_route audio_map[] = { -- 1.7.1

4 7

[alsa-devel] [RFC] disabling ALSA period interrupts
by pl bossart 20 May '10

20 May '10

Howdy, When PulseAudio is used and all PCM is routed through PulseAudio (Fedora, Meego, etc), the notion of ALSA periods isn't very useful. PulseAudio uses a timer to refill buffers and the period interrupts are not used at all. So why not disable them entirely to reduce the number of wakeups? This is possible with a number of existing DMA controllers. The simple patch attached shows a proof-of-concept on HDAudio, it's been working for 5 hours on my Fedora12 laptop without any glitches and powertop does show _zero_ wakeups from the HDAudio controller (except on startup). I am told by my colleagues working in Windows environments that this is what's done on Vista/Windows7 btw. This isn't to say Windows is great but that artificial generation of not-needed interrupts is avoidable. There are probably some cases where you don't want this type of behavior (broken hardware, legacy code with multiple-buffering, disabled timer in PulseAudio), so I think it would make sense to request the disabling of interrupts when hw_params are set, since this is also the time when period sizes are set. I am aware that some changes would be needed in pcm_lib.c, where all the error checks are done. Takashi, Jaroslav, Lennart, what do you think? Feedback and suggestions welcome. Cheers - Pierre

10 56

[alsa-devel] [GIT PULL] ASoC update for 2.6.35
by Liam Girdwood 20 May '10

20 May '10

Hi Takashi, Hope this is not too late. Thanks Liam --- The following changes since commit 266d38c8e3d7f62152b1448fd9a7265f32f32d87: Jarkko Nikula (1): ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.35 Jarkko Nikula (1): ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies sound/soc/codecs/tpa6130a2.c | 52 ++++++++++++++--------------------------- 1 files changed, 18 insertions(+), 34 deletions(-)

2 1

[alsa-devel] [PATCH 11/11] ALSA: hda: Storage class should be before const qualifier
by Tobias Klauser 20 May '10

20 May '10

The C99 specification states in section 6.11.5: The placement of a storage-class specifier other than at the beginning of the declaration specifiers in a declaration is an obsolescent feature. Signed-off-by: Tobias Klauser <tklauser(a)distanz.ch> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 80a183f..53538b0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14207,7 +14207,7 @@ enum { ALC269_FIXUP_SONY_VAIO, }; -const static struct hda_verb alc269_sony_vaio_fixup_verbs[] = { +static const struct hda_verb alc269_sony_vaio_fixup_verbs[] = { {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, {} }; -- 1.6.3.3

2 1

[alsa-devel] [PATCHv2] ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
by Jarkko Nikula 20 May '10

20 May '10

These pins are for decoupling capacitors for the internal charge pumps in TPA6130A2 and TPA6140A2 and not for connecting external supply. Thanks to Eduardo Valentin <eduardo.valentin(a)nokia.com> for pointing out the issue with TPA6130A2 and Ilkka Koskinen <ilkka.koskinen(a)nokia.com> with TPA6140A2. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Reviewed-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- v2: - I took freedom to add ack and reviewed by lines as Peter and Ilkka gave those to v1 - HPVdd for the TPA6140A2 is also removed as it's for decoupling capacitor as well - No need to use regulator_bulk API as there s only single supply for both supported chips now --- sound/soc/codecs/tpa6130a2.c | 50 ++++++++++++++--------------------------- 1 files changed, 17 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 221bb90..f9b3d6d 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -36,22 +36,11 @@ static struct i2c_client *tpa6130a2_client; -#define TPA6130A2_NUM_SUPPLIES 2 -static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { - "CPVSS", - "Vdd", -}; - -static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { - "HPVdd", - "AVdd", -}; - /* This struct is used to save the context */ struct tpa6130a2_data { struct mutex mutex; unsigned char regs[TPA6130A2_CACHEREGNUM]; - struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES]; + struct regulator *supply; int power_gpio; unsigned char power_state; enum tpa_model id; @@ -136,11 +125,10 @@ static int tpa6130a2_power(int power) if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); - ret = regulator_bulk_enable(ARRAY_SIZE(data->supplies), - data->supplies); + ret = regulator_enable(data->supply); if (ret != 0) { dev_err(&tpa6130a2_client->dev, - "Failed to enable supplies: %d\n", ret); + "Failed to enable supply: %d\n", ret); goto exit; } @@ -161,11 +149,10 @@ static int tpa6130a2_power(int power) if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 0); - ret = regulator_bulk_disable(ARRAY_SIZE(data->supplies), - data->supplies); + ret = regulator_disable(data->supply); if (ret != 0) { dev_err(&tpa6130a2_client->dev, - "Failed to disable supplies: %d\n", ret); + "Failed to disable supply: %d\n", ret); goto exit; } @@ -412,6 +399,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, struct device *dev; struct tpa6130a2_data *data; struct tpa6130a2_platform_data *pdata; + const char *regulator; int i, ret; dev = &client->dev; @@ -454,25 +442,21 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, } switch (data->id) { + default: + dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", + pdata->id); case TPA6130A2: - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; + regulator = "Vdd"; break; case TPA6140A2: - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6140a2_supply_names[i];; + regulator = "AVdd"; break; - default: - dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", - pdata->id); - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; } - ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), - data->supplies); - if (ret != 0) { - dev_err(dev, "Failed to request supplies: %d\n", ret); + data->supply = regulator_get(dev, regulator); + if (IS_ERR(data->supply)) { + ret = PTR_ERR(data->supply); + dev_err(dev, "Failed to request supply: %d\n", ret); goto err_regulator; } @@ -495,7 +479,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, return 0; err_power: - regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + regulator_put(data->supply); err_regulator: if (data->power_gpio >= 0) gpio_free(data->power_gpio); @@ -516,7 +500,7 @@ static int __devexit tpa6130a2_remove(struct i2c_client *client) if (data->power_gpio >= 0) gpio_free(data->power_gpio); - regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + regulator_put(data->supply); kfree(data); tpa6130a2_client = NULL; -- 1.7.1

2 2

[alsa-devel] First 50 milliseconds of audio clipping with dmix
by Tyler Conant 20 May '10

20 May '10

I'm trying to use dmix on an OMAP processor. I'm experiencing the first 50 ms of the sample getting clipped. When I aplay directly to the hw:0 I don't get the clipping. The difference I see between the alsa configuration between dmix and hw:0 is the start_threshold, stop_threshold and silence_size. When it's direct to hw:0 the start and stop threshold are equal to the buffer size. For dmix the start threshold is 1 and the stop threshold is 1445199872. Is this dmix which is setting the start and stop or is this aplay. I'm only guessing this is the reason why the first part is clipping. The only bug which seem similar is http://bugs.gentoo.org/show_bug.cgi?id=218641 and it's already been closed out. I'm using alsa lib 1.0.23 with hw:0 Audio_Prompts # aplay -v -D hw:0 Click03.wav Playing WAVE 'Click03.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22052 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 22052 stop_threshold : 22052 silence_threshold: 0 silence_size : 0 boundary : 1445199872 appl_ptr : 0 hw_ptr : 0 /opt/axonapp/Audio_Prompts # With dmix: /Audio_Prompts # aplay -v Click03.wav Playing WAVE 'Click03.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Plug PCM: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22052 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 22052 stop_threshold : 22052 silence_threshold: 0 silence_size : 0 boundary : 1445199872 Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22052 period_size : 5513 period_time : 125011 tstamp_mode : ENABLE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 1 stop_threshold : 1445199872 silence_threshold: 0 silence_size : 1445199872 boundary : 1445199872 appl_ptr : 0 hw_ptr : 0 /opt/axonapp/Audio_Prompts # asound.conf: pcm.dmixer { type dmix ipc_key 2048 ipc_key_add_uid true slave { channels 1 pcm hw:0 rate 44100 period_time 125011 period_size 5513 buffer_size 22052 } } pcm.asymed { type asym playback.pcm "dmixer" capture.pcm "hw:0,0" } pcm.!default { type plug slave.pcm "asymed" }

3 2

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: add support for Akai MPD16
by Krzysztof Foltman 19 May '10

19 May '10

The decoding routine is based on own reverse-engineering. It seems to recognize all the messages that MPD16 sends during normal operation (that is, via standard MIDI messages). Configuration (changing pad sensitivity, slider controller and MIDI notes) is not supported in this version. Signed-off-by: Krzysztof Foltman <wdev(a)foltman.com> diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 2c1558c..cce5dbe 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -645,6 +645,33 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = { }; /* + * AKAI MPD16 protocol: one or more chunks consisting of first byte of + * (0x20 | msg_len) and then a MIDI message (msg_len bytes long) + * + * Messages sent: + * 21 FE (active sense) + * 23 90 xx xx (note on) + * 23 Ax xx xx (polyphonic pressure) + * 23 Bx xx xx (control change) + */ +static void snd_usbmidi_akai_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) +{ + unsigned int pos = 0; + while (pos < (unsigned)buffer_length && (buffer[pos] & 0xF8) == 0x20) { + int msg_len = buffer[pos] & 0x0f; + snd_usbmidi_input_data(ep, 0, &buffer[pos + 1], msg_len); + pos += 1 + msg_len; + } +} + +static struct usb_protocol_ops snd_usbmidi_akai_ops = { + .input = snd_usbmidi_akai_input, + .output = snd_usbmidi_standard_output, + .output_packet = snd_usbmidi_output_standard_packet, +}; + +/* * Novation USB MIDI protocol: number of data bytes is in the first byte * (when receiving) (+1!) or in the second byte (when sending); data begins * at the third byte. @@ -1434,6 +1461,8 @@ static struct port_info { EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), + /* Akai MPD16 */ + EXTERNAL_PORT(0x09e8, 0x0062, 0, "%s MIDI"), /* Access Music Virus TI */ EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"), PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0, @@ -2035,6 +2064,13 @@ int snd_usbmidi_create(struct snd_card *card, umidi->usb_protocol_ops = &snd_usbmidi_cme_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_AKAI: + umidi->usb_protocol_ops = &snd_usbmidi_akai_ops; + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + /* endpoint 0 does not carry MIDI data */ + endpoints[0].out_cables = 0; + endpoints[0].in_cables = 0; + break; default: snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); err = -ENXIO; diff --git a/sound/usb/midi.h b/sound/usb/midi.h index 2089ec9..2fca80b 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -37,6 +37,8 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_MIDI_CME, data is NULL */ +/* for QUIRK_MIDI_AKAI, data is NULL */ + int snd_usbmidi_create(struct snd_card *card, struct usb_interface *iface, struct list_head *midi_list, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 91ddef3..7ecad05 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1973,6 +1973,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* AKAI devices */ +{ + USB_DEVICE(0x09e8, 0x0062), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "AKAI", + .product_name = "MPD16", + .ifnum = 0, + .type = QUIRK_MIDI_AKAI, + } +}, + /* TerraTec devices */ { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 136e5b4..b45e54c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -289,6 +289,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, + [QUIRK_MIDI_AKAI] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index d679e72..06ebf24 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -74,6 +74,7 @@ enum quirk_type { QUIRK_MIDI_FASTLANE, QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, + QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, -- 1.7.0.4

2 2

[alsa-devel] [GIT PULL] ASoC updates for 2.6.35
by Liam Girdwood 19 May '10

19 May '10

Hi Takashi, The last one for 2.6.35 :) Thanks Liam ---- The following changes since commit 2d4cdd6fc974716555fcbaf7ec1d4dda22784c1b: Peter Ujfalusi (1): ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.35 Jarkko Nikula (1): ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT Jorge Eduardo Candelaria (2): ASoC: SDP4430: Add support for Earphone speaker ASoC: TWL6040: Enable earphone path in codec Liam Girdwood (1): ASoC: sdp4430 - add sdp4430 pcm ops to DAI. Misael Lopez Cruz (1): ASoC: SDP4430: Add sdp4430 machine driver sound/soc/codecs/tpa6130a2.c | 4 +- sound/soc/codecs/twl6040.c | 21 ++++- sound/soc/omap/Kconfig | 9 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp4430.c | 233 ++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 266 insertions(+), 3 deletions(-) create mode 100644 sound/soc/omap/sdp4430.c

2 1

[alsa-devel] [PATCH 0/4] McBSP smart idle and DMA op mode updates for ASoC
by Liam Girdwood 19 May '10

19 May '10

This series expands the OMAP mcbsp driver to support changing it's DMA operating mode and smart idle mode from client drivers. It's primarily aimed at lowering the power consumption for OMAP ASoC drivers by providing methods to gate clocks on the mcbsp interface at runtime. I've also added a patch to remove the mcbsp DMA op mode sysfs set functionality. I think DMA op mode is very specific to the mcbsp client driver _only_ and shouldn't really be changed by userspace. Please let me know if you use this feature and I'll drop this patch. Tony, can I get your ack to upstream via ALSA if it's ok Thanks Liam arch/arm/plat-omap/include/plat/mcbsp.h | 17 +++ arch/arm/plat-omap/mcbsp.c | 183 ++++++++++++++++++++++++++----- sound/soc/omap/omap-mcbsp.c | 22 ++++- sound/soc/omap/omap-mcbsp.h | 2 + 4 files changed, 194 insertions(+), 30 deletions(-)

7 31

Re: [alsa-devel] [PATCH v2] NUC900/audio: add nuc900 audio driver support
by Wan ZongShun 19 May '10

19 May '10

Hi Mark, How about this patch? Could it be merged into mainline? Thanks! 在 2010年5月18日下午1:41，Wan ZongShun <mcuos.com(a)gmail.com> 写道： > Dear Mark, > > I have modified some codes according to your comments, > now I submit the patch as v2 again. > > Thanks a lot for your help. > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> > --- > sound/soc/Kconfig | 1 + > sound/soc/Makefile | 1 + > sound/soc/nuc900/Kconfig | 27 +++ > sound/soc/nuc900/Makefile | 11 + > sound/soc/nuc900/nuc900-ac97.c | 443 +++++++++++++++++++++++++++++++++++++++ > sound/soc/nuc900/nuc900-audio.c | 81 +++++++ > sound/soc/nuc900/nuc900-auido.h | 121 +++++++++++ > sound/soc/nuc900/nuc900-pcm.c | 352 +++++++++++++++++++++++++++++++ > 8 files changed, 1037 insertions(+), 0 deletions(-) > create mode 100644 sound/soc/nuc900/Kconfig > create mode 100644 sound/soc/nuc900/Makefile > create mode 100644 sound/soc/nuc900/nuc900-ac97.c > create mode 100644 sound/soc/nuc900/nuc900-audio.c > create mode 100644 sound/soc/nuc900/nuc900-auido.h > create mode 100644 sound/soc/nuc900/nuc900-pcm.c > > diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig > index b1749bc..6e04fc2 100644 > --- a/sound/soc/Kconfig > +++ b/sound/soc/Kconfig > @@ -30,6 +30,7 @@ source "sound/soc/blackfin/Kconfig" > source "sound/soc/davinci/Kconfig" > source "sound/soc/fsl/Kconfig" > source "sound/soc/imx/Kconfig" > +source "sound/soc/nuc900/Kconfig" > source "sound/soc/omap/Kconfig" > source "sound/soc/pxa/Kconfig" > source "sound/soc/s3c24xx/Kconfig" > diff --git a/sound/soc/Makefile b/sound/soc/Makefile > index 1470141..ccec241 100644 > --- a/sound/soc/Makefile > +++ b/sound/soc/Makefile > @@ -8,6 +8,7 @@ obj-$(CONFIG_SND_SOC) += blackfin/ > obj-$(CONFIG_SND_SOC) += davinci/ > obj-$(CONFIG_SND_SOC) += fsl/ > obj-$(CONFIG_SND_SOC) += imx/ > +obj-$(CONFIG_SND_SOC) += nuc900/ > obj-$(CONFIG_SND_SOC) += omap/ > obj-$(CONFIG_SND_SOC) += pxa/ > obj-$(CONFIG_SND_SOC) += s3c24xx/ > diff --git a/sound/soc/nuc900/Kconfig b/sound/soc/nuc900/Kconfig > new file mode 100644 > index 0000000..a0ed1c6 > --- /dev/null > +++ b/sound/soc/nuc900/Kconfig > @@ -0,0 +1,27 @@ > +## > +## NUC900 series AC97 API > +## > +config SND_SOC_NUC900 > + tristate "SoC Audio for NUC900 series" > + depends on ARCH_W90X900 > + help > + This option enables support for AC97 mode on the NUC900 SoC. > + > +config SND_SOC_NUC900_AC97 > + tristate > + select AC97_BUS > + select SND_AC97_CODEC > + select SND_SOC_AC97_BUS > + > + > +## > +## Boards > +## > +config SND_SOC_NUC900EVB > + tristate "NUC900 AC97 support for demo board" > + depends on SND_SOC_NUC900 > + select SND_SOC_NUC900_AC97 > + select SND_SOC_AC97_CODEC > + help > + Select this option to enable audio (AC97) on the > + NUC900 demoboard. > diff --git a/sound/soc/nuc900/Makefile b/sound/soc/nuc900/Makefile > new file mode 100644 > index 0000000..7e46c71 > --- /dev/null > +++ b/sound/soc/nuc900/Makefile > @@ -0,0 +1,11 @@ > +# NUC900 series audio > +snd-soc-nuc900-pcm-objs := nuc900-pcm.o > +snd-soc-nuc900-ac97-objs := nuc900-ac97.o > + > +obj-$(CONFIG_SND_SOC_NUC900) += snd-soc-nuc900-pcm.o > +obj-$(CONFIG_SND_SOC_NUC900_AC97) += snd-soc-nuc900-ac97.o > + > +# Boards > +snd-soc-nuc900-audio-objs := nuc900-audio.o > + > +obj-$(CONFIG_SND_SOC_NUC900EVB) += snd-soc-nuc900-audio.o > diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c > new file mode 100644 > index 0000000..2c345e2 > --- /dev/null > +++ b/sound/soc/nuc900/nuc900-ac97.c > @@ -0,0 +1,443 @@ > +/* > + * Copyright (c) 2009-2010 Nuvoton technology corporation. > + * > + * Wan ZongShun <mcuos.com(a)gmail.com> > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation;version 2 of the License. > + * > + */ > + > +#include <linux/init.h> > +#include <linux/module.h> > +#include <linux/slab.h> > +#include <linux/device.h> > +#include <linux/delay.h> > +#include <linux/mutex.h> > +#include <linux/suspend.h> > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/initval.h> > +#include <sound/soc.h> > +#include <linux/device.h> > +#include <linux/clk.h> > + > +#include <mach/mfp.h> > + > +#include "nuc900-auido.h" > + > +static DEFINE_MUTEX(ac97_mutex); > +struct nuc900_audio *nuc900_ac97_data; > + > +static int nuc900_checkready(void) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + > + if (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS0) & CODEC_READY)) > + return -EPERM; > + > + return 0; > +} > + > +/* AC97 controller reads codec register */ > +static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97, > + unsigned short reg) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + unsigned long timeout = 0x10000, val; > + > + mutex_lock(&ac97_mutex); > + > + val = nuc900_checkready(); > + if (!!val) { > + dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); > + goto out; > + } > + > + /* set the R_WB bit and write register index */ > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, R_WB | reg); > + > + /* set the valid frame bit and valid slots */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > + val |= (VALID_FRAME | SLOT1_VALID); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val); > + > + udelay(100); > + > + /* polling the AC_R_FINISH */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + val &= AC_R_FINISH; > + while (!val && timeout--) > + mdelay(1); > + > + if (!timeout) { > + dev_err(nuc900_audio->dev, "AC97 read register time out !\n"); > + val = -EPERM; > + goto out; > + } > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0) ; > + val &= ~SLOT1_VALID; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val); > + > + if (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS1) >> 2 != reg) { > + dev_err(nuc900_audio->dev, > + "R_INDEX of REG_ACTL_ACIS1 not match!\n"); > + } > + > + udelay(100); > + val = (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS2) & 0xFFFF); > + > +out: > + mutex_unlock(&ac97_mutex); > + return val; > +} > + > +/* AC97 controller writes to codec register */ > +static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg, > + unsigned short val) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + unsigned long tmp, timeout = 0x10000; > + > + mutex_lock(&ac97_mutex); > + > + tmp = nuc900_checkready(); > + if (!!tmp) > + dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); > + > + /* clear the R_WB bit and write register index */ > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, reg); > + > + /* write register value */ > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS2, val); > + > + /* set the valid frame bit and valid slots */ > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > + tmp |= SLOT1_VALID | SLOT2_VALID | VALID_FRAME; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); > + > + udelay(100); > + > + /* polling the AC_W_FINISH */ > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + tmp &= AC_W_FINISH; > + while (tmp && timeout--) > + mdelay(1); > + > + if (!timeout) > + dev_err(nuc900_audio->dev, "AC97 write register time out !\n"); > + > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > + tmp &= ~(SLOT1_VALID | SLOT2_VALID); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); > + > + mutex_unlock(&ac97_mutex); > + > +} > + > +static void nuc900_ac97_warm_reset(struct snd_ac97 *ac97) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + unsigned long val; > + > + mutex_lock(&ac97_mutex); > + > + /* warm reset AC 97 */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + val |= AC_W_RES; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); > + > + udelay(1000); > + > + val = nuc900_checkready(); > + if (!!val) > + dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); > + > + mutex_unlock(&ac97_mutex); > +} > + > +static void nuc900_ac97_cold_reset(struct snd_ac97 *ac97) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + unsigned long val; > + > + mutex_lock(&ac97_mutex); > + > + /* reset Audio Controller */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + val |= ACTL_RESET_BIT; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + udelay(1000); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + val &= (~ACTL_RESET_BIT); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + udelay(1000); > + > + /* reset AC-link interface */ > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + val |= AC_RESET; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + udelay(1000); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + val &= ~AC_RESET; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + udelay(1000); > + > + /* cold reset AC 97 */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + val |= AC_C_RES; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); > + > + udelay(1000); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + val &= (~AC_C_RES); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); > + > + udelay(1000); > + > + mutex_unlock(&ac97_mutex); > + > +} > + > +/* AC97 controller operations */ > +struct snd_ac97_bus_ops soc_ac97_ops = { > + .read = nuc900_ac97_read, > + .write = nuc900_ac97_write, > + .reset = nuc900_ac97_cold_reset, > + .warm_reset = nuc900_ac97_warm_reset, > +} > +EXPORT_SYMBOL_GPL(soc_ac97_ops); > + > +static int nuc900_ac97_trigger(struct snd_pcm_substream *substream, > + int cmd, struct snd_soc_dai *dai) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + int ret, stype = SUBSTREAM_TYPE(substream); > + unsigned long val, tmp; > + > + ret = 0; > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + case SNDRV_PCM_TRIGGER_RESUME: > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + if (PCM_TX == stype) { > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > + tmp |= (SLOT3_VALID | SLOT4_VALID | VALID_FRAME); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); > + > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR); > + tmp |= (P_DMA_END_IRQ | P_DMA_MIDDLE_IRQ); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, tmp); > + val |= AC_PLAY; > + } else { > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR); > + tmp |= (R_DMA_END_IRQ | R_DMA_MIDDLE_IRQ); > + > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, tmp); > + val |= AC_RECORD; > + } > + > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + break; > + case SNDRV_PCM_TRIGGER_STOP: > + case SNDRV_PCM_TRIGGER_SUSPEND: > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + if (PCM_TX == stype) { > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > + tmp &= ~(SLOT3_VALID | SLOT4_VALID); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); > + > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, RESET_PRSR); > + val &= ~AC_PLAY; > + } else { > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, RESET_PRSR); > + val &= ~AC_RECORD; > + } > + > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + break; > + default: > + ret = -EINVAL; > + } > + > + return ret; > +} > + > +static int nuc900_ac97_probe(struct platform_device *pdev, > + struct snd_soc_dai *dai) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + unsigned long val; > + > + mutex_lock(&ac97_mutex); > + > + /* enable unit clock */ > + clk_enable(nuc900_audio->clk); > + > + /* enable audio controller and AC-link interface */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); > + val |= (IIS_AC_PIN_SEL | ACLINK_EN); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val); > + > + mutex_unlock(&ac97_mutex); > + > + return 0; > +} > + > +static void nuc900_ac97_remove(struct platform_device *pdev, > + struct snd_soc_dai *dai) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + > + clk_disable(nuc900_audio->clk); > +} > + > +static struct snd_soc_dai_ops nuc900_ac97_dai_ops = { > + .trigger = nuc900_ac97_trigger, > +}; > + > +struct snd_soc_dai nuc900_ac97_dai = { > + .name = "nuc900-ac97", > + .probe = nuc900_ac97_probe, > + .remove = nuc900_ac97_remove, > + .ac97_control = 1, > + .playback = { > + .rates = SNDRV_PCM_RATE_8000_48000, > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels_min = 1, > + .channels_max = 2, > + }, > + .capture = { > + .rates = SNDRV_PCM_RATE_8000_48000, > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels_min = 1, > + .channels_max = 2, > + }, > + .ops = &nuc900_ac97_dai_ops, > +} > +EXPORT_SYMBOL_GPL(nuc900_ac97_dai); > + > +static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev) > +{ > + struct nuc900_audio *nuc900_audio; > + int ret; > + > + if (nuc900_ac97_data) > + return -EBUSY; > + > + nuc900_audio = kzalloc(sizeof(struct nuc900_audio), GFP_KERNEL); > + if (!nuc900_audio) > + return -ENOMEM; > + > + spin_lock_init(&nuc900_audio->lock); > + > + nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > + if (!nuc900_audio->res) { > + ret = -ENODEV; > + goto out0; > + } > + > + if (!request_mem_region(nuc900_audio->res->start, > + resource_size(nuc900_audio->res), pdev->name)) { > + ret = -EBUSY; > + goto out0; > + } > + > + nuc900_audio->mmio = ioremap(nuc900_audio->res->start, > + resource_size(nuc900_audio->res)); > + if (!nuc900_audio->mmio) { > + ret = -ENOMEM; > + goto out1; > + } > + > + nuc900_audio->clk = clk_get(&pdev->dev, NULL); > + if (IS_ERR(nuc900_audio->clk)) { > + ret = PTR_ERR(nuc900_audio->clk); > + goto out2; > + } > + > + nuc900_audio->irq_num = platform_get_irq(pdev, 0); > + if (!nuc900_audio->irq_num) { > + ret = -EBUSY; > + goto out2; > + } > + > + nuc900_ac97_data = nuc900_audio; > + > + nuc900_audio->dev = nuc900_ac97_dai.dev = &pdev->dev; > + > + ret = snd_soc_register_dai(&nuc900_ac97_dai); > + if (ret) > + goto out3; > + > + mfp_set_groupg(nuc900_audio->dev); /* enbale ac97 multifunction pin*/ > + > + return 0; > + > +out3: > + clk_put(nuc900_audio->clk); > +out2: > + iounmap(nuc900_audio->mmio); > +out1: > + release_mem_region(nuc900_audio->res->start, > + resource_size(nuc900_audio->res)); > +out0: > + kfree(nuc900_audio); > + return ret; > +} > + > +static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev) > +{ > + > + snd_soc_unregister_dai(&nuc900_ac97_dai); > + > + clk_put(nuc900_ac97_data->clk); > + iounmap(nuc900_ac97_data->mmio); > + release_mem_region(nuc900_ac97_data->res->start, > + resource_size(nuc900_ac97_data->res)); > + > + nuc900_ac97_data = NULL; > + > + return 0; > +} > + > +static struct platform_driver nuc900_ac97_driver = { > + .driver = { > + .name = "nuc900-audio", > + .owner = THIS_MODULE, > + }, > + .probe = nuc900_ac97_drvprobe, > + .remove = __devexit_p(nuc900_ac97_drvremove), > +}; > + > +static int __init nuc900_ac97_init(void) > +{ > + return platform_driver_register(&nuc900_ac97_driver); > +} > + > +static void __exit nuc900_ac97_exit(void) > +{ > + platform_driver_unregister(&nuc900_ac97_driver); > +} > + > +module_init(nuc900_ac97_init); > +module_exit(nuc900_ac97_exit); > + > +MODULE_AUTHOR("Wan ZongShun <mcuos.com(a)gmail.com>"); > +MODULE_DESCRIPTION("NUC900 AC97 SoC driver!"); > +MODULE_LICENSE("GPL"); > +MODULE_ALIAS("platform:nuc900-ac97"); > + > diff --git a/sound/soc/nuc900/nuc900-audio.c b/sound/soc/nuc900/nuc900-audio.c > new file mode 100644 > index 0000000..b33d5b8 > --- /dev/null > +++ b/sound/soc/nuc900/nuc900-audio.c > @@ -0,0 +1,81 @@ > +/* > + * Copyright (c) 2010 Nuvoton technology corporation. > + * > + * Wan ZongShun <mcuos.com(a)gmail.com> > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation;version 2 of the License. > + * > + */ > + > +#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 "../codecs/ac97.h" > +#include "nuc900-auido.h" > + > +static struct snd_soc_dai_link nuc900evb_ac97_dai = { > + .name = "AC97", > + .stream_name = "AC97 HiFi", > + .cpu_dai = &nuc900_ac97_dai, > + .codec_dai = &ac97_dai, > +}; > + > +static struct snd_soc_card nuc900evb_audio_machine = { > + .name = "NUC900EVB_AC97", > + .dai_link = &nuc900evb_ac97_dai, > + .num_links = 1, > + .platform = &nuc900_soc_platform, > +}; > + > +static struct snd_soc_device nuc900evb_ac97_devdata = { > + .card = &nuc900evb_audio_machine, > + .codec_dev = &soc_codec_dev_ac97, > +}; > + > +static struct platform_device *nuc900evb_asoc_dev; > + > +static int __init nuc900evb_audio_init(void) > +{ > + int ret; > + > + ret = -ENOMEM; > + nuc900evb_asoc_dev = platform_device_alloc("soc-audio", -1); > + if (!nuc900evb_asoc_dev) > + goto out; > + > + /* nuc900 board audio device */ > + platform_set_drvdata(nuc900evb_asoc_dev, &nuc900evb_ac97_devdata); > + > + nuc900evb_ac97_devdata.dev = &nuc900evb_asoc_dev->dev; > + ret = platform_device_add(nuc900evb_asoc_dev); > + > + if (ret) { > + platform_device_put(nuc900evb_asoc_dev); > + nuc900evb_asoc_dev = NULL; > + } > + > +out: > + return ret; > +} > + > +static void __exit nuc900evb_audio_exit(void) > +{ > + platform_device_unregister(nuc900evb_asoc_dev); > +} > + > +module_init(nuc900evb_audio_init); > +module_exit(nuc900evb_audio_exit); > + > +MODULE_LICENSE("GPL"); > +MODULE_DESCRIPTION("NUC900 Series ASoC audio support"); > +MODULE_AUTHOR("Wan ZongShun"); > diff --git a/sound/soc/nuc900/nuc900-auido.h b/sound/soc/nuc900/nuc900-auido.h > new file mode 100644 > index 0000000..95ac4ef > --- /dev/null > +++ b/sound/soc/nuc900/nuc900-auido.h > @@ -0,0 +1,121 @@ > +/* > + * Copyright (c) 2010 Nuvoton technology corporation. > + * > + * Wan ZongShun <mcuos.com(a)gmail.com> > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation;version 2 of the License. > + * > + */ > + > +#ifndef _NUC900_AUDIO_H > +#define _NUC900_AUDIO_H > + > +#include <linux/io.h> > + > +/* Audio Control Registers */ > +#define ACTL_CON 0x00 > +#define ACTL_RESET 0x04 > +#define ACTL_RDSTB 0x08 > +#define ACTL_RDST_LENGTH 0x0C > +#define ACTL_RDSTC 0x10 > +#define ACTL_RSR 0x14 > +#define ACTL_PDSTB 0x18 > +#define ACTL_PDST_LENGTH 0x1C > +#define ACTL_PDSTC 0x20 > +#define ACTL_PSR 0x24 > +#define ACTL_IISCON 0x28 > +#define ACTL_ACCON 0x2C > +#define ACTL_ACOS0 0x30 > +#define ACTL_ACOS1 0x34 > +#define ACTL_ACOS2 0x38 > +#define ACTL_ACIS0 0x3C > +#define ACTL_ACIS1 0x40 > +#define ACTL_ACIS2 0x44 > +#define ACTL_COUNTER 0x48 > + > +/* bit definition of REG_ACTL_CON register */ > +#define R_DMA_IRQ 0x1000 > +#define T_DMA_IRQ 0x0800 > +#define IIS_AC_PIN_SEL 0x0100 > +#define FIFO_TH 0x0080 > +#define ADC_EN 0x0010 > +#define M80_EN 0x0008 > +#define ACLINK_EN 0x0004 > +#define IIS_EN 0x0002 > + > +/* bit definition of REG_ACTL_RESET register */ > +#define W5691_PLAY 0x20000 > +#define ACTL_RESET_BIT 0x10000 > +#define RECORD_RIGHT_CHNNEL 0x08000 > +#define RECORD_LEFT_CHNNEL 0x04000 > +#define PLAY_RIGHT_CHNNEL 0x02000 > +#define PLAY_LEFT_CHNNEL 0x01000 > +#define DAC_PLAY 0x00800 > +#define ADC_RECORD 0x00400 > +#define M80_PLAY 0x00200 > +#define AC_RECORD 0x00100 > +#define AC_PLAY 0x00080 > +#define IIS_RECORD 0x00040 > +#define IIS_PLAY 0x00020 > +#define DAC_RESET 0x00010 > +#define ADC_RESET 0x00008 > +#define M80_RESET 0x00004 > +#define AC_RESET 0x00002 > +#define IIS_RESET 0x00001 > + > +/* bit definition of REG_ACTL_ACCON register */ > +#define AC_BCLK_PU_EN 0x20 > +#define AC_R_FINISH 0x10 > +#define AC_W_FINISH 0x08 > +#define AC_W_RES 0x04 > +#define AC_C_RES 0x02 > + > +/* bit definition of ACTL_RSR register */ > +#define R_FIFO_EMPTY 0x04 > +#define R_DMA_END_IRQ 0x02 > +#define R_DMA_MIDDLE_IRQ 0x01 > + > +/* bit definition of ACTL_PSR register */ > +#define P_FIFO_EMPTY 0x04 > +#define P_DMA_END_IRQ 0x02 > +#define P_DMA_MIDDLE_IRQ 0x01 > + > +/* bit definition of ACTL_ACOS0 register */ > +#define SLOT1_VALID 0x01 > +#define SLOT2_VALID 0x02 > +#define SLOT3_VALID 0x04 > +#define SLOT4_VALID 0x08 > +#define VALID_FRAME 0x10 > + > +/* bit definition of ACTL_ACOS1 register */ > +#define R_WB 0x80 > + > +#define CODEC_READY 0x10 > +#define RESET_PRSR 0x00 > +#define AUDIO_WRITE(addr, val) __raw_writel(val, addr) > +#define AUDIO_READ(addr) __raw_readl(addr) > +#define PCM_TX 0 > +#define PCM_RX 1 > +#define SUBSTREAM_TYPE(substream) \ > + ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX) > + > +struct nuc900_audio { > + void __iomem *mmio; > + spinlock_t lock; > + dma_addr_t dma_addr[2]; > + unsigned long buffersize[2]; > + unsigned long irq_num; > + struct snd_pcm_substream *substream; > + struct resource *res; > + struct clk *clk; > + struct device *dev; > + > +}; > + > +extern struct nuc900_audio *nuc900_ac97_data; > +extern struct snd_soc_dai nuc900_ac97_dai; > +extern struct snd_soc_platform nuc900_soc_platform; > + > +#endif /*end _NUC900_AUDIO_H */ > diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c > new file mode 100644 > index 0000000..32a503c > --- /dev/null > +++ b/sound/soc/nuc900/nuc900-pcm.c > @@ -0,0 +1,352 @@ > +/* > + * Copyright (c) 2010 Nuvoton technology corporation. > + * > + * Wan ZongShun <mcuos.com(a)gmail.com> > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation;version 2 of the License. > + * > + */ > + > +#include <linux/module.h> > +#include <linux/init.h> > +#include <linux/io.h> > +#include <linux/platform_device.h> > +#include <linux/slab.h> > +#include <linux/dma-mapping.h> > + > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/pcm_params.h> > +#include <sound/soc.h> > + > +#include <mach/hardware.h> > + > +#include "nuc900-auido.h" > + > +static const struct snd_pcm_hardware nuc900_pcm_hardware = { > + .info = SNDRV_PCM_INFO_INTERLEAVED | > + SNDRV_PCM_INFO_BLOCK_TRANSFER | > + SNDRV_PCM_INFO_MMAP | > + SNDRV_PCM_INFO_MMAP_VALID | > + SNDRV_PCM_INFO_PAUSE | > + SNDRV_PCM_INFO_RESUME, > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels_min = 1, > + .channels_max = 2, > + .buffer_bytes_max = 4*1024, > + .period_bytes_min = 1*1024, > + .period_bytes_max = 4*1024, > + .periods_min = 1, > + .periods_max = 1024, > +}; > + > +static int nuc900_dma_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + unsigned long flags, stype = SUBSTREAM_TYPE(substream); > + int ret = 0; > + > + spin_lock_irqsave(&nuc900_audio->lock, flags); > + > + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); > + if (ret < 0) > + return ret; > + > + nuc900_audio->substream = substream; > + nuc900_audio->dma_addr[stype] = runtime->dma_addr; > + nuc900_audio->buffersize[stype] = params_buffer_bytes(params); > + > + spin_unlock_irqrestore(&nuc900_audio->lock, flags); > + > + return ret; > +} > + > +static void nuc900_update_dma_register(struct snd_pcm_substream *substream, > + dma_addr_t dma_addr, size_t count) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + void __iomem *mmio_addr, *mmio_len; > + > + if (SUBSTREAM_TYPE(substream) == PCM_TX) { > + mmio_addr = nuc900_audio->mmio + ACTL_PDSTB; > + mmio_len = nuc900_audio->mmio + ACTL_PDST_LENGTH; > + } else { > + mmio_addr = nuc900_audio->mmio + ACTL_RDSTB; > + mmio_len = nuc900_audio->mmio + ACTL_RDST_LENGTH; > + } > + > + AUDIO_WRITE(mmio_addr, dma_addr); > + AUDIO_WRITE(mmio_len, count); > +} > + > +static void nuc900_dma_start(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + unsigned long val; > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); > + val |= (T_DMA_IRQ | R_DMA_IRQ); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val); > +} > + > +static void nuc900_dma_stop(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + unsigned long val; > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); > + val &= ~(T_DMA_IRQ | R_DMA_IRQ); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val); > +} > + > +static irqreturn_t nuc900_dma_interrupt(int irq, void *dev_id) > +{ > + struct snd_pcm_substream *substream = dev_id; > + struct nuc900_audio *nuc900_audio = substream->runtime->private_data; > + unsigned long val; > + > + spin_lock(&nuc900_audio->lock); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); > + > + if (val & R_DMA_IRQ) { > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | R_DMA_IRQ); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR); > + > + if (val & R_DMA_MIDDLE_IRQ) { > + val |= R_DMA_MIDDLE_IRQ; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val); > + } > + > + if (val & R_DMA_END_IRQ) { > + val |= R_DMA_END_IRQ; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val); > + } > + } else if (val & T_DMA_IRQ) { > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | T_DMA_IRQ); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR); > + > + if (val & P_DMA_MIDDLE_IRQ) { > + val |= P_DMA_MIDDLE_IRQ; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val); > + } > + > + if (val & P_DMA_END_IRQ) { > + val |= P_DMA_END_IRQ; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val); > + } > + } else { > + dev_err(nuc900_audio->dev, "Wrong DMA interrupt status!\n"); > + spin_unlock(&nuc900_audio->lock); > + return IRQ_HANDLED; > + } > + > + spin_unlock(&nuc900_audio->lock); > + > + snd_pcm_period_elapsed(substream); > + > + return IRQ_HANDLED; > +} > + > +static int nuc900_dma_hw_free(struct snd_pcm_substream *substream) > +{ > + snd_pcm_lib_free_pages(substream); > + return 0; > +} > + > +static int nuc900_dma_prepare(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + unsigned long flags, val, stype = SUBSTREAM_TYPE(substream);; > + > + spin_lock_irqsave(&nuc900_audio->lock, flags); > + > + nuc900_update_dma_register(substream, > + nuc900_audio->dma_addr[stype], nuc900_audio->buffersize[stype]); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + > + switch (runtime->channels) { > + case 1: > + if (PCM_TX == stype) { > + val &= ~(PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL); > + val |= PLAY_RIGHT_CHNNEL; > + } else { > + val &= ~(RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL); > + val |= RECORD_RIGHT_CHNNEL; > + } > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + break; > + case 2: > + if (PCM_TX == stype) > + val |= (PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL); > + else > + val |= (RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + break; > + default: > + return -EINVAL; > + } > + spin_unlock_irqrestore(&nuc900_audio->lock, flags); > + return 0; > +} > + > +static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd) > +{ > + int ret = 0; > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + case SNDRV_PCM_TRIGGER_RESUME: > + nuc900_dma_start(substream); > + break; > + > + case SNDRV_PCM_TRIGGER_STOP: > + case SNDRV_PCM_TRIGGER_SUSPEND: > + nuc900_dma_stop(substream); > + break; > + > + default: > + ret = -EINVAL; > + break; > + } > + > + return ret; > +} > + > +int nuc900_dma_getposition(struct snd_pcm_substream *substream, > + dma_addr_t *src, dma_addr_t *dst) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + > + if (src != NULL) > + *src = AUDIO_READ(nuc900_audio->mmio + ACTL_PDSTC); > + > + if (dst != NULL) > + *dst = AUDIO_READ(nuc900_audio->mmio + ACTL_RDSTC); > + > + return 0; > +} > + > +static snd_pcm_uframes_t nuc900_dma_pointer(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + dma_addr_t src, dst; > + unsigned long res; > + > + nuc900_dma_getposition(substream, &src, &dst); > + > + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) > + res = dst - runtime->dma_addr; > + else > + res = src - runtime->dma_addr; > + > + return bytes_to_frames(substream->runtime, res); > +} > + > +static int nuc900_dma_open(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio; > + > + snd_soc_set_runtime_hwparams(substream, &nuc900_pcm_hardware); > + > + nuc900_audio = nuc900_ac97_data; > + > + if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt, > + IRQF_DISABLED, "nuc900-dma", substream)) > + return -EBUSY; > + > + runtime->private_data = nuc900_audio; > + > + return 0; > +} > + > +static int nuc900_dma_close(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + > + free_irq(nuc900_audio->irq_num, substream); > + > + return 0; > +} > + > +static int nuc900_dma_mmap(struct snd_pcm_substream *substream, > + struct vm_area_struct *vma) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + > + return dma_mmap_writecombine(substream->pcm->card->dev, vma, > + runtime->dma_area, > + runtime->dma_addr, > + runtime->dma_bytes); > +} > + > +static struct snd_pcm_ops nuc900_dma_ops = { > + .open = nuc900_dma_open, > + .close = nuc900_dma_close, > + .ioctl = snd_pcm_lib_ioctl, > + .hw_params = nuc900_dma_hw_params, > + .hw_free = nuc900_dma_hw_free, > + .prepare = nuc900_dma_prepare, > + .trigger = nuc900_dma_trigger, > + .pointer = nuc900_dma_pointer, > + .mmap = nuc900_dma_mmap, > +}; > + > +static void nuc900_dma_free_dma_buffers(struct snd_pcm *pcm) > +{ > + snd_pcm_lib_preallocate_free_for_all(pcm); > +} > + > +static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); > +static int nuc900_dma_new(struct snd_card *card, > + struct snd_soc_dai *dai, struct snd_pcm *pcm) > +{ > + if (!card->dev->dma_mask) > + card->dev->dma_mask = &nuc900_pcm_dmamask; > + if (!card->dev->coherent_dma_mask) > + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + > + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, > + card->dev, 4 * 1024, (4 * 1024) - 1); > + > + return 0; > +} > + > +struct snd_soc_platform nuc900_soc_platform = { > + .name = "nuc900-dma", > + .pcm_ops = &nuc900_dma_ops, > + .pcm_new = nuc900_dma_new, > + .pcm_free = nuc900_dma_free_dma_buffers, > +} > +EXPORT_SYMBOL_GPL(nuc900_soc_platform); > + > +static int __init nuc900_soc_platform_init(void) > +{ > + return snd_soc_register_platform(&nuc900_soc_platform); > +} > + > +static void __exit nuc900_soc_platform_exit(void) > +{ > + snd_soc_unregister_platform(&nuc900_soc_platform); > +} > + > +module_init(nuc900_soc_platform_init); > +module_exit(nuc900_soc_platform_exit); > + > +MODULE_AUTHOR("Wan ZongShun, <mcuos.com(a)gmail.com>"); > +MODULE_DESCRIPTION("nuc900 Audio DMA module"); > +MODULE_LICENSE("GPL"); > -- > 1.6.3.3 > -- *linux-arm-kernel mailing list mail addr:linux-arm-kernel@lists.infradead.org you can subscribe by: http://lists.infradead.org/mailman/listinfo/linux-arm-kernel * linux-arm-NUC900 mailing list mail addr:NUC900@googlegroups.com main web: https://groups.google.com/group/NUC900 you can subscribe it by sending me mail: mcuos.com(a)gmail.com

3 4

[alsa-devel] ASoC: Dummy codec driver?
by Joonyoung Shim 19 May '10

19 May '10

Hi, I have a target(The cpu is S5PC110) which doesn't have audio codec, but it needs the audio device node to play audio at the TV via HDMI. The S5PC110 HDMI transmits itself to TV the audio data from I2S buffer, so i have made dummy codec driver to create audio device node and to transmit audio data to I2S buffer. How do you think about the dummy codec driver?

2 1

[alsa-devel] [PATCH v2 0/2] Add earphone speaker in OMAP4
by Jorge Eduardo Candelaria 19 May '10

19 May '10

Add support for earphone speaker in SDP4430 board. Changes from v1: - Separated changes for codec and machine drivers in two patches. - Added Acked-by tag to patches. Jorge Eduardo Candelaria (2): ASoC: SDP4430: Add support for Earphone speaker ASoC: TWL6040: Enable earphone path in codec sound/soc/codecs/twl6040.c | 21 ++++++++++++++++++++- sound/soc/omap/sdp4430.c | 4 ++++ 2 files changed, 24 insertions(+), 1 deletions(-)

2 3

[alsa-devel] [PATCH 1/2] ASoC: SDP4430: Add sdp4430 machine driver
by Jorge Eduardo Candelaria 19 May '10

19 May '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Add ASoC support for TI SDP4430. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> --- sound/soc/omap/Kconfig | 9 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp4430.c | 228 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/sdp4430.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4292701..c23c4a6 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -102,6 +102,15 @@ config SND_OMAP_SOC_SDP3430 Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. +config SND_OMAP_SOC_SDP4430 + tristate "SoC Audio support for Texas Instruments SDP4430" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP + select SND_OMAP_SOC_MCPDM + select SND_SOC_TWL6040 + help + Say Y if you want to add support for SoC audio on Texas Instruments + SDP4430. + config SND_OMAP_SOC_OMAP3_PANDORA tristate "SoC Audio support for OMAP3 Pandora" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3a75755..ba9fc65 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -17,6 +17,7 @@ snd-soc-omap2evm-objs := omap2evm.o snd-soc-omap3evm-objs := omap3evm.o snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o +snd-soc-sdp4430-objs := sdp4430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o @@ -31,6 +32,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o +obj-$(CONFIG_SND_OMAP_SOC_SDP4430) += snd-soc-sdp4430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c new file mode 100644 index 0000000..83831ee --- /dev/null +++ b/sound/soc/omap/sdp4430.c @@ -0,0 +1,228 @@ +/* + * sdp4430.c -- SoC audio for TI OMAP4430 SDP + * + * Author: Misael Lopez Cruz <x0052729(a)ti.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 <plat/hardware.h> +#include <plat/mux.h> + +#include "mcpdm.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" +#include "../codecs/twl6040.h" + +static int twl6040_power_mode; + +static int sdp4430_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; + int clk_id, freq; + int ret; + + if (twl6040_power_mode) { + clk_id = TWL6040_SYSCLK_SEL_HPPLL; + freq = 38400000; + } else { + clk_id = TWL6040_SYSCLK_SEL_LPPLL; + freq = 32768; + } + + /* set the codec mclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq, + SND_SOC_CLOCK_IN); + if (ret) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } +} + +static struct snd_soc_ops sdp4430_ops = { + .hw_params = sdp4430_hw_params, +}; + +static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = twl6040_power_mode; + return 0; +} + +static int sdp4430_set_power_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (twl6040_power_mode == ucontrol->value.integer.value[0]) + return 0; + + twl6040_power_mode = ucontrol->value.integer.value[0]; + + return 1; +} + +static const char *power_texts[] = {"Low-Power", "High-Performance"}; + +static const struct soc_enum sdp4430_enum[] = { + SOC_ENUM_SINGLE_EXT(2, power_texts), +}; + +static const struct snd_kcontrol_new sdp4430_controls[] = { + SOC_ENUM_EXT("TWL6040 Power Mode", sdp4430_enum[0], + sdp4430_get_power_mode, sdp4430_set_power_mode), +}; + +/* SDP4430 machine DAPM */ +static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Main Mic Bias"}, + {"SUBMIC", NULL, "Main Mic Bias"}, + {"Main Mic Bias", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Headset Stereophone (Headphone): HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, +}; + +static int sdp4430_twl6040_init(struct snd_soc_codec *codec) +{ + int ret; + + /* Add SDP4430 specific controls */ + ret = snd_soc_add_controls(codec, sdp4430_controls, + ARRAY_SIZE(sdp4430_controls)); + if (ret) + return ret; + + /* Add SDP4430 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, sdp4430_twl6040_dapm_widgets, + ARRAY_SIZE(sdp4430_twl6040_dapm_widgets)); + if (ret) + return ret; + + /* Set up SDP4430 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* SDP4430 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Stereophone"); + + /* TWL6040 not connected pins */ + snd_soc_dapm_nc_pin(codec, "AFML"); + snd_soc_dapm_nc_pin(codec, "AFMR"); + + ret = snd_soc_dapm_sync(codec); + + return ret; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sdp4430_dai = { + .name = "TWL6040", + .stream_name = "TWL6040", + .cpu_dai = &omap_mcpdm_dai, + .codec_dai = &twl6040_dai, + .init = sdp4430_twl6040_init, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_sdp4430 = { + .name = "SDP4430", + .platform = &omap_soc_platform, + .dai_link = &sdp4430_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device sdp4430_snd_devdata = { + .card = &snd_soc_sdp4430, + .codec_dev = &soc_codec_dev_twl6040, +}; + +static struct platform_device *sdp4430_snd_device; + +static int __init sdp4430_soc_init(void) +{ + int ret; + + if (!machine_is_omap_4430sdp()) { + pr_debug("Not SDP4430!\n"); + return -ENODEV; + } + printk(KERN_INFO "SDP4430 SoC init\n"); + + sdp4430_snd_device = platform_device_alloc("soc-audio", -1); + if (!sdp4430_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sdp4430_snd_device, &sdp4430_snd_devdata); + sdp4430_snd_devdata.dev = &sdp4430_snd_device->dev; + + ret = platform_device_add(sdp4430_snd_device); + if (ret) + goto err; + + /* Codec starts in HP mode */ + twl6040_power_mode = 1; + + return 0; + +err: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(sdp4430_snd_device); + return ret; +} +module_init(sdp4430_soc_init); + +static void __exit sdp4430_soc_exit(void) +{ + platform_device_unregister(sdp4430_snd_device); +} +module_exit(sdp4430_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC SDP4430"); +MODULE_LICENSE("GPL"); + -- 1.6.3.3

3 2

[alsa-devel] [PATCH - alsa-lib] tlv: Check dB range only within the control's volume range
by Peter Ujfalusi 19 May '10

19 May '10

The DB_RANGE need to be used on some HW, since the gain on volume control is not continuous, and has to be divided into several sub DB_SCALE ranges. ASoC has a feature to override the HW default volume range, and in this case when the volume range is less than the HW maximum we do not need to go through the whole DB_RANGE, but we need to stop where the kcontrol's maximum tell us. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello, this patch fixes the ASoC related problem, when the volume is limited by the machine driver on a given control, which happen to have DB_RANGE gain mapping. The DB_RANGE has to have _only_ DB_SCALE type of mapping, since with LINEAR, MINMAX, MINMAX_MUTE it is not possible to accurately calculate the maximum dB limit on the control, when the volume is limited. Applications like PulseAudio need to know the actual dB range, and not only the theoretical maximum. src/control/tlv.c | 11 ++++++++--- 1 files changed, 8 insertions(+), 3 deletions(-) diff --git a/src/control/tlv.c b/src/control/tlv.c index cfd9b97..0ff052e 100644 --- a/src/control/tlv.c +++ b/src/control/tlv.c @@ -140,10 +140,13 @@ int snd_tlv_get_dB_range(unsigned int *tlv, long rangemin, long rangemax, pos = 2; while (pos + 4 <= len) { long rmin, rmax; - rangemin = (int)tlv[pos]; - rangemax = (int)tlv[pos + 1]; + long submin, submax; + submin = (int)tlv[pos]; + submax = (int)tlv[pos + 1]; + if (rangemax < submax) + submax = rangemax; err = snd_tlv_get_dB_range(tlv + pos + 2, - rangemin, rangemax, + submin, submax, &rmin, &rmax); if (err < 0) return err; @@ -156,6 +159,8 @@ int snd_tlv_get_dB_range(unsigned int *tlv, long rangemin, long rangemax, *min = rmin; *max = rmax; } + if (rangemax == submax) + return 0; pos += int_index(tlv[pos + 3]) + 4; } return 0; -- 1.7.1

2 4

[alsa-devel] [PATCH] ASoC: tpa6130a2: Remove CPVSS supply
by Jarkko Nikula 19 May '10

19 May '10

I believe there is no need for the CPVSS supply as this pin is for decoupling capacitor for the internal charge pump. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- sound/soc/codecs/tpa6130a2.c | 40 +++++++++++++++++++++------------------- 1 files changed, 21 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 221bb90..87a5b09 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -36,13 +36,14 @@ static struct i2c_client *tpa6130a2_client; -#define TPA6130A2_NUM_SUPPLIES 2 +#define TPA6130A2_NUM_SUPPLIES 1 +#define TPA6140A2_NUM_SUPPLIES 2 +#define TPA61x0A2_MAX_SUPPLIES 2 static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { - "CPVSS", "Vdd", }; -static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { +static const char *tpa6140a2_supply_names[TPA6140A2_NUM_SUPPLIES] = { "HPVdd", "AVdd", }; @@ -51,7 +52,8 @@ static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { struct tpa6130a2_data { struct mutex mutex; unsigned char regs[TPA6130A2_CACHEREGNUM]; - struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES]; + struct regulator_bulk_data supplies[TPA61x0A2_MAX_SUPPLIES]; + int num_supplies; int power_gpio; unsigned char power_state; enum tpa_model id; @@ -136,7 +138,7 @@ static int tpa6130a2_power(int power) if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); - ret = regulator_bulk_enable(ARRAY_SIZE(data->supplies), + ret = regulator_bulk_enable(data->num_supplies, data->supplies); if (ret != 0) { dev_err(&tpa6130a2_client->dev, @@ -161,7 +163,7 @@ static int tpa6130a2_power(int power) if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 0); - ret = regulator_bulk_disable(ARRAY_SIZE(data->supplies), + ret = regulator_bulk_disable(data->num_supplies, data->supplies); if (ret != 0) { dev_err(&tpa6130a2_client->dev, @@ -412,6 +414,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, struct device *dev; struct tpa6130a2_data *data; struct tpa6130a2_platform_data *pdata; + const char **supply_names; int i, ret; dev = &client->dev; @@ -454,23 +457,22 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, } switch (data->id) { + default: + dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", + pdata->id); case TPA6130A2: - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; + data->num_supplies = TPA6130A2_NUM_SUPPLIES; + supply_names = tpa6130a2_supply_names; break; case TPA6140A2: - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6140a2_supply_names[i];; + data->num_supplies = TPA6140A2_NUM_SUPPLIES; + supply_names = tpa6140a2_supply_names; break; - default: - dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", - pdata->id); - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; } + for (i = 0; i < data->num_supplies; i++) + data->supplies[i].supply = supply_names[i]; - ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), - data->supplies); + ret = regulator_bulk_get(dev, data->num_supplies, data->supplies); if (ret != 0) { dev_err(dev, "Failed to request supplies: %d\n", ret); goto err_regulator; @@ -495,7 +497,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, return 0; err_power: - regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + regulator_bulk_free(data->num_supplies, data->supplies); err_regulator: if (data->power_gpio >= 0) gpio_free(data->power_gpio); @@ -516,7 +518,7 @@ static int __devexit tpa6130a2_remove(struct i2c_client *client) if (data->power_gpio >= 0) gpio_free(data->power_gpio); - regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + regulator_bulk_free(data->num_supplies, data->supplies); kfree(data); tpa6130a2_client = NULL; -- 1.7.1

4 5

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: add support for Akai MPD16
by wdev＠foltman.com 18 May '10

18 May '10

From: Krzysztof Foltman <wdev(a)foltman.com> The decoding routine is based on own reverse-engineering. It seems to recognize all the messages that MPD16 sends during normal operation (that is, via standard MIDI messages). Configuration (changing pad sensitivity, slider controller and MIDI notes) is not supported in this version. Signed-off-by: Krzysztof Foltman <wdev(a)foltman.com> diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 2c1558c..57d642a 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -645,6 +645,34 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = { }; /* + * AKAI MPD16 protocol: one or more chunks consisting of first byte of + * (0x20 | msg_len) and then a MIDI message (msg_len bytes long) + * + * Messages sent: + * 21 FE (active sense) + * 23 90 xx xx (note on) + * 23 Ax xx xx (polyphonic pressure) + * 23 Bx xx xx (control change) + */ +static void snd_usbmidi_akai_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) +{ + unsigned int pos = 0; + while (pos < (unsigned)buffer_length && (buffer[pos] & 0xF8) == 0x20) + { + int msg_len = buffer[pos] & 0x0f; + snd_usbmidi_input_data(ep, 0, &buffer[pos + 1], msg_len); + pos += 1 + msg_len; + } +} + +static struct usb_protocol_ops snd_usbmidi_akai_ops = { + .input = snd_usbmidi_akai_input, + .output = snd_usbmidi_standard_output, + .output_packet = snd_usbmidi_output_standard_packet, +}; + +/* * Novation USB MIDI protocol: number of data bytes is in the first byte * (when receiving) (+1!) or in the second byte (when sending); data begins * at the third byte. @@ -1434,6 +1462,8 @@ static struct port_info { EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), + /* Akai MPD16 */ + EXTERNAL_PORT(0x09e8, 0x0062, 0, "%s MIDI"), /* Access Music Virus TI */ EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"), PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0, @@ -2035,6 +2065,13 @@ int snd_usbmidi_create(struct snd_card *card, umidi->usb_protocol_ops = &snd_usbmidi_cme_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_AKAI: + umidi->usb_protocol_ops = &snd_usbmidi_akai_ops; + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + /* endpoint 0 does not carry MIDI data */ + endpoints[0].out_cables = 0; + endpoints[0].in_cables = 0; + break; default: snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); err = -ENXIO; diff --git a/sound/usb/midi.h b/sound/usb/midi.h index 2089ec9..2fca80b 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -37,6 +37,8 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_MIDI_CME, data is NULL */ +/* for QUIRK_MIDI_AKAI, data is NULL */ + int snd_usbmidi_create(struct snd_card *card, struct usb_interface *iface, struct list_head *midi_list, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 91ddef3..f8797f6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1973,6 +1973,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* AKAI devices */ +{ + USB_DEVICE(0x09e8, 0x0062), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "AKAI", + .product_name = "MPD16", + .ifnum = 0, + .type = QUIRK_MIDI_AKAI, + } +}, + /* TerraTec devices */ { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 136e5b4..b45e54c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -289,6 +289,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, + [QUIRK_MIDI_AKAI] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index d679e72..06ebf24 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -74,6 +74,7 @@ enum quirk_type { QUIRK_MIDI_FASTLANE, QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, + QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, -- 1.7.0.4

3 2

[alsa-devel] [GIT PULL] ASoC updates for 2.6.35
by Liam Girdwood 18 May '10

18 May '10

Hi Takashi, Please pull the following ASoC updates for 2.6.35. We do have Tony's Ack for the OMAP mcbsp stuff (as it simplifies the merging). Thanks Liam --- The following changes since commit 6a2f1ee1f9bb5346644105c9355e9e06f6a847d3: Mark Brown (1): ASoC: Don't restart unconfigured WM8994 FLLs are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.35 Felipe Balbi (1): ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function Jorge Eduardo Candelaria (2): ARM: McBSP: Fix request for irq in OMAP4 ARM: McBSP: Add support for omap4 in McBSP driver Peter Ujfalusi (2): ASoC: TWL4030: Add control for digimic Left Right swap ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF Sergey Lapin (1): OMAP: McBSP: Add 32-bit mode support arch/arm/mach-omap2/mcbsp.c | 12 ++++-------- arch/arm/plat-omap/mcbsp.c | 34 +++++++++++++++++++--------------- sound/soc/codecs/tlv320dac33.c | 5 ++++- sound/soc/codecs/twl4030.c | 12 ++++++++++++ sound/soc/omap/omap-mcbsp.c | 28 ++++++++++++++++++++++++---- 5 files changed, 63 insertions(+), 28 deletions(-)

2 1

[alsa-devel] ASoC: OMAP4: Add support for Earphone speaker
by Jorge Eduardo Candelaria 18 May '10

18 May '10

Add control to enable earphone driver in TWL6040 codec. This driver is connected to HSDAC Left. Also enable earphone speaker in sdp4430 machine driver. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 21 ++++++++++++++++++++- sound/soc/omap/sdp4430.c | 5 +++++ 2 files changed, 25 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2ae442e..af36346 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -432,6 +432,12 @@ static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); */ static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); +/* + * EPGAIN volume control: + * from -24 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(ep_tlv, -2400, 200, 0); + /* Left analog microphone selection */ static const char *twl6040_amicl_texts[] = {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; @@ -479,6 +485,9 @@ static const struct snd_kcontrol_new hfl_driver_switch_controls = static const struct snd_kcontrol_new hfr_driver_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); +static const struct snd_kcontrol_new ep_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); + static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* Capture gains */ SOC_DOUBLE_TLV("Capture Preamplifier Volume", @@ -491,7 +500,8 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), SOC_DOUBLE_R_TLV("Handsfree Playback Volume", TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), - + SOC_SINGLE_TLV("Earphone Playback Volume", + TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv), }; static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { @@ -507,6 +517,7 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HSOR"), SND_SOC_DAPM_OUTPUT("HFL"), SND_SOC_DAPM_OUTPUT("HFR"), + SND_SOC_DAPM_OUTPUT("EP"), /* Analog input muxes for the capture amplifiers */ SND_SOC_DAPM_MUX("Analog Left Capture Route", @@ -572,6 +583,10 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH_E("Earphone Driver", + SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback PGAs */ SND_SOC_DAPM_PGA("HFDAC Left PGA", @@ -607,6 +622,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSOL", NULL, "Headset Left Driver"}, {"HSOR", NULL, "Headset Right Driver"}, + /* Earphone playback path */ + {"Earphone Driver", "Switch", "HSDAC Left"}, + {"EP", NULL, "Earphone Driver"}, + /* Handsfree playback path */ {"HFDAC Left Playback", "Switch", "HFDAC Left"}, {"HFDAC Right Playback", "Switch", "HFDAC Right"}, diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 83831ee..ab45621 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -101,6 +101,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_SPK("Earphone Spk", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -120,6 +121,10 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Headset Stereophone (Headphone): HSOL, HSOR */ {"Headset Stereophone", NULL, "HSOL"}, {"Headset Stereophone", NULL, "HSOR"}, + + /* Earphone speaker */ + {"Earphone Spk", NULL, "EP"}, + }; static int sdp4430_twl6040_init(struct snd_soc_codec *codec) -- 1.6.3.3

3 2

[alsa-devel] Creating a /dev entry.
by Adish Kuvelker 18 May '10

18 May '10

Hello, I am writing a ALSA driver for the AtlasIII AC'97 Controller. The controller communicates with the AC'97 Codec and the necessary PCM, Sequencer, Control and Timer nodes are created in the /sys. Also I get the necessary information registered in the procfs under /proc/asound. However, I do not see any sound nodes created under /dev. Are these nodes created by the sound driver registration function which creates necessary files under /sys or will I have to use utilities like "snddevices" to create the "snd" node under /dev. I read that "snddevices" would create the necessary entries under /dev/snd, and I reckon that it does not create the entry "snd" under /dev. Although I am using kernel 2.6.19.2, it doesn't have the "udev" enabled. Will I need the "udev" to create these nodes under /dev? Regards; Aadish

2 1

[alsa-devel] RFC:
by Peter Ujfalusi 18 May '10

18 May '10

Hello, I apologize for the long mail, but I feel that I need to explain this in a bit longish way... ASoC core recently gained support for limiting the volume control's range (maximum), which is really useful in embedded systems. The implementation for this is really simple: The codec drivers are implementing the full range the HW can support. Machine drivers can than override the maximum volume on a given control if it is necessary. This works well with most of the controls: Volume controls without dB scale. Volume controls with DECLARE_TLV_DB_SCALE() Problems started to emerge, when the volume is limited on control, which has SNDRV_CTL_TLVT_DB_RANGE (from the sound/soc/codecs/tpa6130a2.c): static const unsigned int tpa6140_tlv[] = { TLV_DB_RANGE_HEAD(3), 0, 8, TLV_DB_SCALE_ITEM(-5900, 400, 0), 9, 16, TLV_DB_SCALE_ITEM(-2500, 200, 0), 17, 31, TLV_DB_SCALE_ITEM(-1000, 100, 0), }; static const struct snd_kcontrol_new tpa6140a2_controls[] = { SOC_SINGLE_EXT_TLV("TPA6140A2 Headphone Playback Volume", TPA6130A2_REG_VOL_MUTE, 1, 0x1f, 0, tpa6130a2_get_volsw, tpa6130a2_put_volsw, tpa6140_tlv), }; The original HW supported range is 0 .. 31 (-59 .. +4 dB). When the machine driver sets the limit to 21 (-6 dB), than the volume range is reported correctly as 0 .. 21, but snd_mixer_selem_get_playback_dB_range() reports range of -59 .. +4 dB. This is not correct, it should report: -59 .. -6 dB. If the control has single DECLARE_TLV_DB_SCALE() after the limiting both volume and dB range is reported correctly. The explanation for this is in the alsa-lib: src/control/tlv.c: /** * \brief Get the dB min/max values * \param tlv the TLV source returned by #snd_tlv_parse_dB_info() * \param rangemin the minimum value of the raw volume * \param rangemax the maximum value of the raw volume * \param min the pointer to store the minimum dB value (in 0.01dB unit) * \param max the pointer to store the maximum dB value (in 0.01dB unit) * \return 0 if successful, or a negative error code */ int snd_tlv_get_dB_range(unsigned int *tlv, long rangemin, long rangemax, long *min, long *max) { int err; switch (tlv[0]) { case SND_CTL_TLVT_DB_RANGE: { unsigned int pos, len; len = int_index(tlv[1]); if (len > MAX_TLV_RANGE_SIZE) return -EINVAL; pos = 2; while (pos + 4 <= len) { long rmin, rmax; rangemin = (int)tlv[pos]; rangemax = (int)tlv[pos + 1]; err = snd_tlv_get_dB_range(tlv + pos + 2, rangemin, rangemax, &rmin, &rmax); if (err < 0) return err; if (pos > 2) { if (rmin < *min) *min = rmin; if (rmax > *max) *max = rmax; } else { *min = rmin; *max = rmax; } pos += int_index(tlv[pos + 3]) + 4; } return 0; } case SND_CTL_TLVT_DB_SCALE: { int step; *min = (int)tlv[2]; step = (tlv[3] & 0xffff); *max = *min + (long)(step * (rangemax - rangemin)); return 0; } case SND_CTL_TLVT_DB_MINMAX: case SND_CTL_TLVT_DB_MINMAX_MUTE: case SND_CTL_TLVT_DB_LINEAR: *min = (int)tlv[2]; *max = (int)tlv[3]; return 0; } return -EINVAL; } In case of DECLARE_TLV_DB_SCALE(), the SND_CTL_TLVT_DB_SCALE case is taken, and the control's maximum, and minimum range are used for the dB scale limit calculation. But, when SNDRV_CTL_TLVT_DB_RANGE is used (tpa6130a2 driver), than that case will dismisses the control's maximum and minimum range, and goes through the whole TLV struct, and reports the HW supported range. If it would stop at the driver reported maximum, than the dB range reported would be correct. So a 'simple' patch to alsa-lib would solve this. But what happens, if we want to limit control, which has DB_MINMAX, DB_MINMAX_MUTE, or DB_LINEAR? There is no way to actually limit those in a way, that the dB scale reported would be correct. We just stretch the steps bigger, like what happens with the SNDRV_CTL_TLVT_DB_RANGE in the tpa6130a2 driver. When looking for users of DB_LINEAR I have found 8 drivers, DB_MINMAX is used by usbmixer only, the DB_MINMAX_MUTE has no users at all. Since both DB_LINEAR and DB_MINMAX can be presented as DB_SCALE, and the effort to convert the drivers should not be that big. In this way, we will only have drivers using DB_SCALE in kernel, we can mark the other DB_ types as deprecated (but keeping the support for out of tree drivers), and the alsa-lib's tlv code can be adjusted to support properly the volume limiting. If there is interest, than the ALSA core could also gain support for platform/configuration dependent volume limiting. If this is too big change to ALSA (which I think it is not, since for the user it makes no difference, and the volume range, dB scale not going to be changed), than probably cleaning up the sound/soc/ to use _only_ DB_SCALE type for volume controls might be the solution, since the volume limiting feature is only in ASoC AFAIK. Either way, the alsa-lib's src/control/tlv.c:snd_tlv_get_dB_range function in case of SND_CTL_TLVT_DB_RANGE needs some fix to handle the situation, when the volume range is _not_ at the HW supported range. To summarize, I'd like to do (or ask help to do) one of the following [1] Replace the use of DB_LINEAR, DB_MINMAX with DB_SCALE in all drivers under sound/ Document, that the now unused DB_ types are deprecated Fix alsa-lib to support the volume limiting in regards of dB scale [2] Replace the use of DB_LINEAR with DB_SCALE in all drivers under sound/soc/ Fix alsa-lib to support the volume limiting in regards of dB scale What do you think? -- Péter

1 1

Re: [alsa-devel] [PATCH] NUC900/audio: add nuc900 audio driver support
by Mark Brown 18 May '10

18 May '10

On Sun, May 16, 2010 at 11:34:01PM +0800, Wan ZongShun wrote: This all looks pretty good. A few comments below but they're all fairly minor and should be easy to address. > index b1749bc..81d4848 100644 > --- a/sound/soc/Kconfig > +++ b/sound/soc/Kconfig > @@ -36,6 +36,7 @@ source "sound/soc/s3c24xx/Kconfig" > source "sound/soc/s6000/Kconfig" > source "sound/soc/sh/Kconfig" > source "sound/soc/txx9/Kconfig" > +source "sound/soc/nuc900/Kconfig" Please keep the Kconfig and Makefile sorted, this helps avoid merge issues. > +#define NUC900_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ > + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ > + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) SNDRV_PCM_RATE_8000_48000. > + if (!timeout) > + dev_err(nuc900_audio->dev, "AC97 read register time out !\n"); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0) ; If the read timed out shouldn't we be returning rather than continuing? > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + val &= (~AC_C_RES); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); > + udelay(1000); > + if (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS0) & CODEC_READY)) > + dev_err(nuc900_audio->dev, "AC97 codec cold reset failed!\n"); What is this actually checking in the hardware? Not all CODECs enable the AC97 link by default after a cold reset, the standard allows them to power up in a low power state which will On the other hand, given that there's no warm reset operation perhaps this isn't a big deal. > +static void nuc900_ac97_remove(struct platform_device *pdev, > + struct snd_soc_dai *dai) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + > + /* Enable unit clock */ > + clk_disable(nuc900_audio->clk); Bit rot in the comments here. > +/* bit definition of REG_ACTL_CON register */ > +#define AUDCLK_EN 0x8000 > +#define PFIFO_EN 0x4000 > +#define RFIFO_EN 0x2000 These constants (and the rest) really should be namespaced - they're likely to collide with other definitions in, for example, CODEC drivers used by machines. > +#define IIS_EN 0x0002 Looks like there's I2S support to come? > +static irqreturn_t nuc900_dma_interrupt(int irq, void *dev_id) > +{ > + > + snd_pcm_period_elapsed(substream); > + > + return IRQ_HANDLED; This is done unconditionally - are you sure there can't be any spurious interrupts (eg, error interrupts). It shouldn't cause any harm, of course.

2 4

[alsa-devel] soc multi chip support
by Attila RS 17 May '10

17 May '10

Hi, I have 2 of the same audio codecs connected to an omap (each codec has it's own i2c address and each has its own mcbsp channel). I can configure and use each one individually, but have not found a way to configure both at the same time so that each codec shows up as an alsa device. I did some searching and found that this may not be supported yet. Is that the case or has anyone done this yet or is working on it? Attila

2 3

[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33: small power handling updates
by Peter Ujfalusi 17 May '10

17 May '10

Hello, While testing suspend, it has been noticed, that the driver would try to power off the codec more than once. Since this case handled in the driver, and results no functional defect we can lower the noise in the kernel log. Also avoid the root cause of the multiple off calls by filtering BIAS_OFF request when the codec was already OFF. --- Felipe Balbi (1): ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function Peter Ujfalusi (1): ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF sound/soc/codecs/tlv320dac33.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-)

3 4

[alsa-devel] enhanced Creative X-Fi support
by Sebastian Reinhardt 17 May '10

17 May '10

@Takashi: I own a Creative X-Fi Elite Pro and I use openSuSE. Since few months, the base card is supported by alsa. Big thanks, great work. But I like to use also the external I/O- terminal, delivered with ElitePro- cards. Is there any chance to get this working. I'm no real linux kernel hacker, but I can give You every infomation about the terminal, if You need (I had already disassembled the I/O- box). I do not need the effects, but In like to use the additional in-/outputs and volume knob. To use the ir- control, would be great, too. -- Kind Regaards Sebastian Reinhardt

1 0

[alsa-devel] [patch -next] asihpi: testing the wrong variable
by Dan Carpenter 17 May '10

17 May '10

There is a typo here. We want to test "*dst" not "dst". Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 254c580..15f0c7e 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -3279,7 +3279,7 @@ static u16 hpi_entity_alloc_and_copy(struct hpi_entity *src, buf_size = hpi_entity_size(src); *dst = kmalloc(buf_size, GFP_KERNEL); - if (dst == NULL) + if (*dst == NULL) return HPI_ERROR_MEMORY_ALLOC; memcpy(*dst, src, buf_size); return 0;

2 1

[alsa-devel] [patch -next] sound/es1688: add pedantic range checks
by Dan Carpenter 17 May '10

17 May '10

Smatch complains that if (dev == SNDRV_CARDS) we're one past the end of the array. That's unlikely to happen in real life, I suppose. Also smatch complains about "strcpy(card->shortname, pcm->name);" The "pcm->name" buffer is 80 characters and "card->shortname" is 32 characters. If you follow the call paths it turns out we never actually use more than 16 characters so it's not a problem. But anyway, let's make it easy for people auditing this in the future. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index fdcce31..0cde813 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -149,10 +149,11 @@ static int __devinit snd_es1688_probe(struct snd_card *card, unsigned int n) if (error < 0) return error; - strcpy(card->driver, "ES1688"); - strcpy(card->shortname, pcm->name); - sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i", pcm->name, - chip->port, chip->irq, chip->dma8); + strlcpy(card->driver, "ES1688", sizeof(card->driver)); + strlcpy(card->shortname, pcm->name, sizeof(card->shortname)); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx, irq %i, dma %i", pcm->name, chip->port, + chip->irq, chip->dma8); if (fm_port[n] == SNDRV_AUTO_PORT) fm_port[n] = port[n]; /* share the same port */ @@ -271,6 +272,8 @@ static int __devinit snd_es968_pnp_detect(struct pnp_card_link *pcard, if (enable[dev] && isapnp[dev]) break; } + if (dev == SNDRV_CARDS) + return -ENODEV; error = snd_card_create(index[dev], id[dev], THIS_MODULE, sizeof(struct snd_es1688), &card);

2 1

[alsa-devel] [PATCH 0/4] updates for Intel SandyBridge/CougarPoint HDMI codec V2
by Wu Fengguang 17 May '10

17 May '10

Hi Takashi, The Intel SandyBridge/CougarPoint HDMI codec is very similar to the IbexPeak HDMI codec, except that it exports one more Audio Converter. Now we have 3 Audio Converters, each of them present one HDMI PCM device: (and expect a 4th HDMI PCM device in future) $ aplay -l | grep HDMI card 0: PCH [HDA Intel PCH], device 3: INTEL HDMI 0 [INTEL HDMI 0] card 0: PCH [HDA Intel PCH], device 7: INTEL HDMI 1 [INTEL HDMI 1] card 0: PCH [HDA Intel PCH], device 8: INTEL HDMI 2 [INTEL HDMI 2] So enable devices minors to support more than 8 PCM playback devices. Thanks, Fengguang

2 5

[alsa-devel] [PATCH 1/20] sound/pci/asihpi: Use kzalloc
by Julia Lawall 17 May '10

17 May '10

From: Julia Lawall <julia(a)diku.dk> Use kzalloc rather than the combination of kmalloc and memset. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression x,size,flags; statement S; @@ -x = kmalloc(size,flags); +x = kzalloc(size,flags); if (x == NULL) S -memset(x, 0, size); // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/pci/asihpi/hpi6000.c | 3 +-- sound/pci/asihpi/hpi6205.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff -u -p a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -448,14 +448,13 @@ static void subsys_create_adapter(struct if (phm->u.s.resource.r.pci->device_id != HPI_PCI_DEV_ID_PCI2040) return; - ao.priv = kmalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL); + ao.priv = kzalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL); if (!ao.priv) { HPI_DEBUG_LOG(ERROR, "cant get mem for adapter object\n"); phr->error = HPI_ERROR_MEMORY_ALLOC; return; } - memset(ao.priv, 0, sizeof(struct hpi_hw_obj)); /* create the adapter object based on the resource information */ /*? memcpy(&ao.Pci,&phm->u.s.Resource.r.Pci,sizeof(ao.Pci)); */ ao.pci = *phm->u.s.resource.r.pci; diff -u -p a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -482,13 +482,12 @@ static void subsys_create_adapter(struct if (phm->u.s.resource.r.pci->device_id != HPI_PCI_DEV_ID_DSP6205) return; - ao.priv = kmalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL); + ao.priv = kzalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL); if (!ao.priv) { HPI_DEBUG_LOG(ERROR, "cant get mem for adapter object\n"); phr->error = HPI_ERROR_MEMORY_ALLOC; return; } - memset(ao.priv, 0, sizeof(struct hpi_hw_obj)); ao.pci = *phm->u.s.resource.r.pci; err = create_adapter_obj(&ao, &os_error_code);

2 1

[alsa-devel] [patch 0/6] kirkwood openrd client audio support - v2
by apatard＠mandriva.com 17 May '10

17 May '10

[ added forgotten linux-arm-kernel cc: ] Hi, It's a new try to get merged support for openrd client audio. I've tried to fix all comments, I hope I didn't miss something. In the changes, there's : - s/kirkwood/orion/ - Added a comment about data centered around 0 in the new control entry - fixed quilt diffstat header braindamage - added dapm support to cs42l51 codec - use ASoC register cache support - modified i2s driver to be a platform driver and move resource stuff to the right place Arnaud Patard

3 13

[alsa-devel] snd_pcm_period_elapsed
by Raymond Yau 16 May '10

16 May '10

>>> do snd_pcm_period_elapsed really handle the case when more than one period >>> are elasped ? >>> >>> For au88x0 , each substream have four sets of hardware registers , it seem >>> that the driver can recover lost interrupt with no underrun when using very >>> small period size >>> >>> >>> http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch05s07.html#pcm-… <http://www.alsa-project.org/%7Etiwai/writing-an-alsa-driver/ch05s07.html#pc…> >>> >>> On calling snd_pcm_period_elapsed() >>> >>> In both cases, even if more than one period are elapsed, you don't have to >>> call snd_pcm_period_elapsed() many times. Call only once. And the pcm layer >>> will check the current hardware pointer and update to the latest status. >> Yes, the new code handles more alapsed periods, too. The in_interrupt >> check is mainly for situations when period count == 1 and it compares >> expected next hw_ptr for interrupt with hw_ptr computed from the hw_base >> and actual position in the ring buffer. >> Jaroslav The value "delta" returned by vortex_adbdma_bufshift() seem to be the number of sub-buffer(periods) elapsed ( the value can be 0 , 1 , 2 or 3 ) when aplay using very small period_size (e.g. 16 ) , the driver write message in system log when delta > 1 (i.e. more than 1 period have been elapsed ) but the sound is still ok since there are two more periods in the sub-buffer But I cannot see any code in snd_pcm_period_elasped() add two period size to hw_ptr

1 0

[alsa-devel] [PATCH] hdspmixer buffer overflow
by Georg Rudolph 15 May '10

15 May '10

Hello, to fix https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5006 the attached patch for alsa-tools is proposed: Summary: Fix buffer overflow in hdspmixer A one off string buffer overflow fixed by handling it properly with string stream. Signed-off-by: Georg Rudolph <georg.rudolph(a)schwaben.de <mailto:foo@bar.com>> --- alsa-tools-1.0.23/hdspmixer/src/HDSPMixerIOMixer.h 2010-04-16 13:11:09.000000000 +0200 +++ alsa-tools-1.0.23-work/hdspmixer/src/HDSPMixerIOMixer.h 2010-05-12 22:07:49.549949993 +0200 @@ -36,6 +36,7 @@ #include "HDSPMixerStripData.h" #include "HDSPMixerMeter.h" #include "pixmaps.h" +#include <sstream> class HDSPMixerWindow; class HDSPMixerSelector; @@ -52,7 +53,7 @@ private: char **p_iomixer_xpm; int channel_num, relative_num, mixer_type; - char channel_name[6]; + std::stringstream channel_name; void update_child(Fl_Widget &widget); public: HDSPMixerStripData *data[3][3][8]; /* data[card][mode(ss/ds/qs)][preset number] */ --- alsa-tools-1.0.23/hdspmixer/src/HDSPMixerIOMixer.cxx 2010-04-16 13:11:09.000000000 +0200 +++ alsa-tools-1.0.23-work/hdspmixer/src/HDSPMixerIOMixer.cxx 2010-05-12 22:00:26.754950707 +0200 @@ -25,9 +25,9 @@ { mixer_type = type; if (type) { - sprintf(channel_name, "Out %d", ch); + channel_name << "Out " << ch; } else { - sprintf(channel_name, "In %d", ch); + channel_name << "In " << ch; } channel_num = ch; if (channel_num%2) { @@ -73,7 +73,7 @@ draw_background(); fl_color(FL_BLACK); fl_font(FL_HELVETICA, 8); - fl_draw(channel_name, x()+4, y()+225, 27, 9, FL_ALIGN_CENTER); + fl_draw(channel_name.str().c_str(), x()+4, y()+225, 27, 9, FL_ALIGN_CENTER); for (int i=children(); i--;) { Fl_Widget& o = **a++; draw_child(o); Best Regards, Georg Rudolph

2 1

[alsa-devel] HDA Intel - Activating S/PDIF output on an HP laptop (8510W).
by Huw Hawkins 15 May '10

15 May '10

Hi all, I have been quite happily using the intel HDA drivers to get audio on my Compaq 8510W, everything works fine out of the box with the exception of the HDMI output. After some poking around I see that this is because the digital output is disabled by the HP model specific code. Setting the model as "basic" gives me perfect audio on the HDMI output, but jack sensing is completely messed up, the mute button status is inverted and it is not possible to get sound from the internal speakers. Setting the model as "HP" diables the digital audio on the HDMI port. It seems from a google search that there are a few people around with the same issue on HP laptops with HDMI output. I don't really have the time to dig through and understand the spec/code for the driver to try to fix the problem myself. On the other hand I probably could spend an hour or two over the next couple of weeks testing or investigating the hardware if someone else wants to have a go. Any takers? Huw

1 0

[alsa-devel] [patch -next] sound/asihpi: incorrect range check
by Dan Carpenter 14 May '10

14 May '10

The entity_type_to_size[] array has LAST_ENTITY_TYPE (11) number of elements, not LAST_ENTITY_ROLE (17). This only affects the debug output. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 254c580..eda26b3 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -3407,7 +3407,7 @@ u16 hpi_entity_alloc_and_pack(const enum e_entity_type type, if (hE) return hE; - HPI_DEBUG_ASSERT(role > entity_role_null && type < LAST_ENTITY_ROLE); + HPI_DEBUG_ASSERT(role > entity_role_null && type < LAST_ENTITY_TYPE); bytes_to_copy = entity_type_to_size[type] * item_count; total_size = hpi_entity_header_size(*entity) + bytes_to_copy;

1 0

[alsa-devel] [PATCH] OMAP: McBSP: Add 32-bit mode support
by Sergey Lapin 14 May '10

14 May '10

This patchs should allow to use 32-bit samples on e.g. TLV320AIC3x codec, or others. Signed-off-by: Sergey Lapin <slapin(a)ossfans.org> --- sound/soc/omap/omap-mcbsp.c | 28 ++++++++++++++++++++++++---- 1 files changed, 24 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8ad9dc9..2dead3f 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -295,8 +295,18 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; - omap_mcbsp_dai_dma_params[id][substream->stream].data_type = - OMAP_DMA_DATA_TYPE_S16; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S32; + break; + default: + return -EINVAL; + } snd_soc_dai_set_dma_data(cpu_dai, substream, &omap_mcbsp_dai_dma_params[id][substream->stream]); @@ -330,6 +340,14 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16); regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16); break; + case SNDRV_PCM_FORMAT_S32_LE: + /* Set word lengths */ + wlen = 32; + regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_32); + regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_32); + regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_32); + regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_32); + break; default: /* Unsupported PCM format */ return -EINVAL; @@ -623,13 +641,15 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { .channels_min = 1, \ .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE, \ }, \ .capture = { \ .channels_min = 1, \ .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE, \ }, \ .ops = &omap_mcbsp_dai_ops, \ .private_data = &mcbsp_data[(link_id)].bus_id, \ -- 1.7.0.4

5 4

[alsa-devel] [RFC][PATCH] fall back to reserved minor numbers
by Wu Fengguang 14 May '10

14 May '10

When the static minor numbers collide because of too many devices, fall back to the reserved minor numbers 2 and 3. This for example will allow Intel HDMI audio driver to create three PCM devices (index 3, 7, 8; the minor number of the last index will otherwise collide with the first record device). The other option is to force enable CONFIG_SND_DYNAMIC_MINORS. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- include/sound/minors.h | 4 +++- sound/core/sound.c | 10 ++++++++-- 2 files changed, 11 insertions(+), 3 deletions(-) --- drm-intel.orig/include/sound/minors.h 2010-05-13 10:06:52.000000000 +0800 +++ drm-intel/include/sound/minors.h 2010-05-13 10:10:06.000000000 +0800 @@ -35,7 +35,9 @@ #define SNDRV_MINOR_TIMER (SNDRV_MINOR_GLOBAL + 1 * 32) #ifndef CONFIG_SND_DYNAMIC_MINORS - /* 2 - 3 (reserved) */ + +#define SNDRV_MINOR_BACKUP1 2 +#define SNDRV_MINOR_BACKUP2 3 #define SNDRV_MINOR_HWDEP 4 /* 4 - 7 */ #define SNDRV_MINOR_RAWMIDI 8 /* 8 - 15 */ #define SNDRV_MINOR_PCM_PLAYBACK 16 /* 16 - 23 */ --- drm-intel.orig/sound/core/sound.c 2010-05-13 10:06:04.000000000 +0800 +++ drm-intel/sound/core/sound.c 2010-05-13 10:11:19.000000000 +0800 @@ -269,8 +269,14 @@ int snd_register_device_for_dev(int type minor = snd_find_free_minor(); #else minor = snd_kernel_minor(type, card, dev); - if (minor >= 0 && snd_minors[minor]) - minor = -EBUSY; + if (minor >= 0) { + if (snd_minors[minor]) + minor = SNDRV_MINOR_BACKUP1; + if (snd_minors[minor]) + minor = SNDRV_MINOR_BACKUP2; + if (snd_minors[minor]) + minor = -EBUSY; + } #endif if (minor < 0) { mutex_unlock(&sound_mutex);

2 3

[alsa-devel] [PATCH 0/5] updates for Intel SandyBridge/CougarPoint HDMI codec
by Wu Fengguang 14 May '10

14 May '10

Hi Takashi, The Intel SandyBridge/CougarPoint HDMI codec is very similar to the IbexPeak HDMI codec, except that it exports one more Audio Converter. Now we have 3 Audio Converters, each of them present one HDMI PCM device: (and expect a 4th HDMI PCM device in future) $ aplay -l | grep HDMI card 0: PCH [HDA Intel PCH], device 3: INTEL HDMI 0 [INTEL HDMI 0] card 0: PCH [HDA Intel PCH], device 7: INTEL HDMI 1 [INTEL HDMI 1] card 0: PCH [HDA Intel PCH], device 8: INTEL HDMI 2 [INTEL HDMI 2] So ALSA has to support more than 8 PCM playback devices now. The first two patches do this dirty work, however I'm not sure if they will introduce some side effects. The last three patches are more straightforward and safe. Thanks, Fengguang

5 19

Re: [alsa-devel] [PATCH v3 0/2] McBSP changes for OMAP4 platform
by Liam Girdwood 14 May '10

14 May '10

On Thu, 2010-05-13 at 09:19 +0100, Mark Brown wrote: > On Wed, May 12, 2010 at 12:18:38PM -0500, Jorge Eduardo Candelaria wrote: > > The following patches enable McBSP driver to be used along with the > > audio driver in SDP4430 and other OMAP4 based boards. > > Both > > Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Tony, do we have you ack/sob to upstream via ASoC ? We do have a lot pending with mcbsp dependencies for ASoC atm and this would simplify upstreaming. Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

2 1

[alsa-devel] underruns over network
by Preston Kulaga 14 May '10

14 May '10

Hello, I am quite new to ALSA and am having a bit of trouble with underruns. I am working on a voice communication system via udp sockets and am getting underruns on receiving end very sporadically (capture end is fine). Have done a great deal of reading and searching on reasons for this but am not sure what approach is best to limit the underruns. I must not be transferring the data quickly enough over the network before the ring buffer runs out of data, which makes sense, but how would you accommodate for this? I am able to buffer a large amount of network data and then playback successfully, but obviously the delay induced is so significant that it makes real-time communication impossible. If I just recover/prepare the stream after an underrun the audio is horribly distorted. Is there a normal way of dealing with this issue that I am unable to find? I would appreciate any insight, guidance, references for any help you can give me. Thank You.

1 0

[alsa-devel] ALSA and Ubuntu addendum
by Andreas Falkenhahn 13 May '10

13 May '10

Addendum: I forgot one thing that doesn't work on Ubuntu either (but works on openSUSE): PCM streams in signed 8 bit format have terrible sound artefacts on Ubuntu. If I convert them to unsigned 8 bit format, they are played correctly. But in signed format, it sounds totally broken.

1 0

[alsa-devel] ALSA and Ubuntu
by Andreas Falkenhahn 13 May '10

13 May '10

Hi, I just downloaded the latest Ubuntu Live CD (10.04) to try my app on it. It is a plain X11 app using ALSA for sound output. I developed it on openSUSE 11.2 and thought to give it a try on Ubuntu... that was a mistake, because on Ubuntu, there seem to be lots of errors with ALSA. Here are some observations: 1) snd_pcm_status_get_tstamp() always returns 0 for sec/usec. 2) snd_pcm_status_get_state() always returns SND_PCM_STATE_RUNNING, even when the PCM stream has long reached a buffer underrun. 3) snd_pcm_delay()'s return value doesn't fall below ~7000 samples or so. Even when the buffer has underrun, the return value is not 0 but about ~7000 samples. This makes it nearly impossible to detect when I should start writing new samples to the buffer... Could someone enlighten me what's going on there? Is this a known issue with Ubuntu? I'm not using any weird addons or stuff. It's really just the latest Ubuntu live CD. As I said, on openSUSE everything is working just fine. But on Ubuntu, hell is really breaking loose here... Thanks for help, Andreas

1 0

[alsa-devel] [PATCH] OMAP: McBSP: Add 32-bit mode support
by Sergey Lapin 13 May '10

13 May '10

This patchs should allow to use 32-bit samples on e.g. TLV320AIC3x codec, or others. Signed-off-by: Sergey Lapin <slapin(a)ossfans.org> --- sound/soc/omap/omap-mcbsp.c | 30 ++++++++++++++++++++++++++---- 1 files changed, 26 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8ad9dc9..a875d5d 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -295,8 +295,20 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; - omap_mcbsp_dai_dma_params[id][substream->stream].data_type = - OMAP_DMA_DATA_TYPE_S16; + switch(params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S32; + break; + default: + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S16; + break; + } snd_soc_dai_set_dma_data(cpu_dai, substream, &omap_mcbsp_dai_dma_params[id][substream->stream]); @@ -330,6 +342,14 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16); regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16); break; + case SNDRV_PCM_FORMAT_S32_LE: + /* Set word lengths */ + wlen = 32; + regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_32); + regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_32); + regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_32); + regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_32); + break; default: /* Unsupported PCM format */ return -EINVAL; @@ -623,13 +643,15 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { .channels_min = 1, \ .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE, \ }, \ .capture = { \ .channels_min = 1, \ .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE, \ }, \ .ops = &omap_mcbsp_dai_ops, \ .private_data = &mcbsp_data[(link_id)].bus_id, \ -- 1.7.0.4

2 2

[alsa-devel] broken microphone
by mutenmaster＠tiscali.it 13 May '10

13 May '10

hello everyone, mi microphone is broken. the os version is the latest, but this bug exists from, 2.6.33 or something. i mailed a message describing the bisection i did back then to vger. Here it is again! $ sudo git bisect bad f5271101faf1655d862849f42518c2a88ef394fb is the first bad commit commit f5271101faf1655d862849f42518c2a88ef394fb Author: Lydia Wang <lydiawang(a)viatech.com.cn> Date: Sat Oct 10 19:07: 35 2009 +0800 ALSA HDA VIA: Add VIA_CTL_WIDGET_ANALOG_MUTE control type Enter low power state if AA-Path volume is muted. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Signed-off-by: Logan Li <loganli(a)viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 ab1ae14ff67f0b08848faeb3ad68def7f3e727f0 0bfbfde11fe32e87e8edbdb3c4d03c2d30370393 M sound after reverting this, everything works fine. i already posted a message with all of my system info attached, but it got moderated. If you need any info, please ask. bye, vasilis. Risparmia con Tutto Incluso Light: telefono + adsl 8 mega a soli 14,95 € al mese. Gratis la Sim Tiscali Mobile con 25 euro di traffico! L'offerta è valida solo se attivi entro il 13/05/10 http://abbonati.tiscali.it/telefono-adsl/prodotti/tc/tuttoincluso_light/?WT…

1 0

[alsa-devel] HP dv series mute LED problems?
by Kunal Gangakhedkar 13 May '10

13 May '10

Hi Takashi, The commit 26ebe0a28986f4845b2c5bea43ac5cc0b9f27f0a seems to undo the mute LED gpio fixes - at least on my dv6 series laptop. Maybe, we need to come up with a better solution to accommodate the problem reported in this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-May/027352.html I believe, that's the reason for the commit, right? Let me do a bit more digging. Thanks, Kunal

3 8

[alsa-devel] Use of start_frame in usbusx2yaudio.c
by Sarah Sharp 12 May '10

12 May '10

Are there any drivers in the kernel that set urb->start_frame on every URB? Could those drivers handle it if only the first URB they submitted to the host controller was scheduled for that frame ID, and all the rest of the URBs were scheduled ASAP? I see there are three drivers that set start_frame (while not setting URB_ISO_ASAP): - drivers/isdn/hisax/st5481_d.c - drivers/usb/core/devio.c - sound/usb/usx2y/usbusx2yaudio.c I'm not really sure what usbusx2yaudio.c is doing. I think when one URB completes, it sets the next URB's start_frame to the previous URB's start_frame plus the number of URBs (2 by default) times the number of packets (4 by default). Isn't this basically like setting URB_ISO_ASAP? I really can't tell what fall back method is if this submission fails. What is usbusx2yaudio.c attempting to do? I've tried to get an overall picture of what it expects the isochronous scheduling to look like, but I'm finding the driver a bit hard to read. I'm not too worried about the other two drivers. The code in st5481_d.c seems to fall back fine to URB_ISO_ASAP if the submission with the start_frame set failed. devio.c is usbfs, and I can't tell what userspace drivers do. (It could be argued that if they needed such tight control over isoc, they should have written a kernel driver.) Sarah Sharp

3 3

[alsa-devel] ESI Maya44 PCI – mixer code errors
by Alex Chernyshoff 12 May '10

12 May '10

Hello! In the current release of ESI Maya44 PCI ALSA driver (1.0.23) selection of “Mic In” capture source is impossible. “Microphone” check in the “Switches “ mixer tab actually does not select anything – no recording audio at all, and the Phantom power switch does not perform correctly too. Original, Rainer Zimmermann patch (for ALSA 1.0.19) performs well, but can`t be compiled with the current ALSA and kernel26. Sadly... I've look into current, rewritten, code in the file ./alsa-driver-1.0.23/alsa-kernel/pci/ice1712/maya44.c and found some interesting details. Further, I've add some changes, and “Mic In” selection now works, but only partially. (Below) In the original maya44.c file, from Rainer Zimmermann patch, some definitions of I/O flags seems as a simple mask: #define GPIO_PHANTOM_OFF (1<<2) #define GPIO_MIC_RELAY (1<<4) #define GPIO_SPDIF_IN_INV (1<<5) /* CHECK! */ #define GPIO_MUST_BE_0 (1<<7) Current branch of Maya44 code is rewritten, and these definitions look like the indexes #define GPIO_PHANTOM_OFF 2 #define GPIO_MIC_RELAY 4 #define GPIO_SPDIF_IN_INV 5 #define GPIO_MUST_BE_0 7 So, at the row 396 of the current branch - maya44.c changed = maya_set_gpio_bits(chip->ice, GPIO_MIC_RELAY, sel ? GPIO_MIC_RELAY : 0); I change this code to: changed = maya_set_gpio_bits(chip->ice, (1 << GPIO_MIC_RELAY), sel ? (1 << GPIO_MIC_RELAY) : 0); And finally, at the row 348 of the current maya44.c /* known working input slots (0-4) */ #define MAYA_LINE_IN 1 /* in-2 */ #define MAYA_MIC_IN 4 /* in-5 */ but usable channels are 2 and 4 (from datasheet), therefore I change defines to #define MAYA_LINE_IN 1 /* in-2 */ #define MAYA_MIC_IN 3 /* in-4 */ After that, we can get some usability from mixer. Now, the “Microphone” check can select “Mic In”, and real audio path from microphone to ADC works! But only once. After that, in the “Capture” tab of mixer, “Mute” knob visually change state to muted. Some kind of strange cross influence between mixer tabs :-) Pressing to this “Mute” knob switches back “Mic In” capture source to “Line”, and clear Phantom Power check. (Very strange) And finally, if we check “Microphone” again - “Mic In” became selected again, and “Mute” enter in “unpressed” state again too. But microphone capture became again operable. Circle is closed. Excuse me for some kind of messed story and poor English (my native is Russian), but such description of mixer behavior maybe useful for understanding... I can't resolve more mistakes in the code. Perhaps, author can do it more better than me. I have a hardware – ESI Maya44 PCI audio board installed to working and usable Arch Linux i686 system. I'm ready for testing, and glad to do it! Sincerely Alex Chernyshoff

2 1

[alsa-devel] [patch 0/6] kirkwood openrd client audio support
by apatard＠mandriva.com 12 May '10

12 May '10

This patchset aims at adding ASoC support to ARM kirkwood systems and more precisely to openrd client boards. Arnaud Patard

7 22

[alsa-devel] [PATCH] ASoC: TWL4030: Add control for digimic Left Right swap
by Peter Ujfalusi 12 May '10

12 May '10

The codec has support for swapping the left and right channels in the digimic interface. New kcontrol to handle this bit. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 0fe74d1..6a34f56 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1102,6 +1102,16 @@ static const struct soc_enum twl4030_vibradir_enum = ARRAY_SIZE(twl4030_vibradir_texts), twl4030_vibradir_texts); +/* Digimic Left and right swapping */ +static const char *twl4030_digimicswap_texts[] = { + "Not swapped", "Swapped", +}; + +static const struct soc_enum twl4030_digimicswap_enum = + SOC_ENUM_SINGLE(TWL4030_REG_MISC_SET_1, 0, + ARRAY_SIZE(twl4030_digimicswap_texts), + twl4030_digimicswap_texts); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Codec operation mode control */ SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum, @@ -1178,6 +1188,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), SOC_ENUM("Vibra H-bridge direction", twl4030_vibradir_enum), + + SOC_ENUM("Digimic LR Swap", twl4030_digimicswap_enum), }; static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { -- 1.7.1

3 2

[alsa-devel] alsa-driver patches
by Eliot Blennerhassett 12 May '10

12 May '10

Hi Takashi, since the acceptance of asihpi driver, and move to alsa-kernel, you created a couple of patch files in alsa-driver/pci/asihpi (that add conditional compilation for older kernels) Can you tell me how these patches are generated and maintained? (I guess this applies to any driver with this kind of patch). I think that this arrangement means that my one upstream file needs to be converted to two files (kernel .c + patch) when I submit updates? thanks -- Eliot Blennerhassett AudioScience Inc.

2 1

[alsa-devel] [PATCH 1/2] ALSA: include/linux/usb/audio.h: add __attribute__((packed))
by Daniel Mack 12 May '10

12 May '10

This was missing on the definition of struct uac_iso_endpoint_descriptor Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Clemens Ladisch <clemens(a)ladisch.de> --- include/linux/usb/audio.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index 57f2055..9fae6bd 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -456,7 +456,7 @@ struct uac_iso_endpoint_descriptor { __u8 bmAttributes; __u8 bLockDelayUnits; __le16 wLockDelay; -}; +} __attribute__((packed)); #define UAC_ISO_ENDPOINT_DESC_SIZE 7 #define UAC_EP_CS_ATTR_SAMPLE_RATE 0x01 -- 1.7.1

3 5

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

11 May '10

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

2 1

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

11 May '10

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

2 8

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

11 May '10

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

3 5

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

11 May '10

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

2 1

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

11 May '10

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

1 0

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

11 May '10

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

2 8

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

11 May '10

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

3 2

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

11 May '10

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

2 1

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

11 May '10

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

2 3

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

11 May '10

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

2 1

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

11 May '10

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

3 6

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

11 May '10

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

2 3

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

10 May '10

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

2 3

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

10 May '10

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

1 0

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

10 May '10

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

1 0

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

10 May '10

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

2 3

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

10 May '10

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

2 1

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

10 May '10

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

1 0

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

10 May '10

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

3 4

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

10 May '10

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

1 0

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

10 May '10

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

2 1

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

10 May '10

A specialised HID driver for the Creative Prodikeys PC-MIDI USB Keyboard. The Prodikeys PC-MIDI is a multifunction keyboard comprising a qwerty keyboard, multimedia keys and a touch sensitive musical keyboard. The specialised HID driver adds full support for the musical keyboard and extra multimedia keys which are not currently handled by the default HID driver. The specialised HID driver interfaces with Alsa, and presents the midi keyboard as a Rawmidi device. Sustain duration, octave shifting and the midi output channel can be read/written form userspace via sysfs. Signed-off-by: Don Prince <dhprince-devel(a)yahoo.co.uk> --- Applied fixes to original patch submission suggested by Clemens Ladisch. The keyboard can be viewed here http://www.prodikeys.com/products/Prodikeys_PC-MIDI/ The driver (older version) and a simple wxWidgets based UI application is available (in tar.gz format) at the Sourceforge project page http://sourceforge.net/projects/pc-midi-linux/ Apply patch with patch -p1 < <path to the driver patch> --- diff -X /opt/scripts/diff-exclude.txt -Npaur a/Documentation/ABI/testing/sysfs-driver-hid-prodikeys b/Documentation/ABI/testing/sysfs-driver-hid-prodikeys --- a/Documentation/ABI/testing/sysfs-driver-hid-prodikeys 1970-01-01 01:00:00.000000000 +0100 +++ b/Documentation/ABI/testing/sysfs-driver-hid-prodikeys 2010-05-09 16:14:33.948863321 +0100 @@ -0,0 +1,29 @@ +What: /sys/bus/hid/drivers/prodikeys/.../channel +Date: April 2010 +KernelVersion: 2.6.34 +Contact: Don Prince <dhprince.devel(a)yahoo.co.uk> +Description: + Allows control (via software) the midi channel to which + that the pc-midi keyboard will output.midi data. + Range: 0..15 + Type: Read/write +What: /sys/bus/hid/drivers/prodikeys/.../sustain +Date: April 2010 +KernelVersion: 2.6.34 +Contact: Don Prince <dhprince.devel(a)yahoo.co.uk> +Description: + Allows control (via software) the sustain duration of a + note held by the pc-midi driver. + 0 means sustain mode is disabled. + Range: 0..5000 (milliseconds) + Type: Read/write +What: /sys/bus/hid/drivers/prodikeys/.../octave +Date: April 2010 +KernelVersion: 2.6.34 +Contact: Don Prince <dhprince.devel(a)yahoo.co.uk> +Description: + Controls the octave shift modifier in the pc-midi driver. + The octave can be shifted via software up/down 2 octaves. + 0 means the no ocatve shift. + Range: -2..2 (minus 2 to plus 2) + Type: Read/Write diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/hid-core.c b/drivers/hid/hid-core.c --- a/drivers/hid/hid-core.c 2010-04-22 18:52:46.445384754 +0100 +++ b/drivers/hid/hid-core.c 2010-04-25 19:10:31.317248975 +0100 @@ -1297,6 +1297,7 @@ static const struct hid_device_id hid_bl { HID_USB_DEVICE(USB_VENDOR_ID_BELKIN, USB_DEVICE_ID_FLIP_KVM) }, { HID_USB_DEVICE(USB_VENDOR_ID_CHERRY, USB_DEVICE_ID_CHERRY_CYMOTION) }, { HID_USB_DEVICE(USB_VENDOR_ID_CHICONY, USB_DEVICE_ID_CHICONY_TACTICAL_PAD) }, + { HID_USB_DEVICE(USB_VENDOR_ID_CREATIVELABS, USB_DEVICE_ID_PRODIKEYS_PCMIDI) }, { HID_USB_DEVICE(USB_VENDOR_ID_CYPRESS, USB_DEVICE_ID_CYPRESS_BARCODE_1) }, { HID_USB_DEVICE(USB_VENDOR_ID_CYPRESS, USB_DEVICE_ID_CYPRESS_BARCODE_2) }, { HID_USB_DEVICE(USB_VENDOR_ID_CYPRESS, USB_DEVICE_ID_CYPRESS_BARCODE_3) }, diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/hid-ids.h b/drivers/hid/hid-ids.h --- a/drivers/hid/hid-ids.h 2010-04-22 18:52:46.455385039 +0100 +++ b/drivers/hid/hid-ids.h 2010-04-25 19:10:31.319248829 +0100 @@ -147,6 +147,9 @@ #define USB_DEVICE_ID_CODEMERCS_IOW_FIRST 0x1500 #define USB_DEVICE_ID_CODEMERCS_IOW_LAST 0x15ff +#define USB_VENDOR_ID_CREATIVELABS 0x041e +#define USB_DEVICE_ID_PRODIKEYS_PCMIDI 0x2801 + #define USB_VENDOR_ID_CYGNAL 0x10c4 #define USB_DEVICE_ID_CYGNAL_RADIO_SI470X 0x818a diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/hid-prodikeys.c b/drivers/hid/hid-prodikeys.c --- a/drivers/hid/hid-prodikeys.c 1970-01-01 01:00:00.000000000 +0100 +++ b/drivers/hid/hid-prodikeys.c 2010-05-09 16:20:42.569736755 +0100 @@ -0,0 +1,910 @@ +/* + * HID driver for the Prodikeys PC-MIDI Keyboard + * providing midi & extra multimedia keys functionality + * + * Copyright (c) 2009 Don Prince <dhprince.devel(a)yahoo.co.uk> + * + * Controls for Octave Shift Up/Down, Channel, and + * Sustain Duration available via sysfs. + * + */ + +/* + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + */ + +#include <linux/device.h> +#include <linux/module.h> +#include <linux/usb.h> +#include <linux/mutex.h> +#include <linux/hid.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> +#include "usbhid/usbhid.h" +#include "hid-ids.h" + + +#define pk_debug(format, arg...) \ + pr_debug("hid-prodikeys: " format "\n" , ## arg) +#define pk_error(format, arg...) \ + pr_err("hid-prodikeys: " format "\n" , ## arg) + +struct pcmidi_snd; + +struct pk_device { + unsigned long quirks; + + struct hid_device *hdev; + struct pcmidi_snd *pm; /* pcmidi device context */ +}; + +struct pcmidi_snd; + +struct pcmidi_sustain { + unsigned long in_use; + struct pcmidi_snd *pm; + struct timer_list timer; + unsigned char status; + unsigned char note; + unsigned char velocity; +}; + +#define PCMIDI_SUSTAINED_MAX 32 +struct pcmidi_snd { + struct pk_device *pk; + unsigned short ifnum; + struct hid_report *pcmidi_report6; + struct input_dev *input_ep82; + unsigned short midi_mode; + unsigned short midi_sustain_mode; + unsigned short midi_sustain; + unsigned short midi_channel; + short midi_octave; + struct pcmidi_sustain sustained_notes[PCMIDI_SUSTAINED_MAX]; + unsigned short fn_state; + unsigned short last_key[24]; + spinlock_t rawmidi_in_lock; + struct snd_card *card; + struct snd_rawmidi *rwmidi; + struct snd_rawmidi_substream *in_substream; + struct snd_rawmidi_substream *out_substream; + unsigned long in_triggered; + unsigned long out_active; +}; + +#define PK_QUIRK_NOGET 0x00010000 +#define PCMIDI_MIDDLE_C 60 +#define PCMIDI_CHANNEL_MIN 0 +#define PCMIDI_CHANNEL_MAX 15 +#define PCMIDI_OCTAVE_MIN (-2) +#define PCMIDI_OCTAVE_MAX 2 +#define PCMIDI_SUSTAIN_MIN 0 +#define PCMIDI_SUSTAIN_MAX 5000 + +static const char shortname[] = "PC-MIDI"; +static const char longname[] = "Prodikeys PC-MIDI Keyboard"; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +module_param_array(id, charp, NULL, 0444); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for the PC-MIDI virtual audio driver"); +MODULE_PARM_DESC(id, "ID string for the PC-MIDI virtual audio driver"); +MODULE_PARM_DESC(enable, "Enable for the PC-MIDI virtual audio driver"); + + +/* Output routine for the sysfs channel file */ +static ssize_t show_channel(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + dbg_hid("pcmidi sysfs read channel=%u\n", pk->pm->midi_channel); + + return sprintf(buf, "%u (min:%u, max:%u)\n", pk->pm->midi_channel, + PCMIDI_CHANNEL_MIN, PCMIDI_CHANNEL_MAX); +} + +/* Input routine for the sysfs channel file */ +static ssize_t store_channel(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + unsigned channel = 0; + + if (sscanf(buf, "%u", &channel) > 0 && channel <= PCMIDI_CHANNEL_MAX) { + dbg_hid("pcmidi sysfs write channel=%u\n", channel); + pk->pm->midi_channel = channel; + return strlen(buf); + } + return -EINVAL; +} + +static DEVICE_ATTR(channel, S_IRUGO | S_IWUGO, show_channel, + store_channel); + +static struct device_attribute *sysfs_device_attr_channel = { + &dev_attr_channel, + }; + +/* Output routine for the sysfs sustain file */ +static ssize_t show_sustain(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + dbg_hid("pcmidi sysfs read sustain=%u\n", pk->pm->midi_sustain); + + return sprintf(buf, "%u (off:%u, max:%u (ms))\n", pk->pm->midi_sustain, + PCMIDI_SUSTAIN_MIN, PCMIDI_SUSTAIN_MAX); +} + +/* Input routine for the sysfs sustain file */ +static ssize_t store_sustain(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + unsigned sustain = 0; + + if (sscanf(buf, "%u", &sustain) > 0 && sustain <= PCMIDI_SUSTAIN_MAX) { + dbg_hid("pcmidi sysfs write sustain=%u\n", sustain); + pk->pm->midi_sustain = sustain; + pk->pm->midi_sustain_mode = + (0 == sustain || !pk->pm->midi_mode) ? 0 : 1; + return strlen(buf); + } + return -EINVAL; +} + +static DEVICE_ATTR(sustain, S_IRUGO | S_IWUGO, show_sustain, + store_sustain); + +static struct device_attribute *sysfs_device_attr_sustain = { + &dev_attr_sustain, + }; + +/* Output routine for the sysfs octave file */ +static ssize_t show_octave(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + dbg_hid("pcmidi sysfs read octave=%d\n", pk->pm->midi_octave); + + return sprintf(buf, "%d (min:%d, max:%d)\n", pk->pm->midi_octave, + PCMIDI_OCTAVE_MIN, PCMIDI_OCTAVE_MAX); +} + +/* Input routine for the sysfs octave file */ +static ssize_t store_octave(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + int octave = 0; + + if (sscanf(buf, "%d", &octave) > 0 && + octave >= PCMIDI_OCTAVE_MIN && octave <= PCMIDI_OCTAVE_MAX) { + dbg_hid("pcmidi sysfs write octave=%d\n", octave); + pk->pm->midi_octave = octave; + return strlen(buf); + } + return -EINVAL; +} + +static DEVICE_ATTR(octave, S_IRUGO | S_IWUGO, show_octave, + store_octave); + +static struct device_attribute *sysfs_device_attr_octave = { + &dev_attr_octave, + }; + + +static void pcmidi_send_note(struct pcmidi_snd *pm, + unsigned char status, unsigned char note, unsigned char velocity) +{ + unsigned long flags; + unsigned char buffer[3]; + + buffer[0] = status; + buffer[1] = note; + buffer[2] = velocity; + + spin_lock_irqsave(&pm->rawmidi_in_lock, flags); + + if (!pm->in_substream) + goto drop_note; + if (!test_bit(pm->in_substream->number, &pm->in_triggered)) + goto drop_note; + + snd_rawmidi_receive(pm->in_substream, buffer, 3); + +drop_note: + spin_unlock_irqrestore(&pm->rawmidi_in_lock, flags); + + return; +} + +void pcmidi_sustained_note_release(unsigned long data) +{ + struct pcmidi_sustain *pms = (struct pcmidi_sustain *)data; + + pcmidi_send_note(pms->pm, pms->status, pms->note, pms->velocity); + pms->in_use = 0; +} + +void init_sustain_timers(struct pcmidi_snd *pm) +{ + struct pcmidi_sustain *pms; + unsigned i; + + for (i = 0; i < PCMIDI_SUSTAINED_MAX; i++) { + pms = &pm->sustained_notes[i]; + pms->in_use = 0; + pms->pm = pm; + setup_timer(&pms->timer, pcmidi_sustained_note_release, + (unsigned long)pms); + } +} + +void stop_sustain_timers(struct pcmidi_snd *pm) +{ + struct pcmidi_sustain *pms; + unsigned i; + + for (i = 0; i < PCMIDI_SUSTAINED_MAX; i++) { + pms = &pm->sustained_notes[i]; + pms->in_use = 1; + del_timer_sync(&pms->timer); + } +} + +static int pcmidi_get_output_report(struct pcmidi_snd *pm) +{ + struct hid_device *hdev = pm->pk->hdev; + struct hid_report *report; + + list_for_each_entry(report, + &hdev->report_enum[HID_OUTPUT_REPORT].report_list, list) { + if (!(6 == report->id)) + continue; + + if (report->maxfield < 1) { + dev_err(&hdev->dev, "output report is empty\n"); + break; + } + if (report->field[0]->report_count != 2) { + dev_err(&hdev->dev, "field count too low\n"); + break; + } + pm->pcmidi_report6 = report; + return 0; + } + /* should never get here */ + return -ENODEV; +} + +static void pcmidi_submit_output_report(struct pcmidi_snd *pm, int state) +{ + struct hid_device *hdev = pm->pk->hdev; + struct hid_report *report = pm->pcmidi_report6; + report->field[0]->value[0] = 0x01; + report->field[0]->value[1] = state; + + usbhid_submit_report(hdev, report, USB_DIR_OUT); +} + +static int pcmidi_handle_report1(struct pcmidi_snd *pm, u8 *data) +{ + u32 bit_mask; + + bit_mask = data[1]; + bit_mask = (bit_mask << 8) | data[2]; + bit_mask = (bit_mask << 8) | data[3]; + + dbg_hid("pcmidi mode: %d\n", pm->midi_mode); + + /*KEY_MAIL or octave down*/ + if (pm->midi_mode && bit_mask == 0x004000) { + /* octave down */ + pm->midi_octave--; + if (pm->midi_octave < -2) + pm->midi_octave = -2; + dbg_hid("pcmidi mode: %d octave: %d\n", + pm->midi_mode, pm->midi_octave); + return 1; + } + /*KEY_WWW or sustain*/ + else if (pm->midi_mode && bit_mask == 0x000004) { + /* sustain on/off*/ + pm->midi_sustain_mode ^= 0x1; + return 1; + } + + return 0; /* continue key processing */ +} + +static int pcmidi_handle_report3(struct pcmidi_snd *pm, u8 *data, int size) +{ + struct pcmidi_sustain *pms; + unsigned i, j; + unsigned char status, note, velocity; + + unsigned num_notes = (size-1)/2; + for (j = 0; j < num_notes; j++) { + note = data[j*2+1]; + velocity = data[j*2+2]; + + if (note < 0x81) { /* note on */ + status = 128 + 16 + pm->midi_channel; /* 1001nnnn */ + note = note - 0x54 + PCMIDI_MIDDLE_C + + (pm->midi_octave * 12); + if (0 == velocity) + velocity = 1; /* force note on */ + } else { /* note off */ + status = 128 + pm->midi_channel; /* 1000nnnn */ + note = note - 0x94 + PCMIDI_MIDDLE_C + + (pm->midi_octave*12); + + if (pm->midi_sustain_mode) { + for (i = 0; i < PCMIDI_SUSTAINED_MAX; i++) { + pms = &pm->sustained_notes[i]; + if (!pms->in_use) { + pms->status = status; + pms->note = note; + pms->velocity = velocity; + pms->in_use = 1; + + mod_timer(&pms->timer, + jiffies + + msecs_to_jiffies(pm->midi_sustain)); + return 1; + } + } + } + } + pcmidi_send_note(pm, status, note, velocity); + } + + return 1; +} + +static int pcmidi_handle_report4(struct pcmidi_snd *pm, u8 *data) +{ + unsigned key; + u32 bit_mask; + u32 bit_index; + + bit_mask = data[1]; + bit_mask = (bit_mask << 8) | data[2]; + bit_mask = (bit_mask << 8) | data[3]; + + /* break keys */ + for (bit_index = 0; bit_index < 24; bit_index++) { + key = pm->last_key[bit_index]; + if (!((0x01 << bit_index) & bit_mask)) { + input_event(pm->input_ep82, EV_KEY, + pm->last_key[bit_index], 0); + pm->last_key[bit_index] = 0; + } + } + + /* make keys */ + for (bit_index = 0; bit_index < 24; bit_index++) { + key = 0; + switch ((0x01 << bit_index) & bit_mask) { + case 0x000010: /* Fn lock*/ + pm->fn_state ^= 0x000010; + if (pm->fn_state) + pcmidi_submit_output_report(pm, 0xc5); + else + pcmidi_submit_output_report(pm, 0xc6); + continue; + case 0x020000: /* midi launcher..send a key (qwerty) or not? */ + pcmidi_submit_output_report(pm, 0xc1); + pm->midi_mode ^= 0x01; + + dbg_hid("pcmidi mode: %d\n", pm->midi_mode); + continue; + case 0x100000: /* KEY_MESSENGER or octave up */ + dbg_hid("pcmidi mode: %d\n", pm->midi_mode); + if (pm->midi_mode) { + pm->midi_octave++; + if (pm->midi_octave > 2) + pm->midi_octave = 2; + dbg_hid("pcmidi mode: %d octave: %d\n", + pm->midi_mode, pm->midi_octave); + continue; + } else + key = KEY_MESSENGER; + break; + case 0x400000: + key = KEY_CALENDAR; + break; + case 0x080000: + key = KEY_ADDRESSBOOK; + break; + case 0x040000: + key = KEY_DOCUMENTS; + break; + case 0x800000: + key = KEY_WORDPROCESSOR; + break; + case 0x200000: + key = KEY_SPREADSHEET; + break; + case 0x010000: + key = KEY_COFFEE; + break; + case 0x000100: + key = KEY_HELP; + break; + case 0x000200: + key = KEY_SEND; + break; + case 0x000400: + key = KEY_REPLY; + break; + case 0x000800: + key = KEY_FORWARDMAIL; + break; + case 0x001000: + key = KEY_NEW; + break; + case 0x002000: + key = KEY_OPEN; + break; + case 0x004000: + key = KEY_CLOSE; + break; + case 0x008000: + key = KEY_SAVE; + break; + case 0x000001: + key = KEY_UNDO; + break; + case 0x000002: + key = KEY_REDO; + break; + case 0x000004: + key = KEY_SPELLCHECK; + break; + case 0x000008: + key = KEY_PRINT; + break; + } + if (key) { + input_event(pm->input_ep82, EV_KEY, key, 1); + pm->last_key[bit_index] = key; + } + } + + return 1; +} + +int pcmidi_handle_report( + struct pcmidi_snd *pm, unsigned report_id, u8 *data, int size) +{ + int ret = 0; + + switch (report_id) { + case 0x01: /* midi keys (qwerty)*/ + ret = pcmidi_handle_report1(pm, data); + break; + case 0x03: /* midi keyboard (musical)*/ + ret = pcmidi_handle_report3(pm, data, size); + break; + case 0x04: /* multimedia/midi keys (qwerty)*/ + ret = pcmidi_handle_report4(pm, data); + break; + } + return ret; +} + +void pcmidi_setup_extra_keys(struct pcmidi_snd *pm, struct input_dev *input) +{ + /* reassigned functionality for N/A keys + MY PICTURES => KEY_WORDPROCESSOR + MY MUSIC=> KEY_SPREADSHEET + */ + unsigned int keys[] = { + KEY_FN, + KEY_MESSENGER, KEY_CALENDAR, + KEY_ADDRESSBOOK, KEY_DOCUMENTS, + KEY_WORDPROCESSOR, + KEY_SPREADSHEET, + KEY_COFFEE, + KEY_HELP, KEY_SEND, + KEY_REPLY, KEY_FORWARDMAIL, + KEY_NEW, KEY_OPEN, + KEY_CLOSE, KEY_SAVE, + KEY_UNDO, KEY_REDO, + KEY_SPELLCHECK, KEY_PRINT, + 0 + }; + + unsigned int *pkeys = &keys[0]; + unsigned short i; + + if (pm->ifnum != 1) /* only set up ONCE for interace 1 */ + return; + + pm->input_ep82 = input; + + for (i = 0; i < 24; i++) + pm->last_key[i] = 0; + + while (*pkeys != 0) { + set_bit(*pkeys, pm->input_ep82->keybit); + ++pkeys; + } +} + +static int pcmidi_set_operational(struct pcmidi_snd *pm) +{ + if (pm->ifnum != 1) + return 0; /* only set up ONCE for interace 1 */ + + pcmidi_get_output_report(pm); + pcmidi_submit_output_report(pm, 0xc1); + return 0; +} + +static int pcmidi_snd_free(struct snd_device *dev) +{ + return 0; +} + +static int pcmidi_in_open(struct snd_rawmidi_substream *substream) +{ + struct pcmidi_snd *pm = substream->rmidi->private_data; + + dbg_hid("pcmidi in open\n"); + pm->in_substream = substream; + return 0; +} + +static int pcmidi_in_close(struct snd_rawmidi_substream *substream) +{ + dbg_hid("pcmidi in close\n"); + return 0; +} + +static void pcmidi_in_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct pcmidi_snd *pm = substream->rmidi->private_data; + + dbg_hid("pcmidi in trigger %d\n", up); + + pm->in_triggered = up; +} + +static struct snd_rawmidi_ops pcmidi_in_ops = { + .open = pcmidi_in_open, + .close = pcmidi_in_close, + .trigger = pcmidi_in_trigger +}; + +int pcmidi_snd_initialise(struct pcmidi_snd *pm) +{ + static int dev; + struct snd_card *card; + struct snd_rawmidi *rwmidi; + int err; + + static struct snd_device_ops ops = { + .dev_free = pcmidi_snd_free, + }; + + if (pm->ifnum != 1) + return 0; /* only set up midi device ONCE for interace 1 */ + + if (dev >= SNDRV_CARDS) + return -ENODEV; + + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* Setup sound card */ + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) { + pk_error("failed to create pc-midi sound card\n"); + err = -ENOMEM; + goto fail; + } + pm->card = card; + + /* Setup sound device */ + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, pm, &ops); + if (err < 0) { + pk_error("failed to create pc-midi sound device: error %d\n", + err); + goto fail; + } + + strncpy(card->driver, shortname, sizeof(card->driver)); + strncpy(card->shortname, shortname, sizeof(card->shortname)); + strncpy(card->longname, longname, sizeof(card->longname)); + + /* Set up rawmidi */ + err = snd_rawmidi_new(card, card->shortname, 0, + 0, 1, &rwmidi); + if (err < 0) { + pk_error("failed to create pc-midi rawmidi device: error %d\n", + err); + goto fail; + } + pm->rwmidi = rwmidi; + strncpy(rwmidi->name, card->shortname, sizeof(rwmidi->name)); + rwmidi->info_flags = SNDRV_RAWMIDI_INFO_INPUT; + rwmidi->private_data = pm; + + snd_rawmidi_set_ops(rwmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &pcmidi_in_ops); + + snd_card_set_dev(card, &pm->pk->hdev->dev); + + /* create sysfs variables */ + err = device_create_file(&pm->pk->hdev->dev, + sysfs_device_attr_channel); + if (err < 0) { + pk_error("failed to create sysfs attribute channel: error %d\n", + err); + goto fail; + } + + err = device_create_file(&pm->pk->hdev->dev, + sysfs_device_attr_sustain); + if (err < 0) { + pk_error("failed to create sysfs attribute sustain: error %d\n", + err); + goto fail_attr_sustain; + } + + err = device_create_file(&pm->pk->hdev->dev, + sysfs_device_attr_octave); + if (err < 0) { + pk_error("failed to create sysfs attribute octave: error %d\n", + err); + goto fail_attr_octave; + } + + spin_lock_init(&pm->rawmidi_in_lock); + + init_sustain_timers(pm); + pcmidi_set_operational(pm); + + /* register it */ + err = snd_card_register(card); + if (err < 0) { + pk_error("failed to register pc-midi sound card: error %d\n", + err); + goto fail_register; + } + + dbg_hid("pcmidi_snd_initialise finished ok\n"); + return 0; + +fail_register: + stop_sustain_timers(pm); + device_remove_file(&pm->pk->hdev->dev, sysfs_device_attr_octave); +fail_attr_octave: + device_remove_file(&pm->pk->hdev->dev, sysfs_device_attr_sustain); +fail_attr_sustain: + device_remove_file(&pm->pk->hdev->dev, sysfs_device_attr_channel); +fail: + if (pm->card) { + snd_card_free(pm->card); + pm->card = NULL; + } + return err; +} + +int pcmidi_snd_terminate(struct pcmidi_snd *pm) +{ + if (pm->card) { + stop_sustain_timers(pm); + + device_remove_file(&pm->pk->hdev->dev, + sysfs_device_attr_channel); + device_remove_file(&pm->pk->hdev->dev, + sysfs_device_attr_sustain); + device_remove_file(&pm->pk->hdev->dev, + sysfs_device_attr_octave); + + snd_card_disconnect(pm->card); + snd_card_free_when_closed(pm->card); + } + + return 0; +} + +/* + * PC-MIDI report descriptor for report id is wrong. + */ +static void pk_report_fixup(struct hid_device *hdev, __u8 *rdesc, + unsigned int rsize) +{ + if (rsize == 178 && + rdesc[111] == 0x06 && rdesc[112] == 0x00 && + rdesc[113] == 0xff) { + dev_info(&hdev->dev, "fixing up pc-midi keyboard report " + "descriptor\n"); + + rdesc[144] = 0x18; /* report 4: was 0x10 report count */ + } +} + +static int pk_input_mapping(struct hid_device *hdev, struct hid_input *hi, + struct hid_field *field, struct hid_usage *usage, + unsigned long **bit, int *max) +{ + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + struct pcmidi_snd *pm; + + pm = pk->pm; + + if (HID_UP_MSVENDOR == (usage->hid & HID_USAGE_PAGE) && + 1 == pm->ifnum) { + pcmidi_setup_extra_keys(pm, hi->input); + return 0; + } + + return 0; +} + + +static int pk_raw_event(struct hid_device *hdev, struct hid_report *report, + u8 *data, int size) +{ + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + int ret = 0; + + if (1 == pk->pm->ifnum) { + if (report->id == data[0]) + switch (report->id) { + case 0x01: /* midi keys (qwerty)*/ + case 0x03: /* midi keyboard (musical)*/ + case 0x04: /* extra/midi keys (qwerty)*/ + ret = pcmidi_handle_report(pk->pm, + report->id, data, size); + break; + } + } + + return ret; +} + +static int pk_probe(struct hid_device *hdev, const struct hid_device_id *id) +{ + int ret; + struct usb_interface *intf = to_usb_interface(hdev->dev.parent); + unsigned short ifnum = intf->cur_altsetting->desc.bInterfaceNumber; + unsigned long quirks = id->driver_data; + struct pk_device *pk; + struct pcmidi_snd *pm = NULL; + + pk = kzalloc(sizeof(*pk), GFP_KERNEL); + if (pk == NULL) { + dev_err(&hdev->dev, "prodikeys: can't alloc descriptor\n"); + return -ENOMEM; + } + + pk->hdev = hdev; + + pm = kzalloc(sizeof(*pm), GFP_KERNEL); + if (pm == NULL) { + dev_err(&hdev->dev, + "prodikeys: can't alloc descriptor\n"); + ret = -ENOMEM; + goto err_free; + } + + pm->pk = pk; + pk->pm = pm; + pm->ifnum = ifnum; + + hid_set_drvdata(hdev, pk); + + ret = hid_parse(hdev); + if (ret) { + dev_err(&hdev->dev, "prodikeys: hid parse failed\n"); + goto err_free; + } + + if (quirks & PK_QUIRK_NOGET) { /* hid_parse cleared all the quirks */ + hdev->quirks |= HID_QUIRK_NOGET; + } + + ret = hid_hw_start(hdev, HID_CONNECT_DEFAULT); + if (ret) { + dev_err(&hdev->dev, "prodikeys: hw start failed\n"); + goto err_free; + } + + ret = pcmidi_snd_initialise(pm); + if (ret < 0) + goto err_stop; + + return 0; +err_stop: + hid_hw_stop(hdev); +err_free: + if (pm != NULL) + kfree(pm); + + kfree(pk); + return ret; +} + +static void pk_remove(struct hid_device *hdev) +{ + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + struct pcmidi_snd *pm; + + pm = pk->pm; + if (pm) { + pcmidi_snd_terminate(pm); + kfree(pm); + } + + hid_hw_stop(hdev); + + kfree(pk); +} + +static const struct hid_device_id pk_devices[] = { + {HID_USB_DEVICE(USB_VENDOR_ID_CREATIVELABS, + USB_DEVICE_ID_PRODIKEYS_PCMIDI), + .driver_data = PK_QUIRK_NOGET}, + { } +}; +MODULE_DEVICE_TABLE(hid, pk_devices); + +static struct hid_driver pk_driver = { + .name = "prodikeys", + .id_table = pk_devices, + .report_fixup = pk_report_fixup, + .input_mapping = pk_input_mapping, + .raw_event = pk_raw_event, + .probe = pk_probe, + .remove = pk_remove, +}; + +static int pk_init(void) +{ + int ret; + + ret = hid_register_driver(&pk_driver); + if (ret) + printk(KERN_ERR "can't register prodikeys driver\n"); + + return ret; +} + +static void pk_exit(void) +{ + hid_unregister_driver(&pk_driver); +} + +module_init(pk_init); +module_exit(pk_exit); +MODULE_LICENSE("GPL"); diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/Kconfig b/drivers/hid/Kconfig --- a/drivers/hid/Kconfig 2010-03-09 13:53:05.930183131 +0000 +++ b/drivers/hid/Kconfig 2010-05-06 14:00:57.880373988 +0100 @@ -100,6 +100,22 @@ config HID_CHICONY ---help--- Support for Chicony Tactical pad. +config HID_PRODIKEYS + tristate "Prodikeys PC-MIDI Keyboard support" if EMBEDDED + depends on USB_HID && SND + select SND_RAWMIDI + default !EMBEDDED + ---help--- + Support for Prodikeys PC-MIDI Keyboard device support. + Say Y here to enable support for this device. + - Prodikeys PC-MIDI keyboard. + The Prodikeys PC-MIDI acts as a USB Audio device, with one MIDI + input and one MIDI output. These MIDI jacks appear as + a sound "card" in the ALSA sound system. + Note: if you say N here, this device will still function as a basic + multimedia keyboard, but will lack support for the musical keyboard + and some additional multimedia keys. + config HID_CYPRESS tristate "Cypress" if EMBEDDED depends on USB_HID diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/Makefile b/drivers/hid/Makefile --- a/drivers/hid/Makefile 2010-03-09 13:53:06.036326844 +0000 +++ b/drivers/hid/Makefile 2010-05-05 12:57:45.986487233 +0100 @@ -41,6 +41,7 @@ obj-$(CONFIG_HID_MONTEREY) += hid-monter obj-$(CONFIG_HID_MOSART) += hid-mosart.o obj-$(CONFIG_HID_NTRIG) += hid-ntrig.o obj-$(CONFIG_HID_ORTEK) += hid-ortek.o +obj-$(CONFIG_HID_PRODIKEYS) += hid-prodikeys.o obj-$(CONFIG_HID_QUANTA) += hid-quanta.o obj-$(CONFIG_HID_PANTHERLORD) += hid-pl.o obj-$(CONFIG_HID_PETALYNX) += hid-petalynx.o ___________________________________________________________ Copy addresses and emails from any email account to Yahoo! Mail - quick, easy and free. http://uk.docs.yahoo.com/trueswitch2.html

2 1

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

10 May '10

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

3 8

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

10 May '10

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

1 0

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

10 May '10

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

3 5

[alsa-devel] about ALC203 linux driver
by Wan ZongShun 10 May '10

10 May '10

Hi all, At present, there is a ALC203 AC97 codec in my Winbond ARM board, and I am programming the audio driver based on ALSA, Who can tell me whether the current ALSA is supporting the ALC203 driver?how to enable it? Thanks! -- *linux-arm-kernel mailing list mail addr:linux-arm-kernel@lists.infradead.org you can subscribe by: http://lists.infradead.org/mailman/listinfo/linux-arm-kernel * linux-arm-NUC900 mailing list mail addr:NUC900@googlegroups.com main web: https://groups.google.com/group/NUC900 you can subscribe it by sending me mail: mcuos.com(a)gmail.com

3 4

[alsa-devel] Xonar D1 front panel microphone not working in kernel 2.6.33
by Peter Hercek 08 May '10

08 May '10

It is not possible to record anything from my front panel microphone jack after I updated kernel to 2.6.33. It still works OK with kernel 2.6.32.10, but does not work with 2.6.33.3, and it did not work with 2.6.33.rc8 too. The front panel switch in alsamixer switches only output but not input (mic) now. Before, the switch had impact on both output and input. When I activate the back panel on the card everything is working well (back panel connectors are active, front panel connectors are inactive). When I activate the front panel then the front panel microphone is not working (back panel microphone works). The alsamixer front panel switch does not have impact on the microphone any more and no switch was added to control the microphone separately from the output. The alsa version is 1.0.23. I use 32 bit kernels. My brother uses 64 bit kernels and the problem is the same for him too. Although he uses Xonar DX. Any idea how to get this fixed? I attached output of alsa-info.sh for both cases. Maybe it is only some new configuration thing which needs to be done but if not then I should be able to test a patch to vanilla kernel 2.6.33.3 if needed. Peter.

1 1

[alsa-devel] using snd_usb_audio module
by Bee Hock Goh 08 May '10

08 May '10

Hi, I wish to make use of snd_usb_audio in my module. Is there a good document/tutorial or reference codes that I get kick-started on? Googling doesnt seem to help. thanks, Hock.

2 4

[alsa-devel] Problem about developing WM8974 audio driver for MX27ipcam
by Jie Jiang 08 May '10

08 May '10

Hi all, I'm developing a driver for wm8974 codec with Freescale mx27ipcam board with linux kernel 2.6.19.2. Our work is based on the mx27vis-wm8974 driver available in alsa-driver-1.0.22. After installing all necessary driver modules, when I trying to playback some wav file using "cat xx.wav > /dev/dsp" command, I got the following error message: "ALSA pcm_lib.c:1709: playback write error (DMA or IRQ trouble?)". This seems to be caused by the full of the transmitting buffer since I can check that "hw_ptr=0x800/buffer_size=0x4000/appl_ptr=0x4800". The hw_ptw doesn't update anymore. It seems that there is something wrong with the DMA operation. Actually, the DMA dest. address is the SSI1's TX0 FIFO. On the other hand, SSI1's SFCSR=0x440844, which indicates that there are 8 bytes data in the TX0 FIFO, which will not generate DMA request as the FIFO empty watermark is 4. It is very possible that the SSI1 does not shift data from the FIFO, which causes no DMA request generated and thus DMA freezes. Our board connects wm8974 codec to mx27 gpio PC20~23 pins (signal name are SSI1_XXX). On these io pins, we can check the FS and bit clock have been generated. The PC20~PC23 gpio pins have been configure as SSI function. My driver also configures AUDMUX host port1(HP1,wired to SSI1 internally) to peripheral port 1(PP1). But I'm not sure whether PP1 is mapped to PC20~23. Is this configuration right? If this map is wrong, the FS and Bit Clock generated by wm8974 can not be routed to SSI1 module, which is the potential reason for no data are shifted. Who can shed some light on the map between mx27 AUDMUX port and GPIO pins? Any more possible reasons which will freeze DMA transfer? Regards, Jie

2 1

[alsa-devel] [PATCH 1/2] es1968: Clear interrupts before enabling them
by Ville Syrjala 08 May '10

08 May '10

From: Ville Syrjälä <syrjala(a)sci.fi> Avoid spurious interrupts when initializing the device. Signed-off-by: Ville Syrjälä <syrjala(a)sci.fi> --- sound/pci/es1968.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index aa973ce..23a58f0 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2401,6 +2401,7 @@ static void snd_es1968_start_irq(struct es1968 *chip) w = ESM_HIRQ_DSIE | ESM_HIRQ_HW_VOLUME; if (chip->rmidi) w |= ESM_HIRQ_MPU401; + outb(w, chip->io_port + 0x1A); outw(w, chip->io_port + ESM_PORT_HOST_IRQ); } -- 1.6.4.4

2 2

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 08 May '10

08 May '10

The following changes since commit aeb29a82de7c80d4d0253b042f17eb1f725b08f1: Takashi Iwai (1): Merge branch 'for-2.6.35' of git://git.kernel.org/.../lrg/asoc-2.6 into topic/asoc are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-2.6.35 Note the change in git server - master.kernel.org appears to be having issues at the minute. Jassi Brar (12): ASoC: S3C: I2Sv2: Move defines closer to driver ASoC: SAMSUNG: I2S: Add bit definitions ASoC: S3C2412: I2S: Debug IMS field ASoC: S3C2412: I2S: Return correct source clock ASoC: S3C: I2Sv2: Discard redundant field iis_clk ASoC: S3C: I2Sv2: Unify i2s_get_clock callback ASoC: S3C64XX: I2S: Use s3c2412 defines ASoC: S3C: I2Sv2: New field for controller feature ASoC: S3C: I2S: Move set_sysclk to common code ARM: S3C2412: DMA: Remove I2S FIFO address ASoC: S3C64XX: IISv4: Add CPU driver ASoC: SMDK64XX: Switch to IISv4 CPU driver Mark Brown (2): ASoC: Add debug output tracing all cache register writes Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.35 Peter Ujfalusi (3): ASoC: core: Support for limiting the volume Revert "ASoC: tpa6130a2: Support for limiting gain" ASoC: tpa6130a2: Fix for the custom kcontrol functions arch/arm/mach-s3c2412/dma.c | 3 - .../plat-samsung/include/plat/regs-s3c2412-iis.h | 82 -------- include/sound/soc.h | 2 + include/sound/tpa6130a2-plat.h | 1 - sound/soc/codecs/tpa6130a2.c | 90 ++------- sound/soc/s3c24xx/Kconfig | 12 +- sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/jive_wm8750.c | 2 +- sound/soc/s3c24xx/regs-i2s-v2.h | 115 +++++++++++ sound/soc/s3c24xx/s3c-i2s-v2.c | 87 ++++++--- sound/soc/s3c24xx/s3c-i2s-v2.h | 11 +- sound/soc/s3c24xx/s3c2412-i2s.c | 41 +---- sound/soc/s3c24xx/s3c2412-i2s.h | 2 - sound/soc/s3c24xx/s3c64xx-i2s-v4.c | 209 ++++++++++++++++++++ sound/soc/s3c24xx/s3c64xx-i2s.c | 57 +----- sound/soc/s3c24xx/s3c64xx-i2s.h | 3 +- sound/soc/s3c24xx/smdk64xx_wm8580.c | 6 +- sound/soc/soc-cache.c | 12 ++ sound/soc/soc-core.c | 39 ++++ 19 files changed, 484 insertions(+), 292 deletions(-) delete mode 100644 arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h create mode 100644 sound/soc/s3c24xx/regs-i2s-v2.h create mode 100644 sound/soc/s3c24xx/s3c64xx-i2s-v4.c

2 1

[alsa-devel] [PATCH 0/2] ASoC: tpa6130a2: Use the core to set the gain limit
by Peter Ujfalusi 07 May '10

07 May '10

Hello, Core will receive support for limiting the volume on controls, so the custom way of doing it in the tpa driver is no longer needed. There was also a problem within the tpa driver when custom limit has been configured. The second patch fixes that. The combination of the snd_soc_limit_volume, and this series works really nicely together. --- Peter Ujfalusi (2): Revert "ASoC: tpa6130a2: Support for limiting gain" ASoC: tpa6130a2: Fix for the custom kcontrol functions include/sound/tpa6130a2-plat.h | 1 - sound/soc/codecs/tpa6130a2.c | 90 +++++++-------------------------------- 2 files changed, 16 insertions(+), 75 deletions(-)

2 3

[alsa-devel] [PATCH] ASoC: tpa6130a2: Support for limiting gain
by Peter Ujfalusi 07 May '10

07 May '10

Add support for platform dependent gain limiting on the tpa6130a2 (and tpa6140a2) Headset amplifier. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- include/sound/tpa6130a2-plat.h | 1 + sound/soc/codecs/tpa6130a2.c | 76 +++++++++++++++++++++++++++++++++++---- 2 files changed, 69 insertions(+), 8 deletions(-) diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h index e29fde6..426f627 100644 --- a/include/sound/tpa6130a2-plat.h +++ b/include/sound/tpa6130a2-plat.h @@ -31,6 +31,7 @@ enum tpa_model { struct tpa6130a2_platform_data { enum tpa_model id; int power_gpio; + int limit_gain; }; #endif diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 0cf3e38..31f67b5 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -46,6 +46,9 @@ static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { "AVdd", }; +#define TPA6130A2_GAIN_MAX 0x3f +#define TPA6140A2_GAIN_MAX 0x1f + /* This struct is used to save the context */ struct tpa6130a2_data { struct mutex mutex; @@ -54,6 +57,7 @@ struct tpa6130a2_data { int power_gpio; unsigned char power_state; enum tpa_model id; + int gain_limit; }; static int tpa6130a2_i2c_read(int reg) @@ -176,6 +180,40 @@ exit: return ret; } +static int tpa6130a2_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct tpa6130a2_data *data; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + mutex_lock(&data->mutex); + switch (mc->reg) { + case TPA6130A2_REG_VOL_MUTE: + if (data->gain_limit != mc->max) + mc->max = data->gain_limit; + break; + default: + dev_err(&tpa6130a2_client->dev, + "Invalid register: 0x02%x\n", mc->reg); + goto out; + } + if (unlikely(mc->max == 1)) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mc->max; +out: + mutex_unlock(&data->mutex); + return 0; +} + static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -239,6 +277,15 @@ static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol, return 1; } +#define SOC_SINGLE_EXT_TLV_TPA(xname, xreg, xshift, xmax, xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = tpa6130a2_info_volsw, \ + .get = tpa6130a2_get_reg, .put = tpa6130a2_set_reg, \ + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } + /* * TPA6130 volume. From -59.5 to 4 dB with increasing step size when going * down in gain. @@ -258,10 +305,9 @@ static const unsigned int tpa6130_tlv[] = { }; static const struct snd_kcontrol_new tpa6130a2_controls[] = { - SOC_SINGLE_EXT_TLV("TPA6130A2 Headphone Playback Volume", - TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0, - tpa6130a2_get_reg, tpa6130a2_set_reg, - tpa6130_tlv), + SOC_SINGLE_EXT_TLV_TPA("TPA6130A2 Headphone Playback Volume", + TPA6130A2_REG_VOL_MUTE, 0, TPA6130A2_GAIN_MAX, 0, + tpa6130_tlv), }; static const unsigned int tpa6140_tlv[] = { @@ -272,10 +318,9 @@ static const unsigned int tpa6140_tlv[] = { }; static const struct snd_kcontrol_new tpa6140a2_controls[] = { - SOC_SINGLE_EXT_TLV("TPA6140A2 Headphone Playback Volume", - TPA6130A2_REG_VOL_MUTE, 1, 0x1f, 0, - tpa6130a2_get_reg, tpa6130a2_set_reg, - tpa6140_tlv), + SOC_SINGLE_EXT_TLV_TPA("TPA6140A2 Headphone Playback Volume", + TPA6130A2_REG_VOL_MUTE, 1, TPA6140A2_GAIN_MAX, 0, + tpa6140_tlv), }; /* @@ -454,16 +499,31 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, case TPA6130A2: for (i = 0; i < ARRAY_SIZE(data->supplies); i++) data->supplies[i].supply = tpa6130a2_supply_names[i]; + if (pdata->limit_gain > 0 && + pdata->limit_gain < TPA6130A2_GAIN_MAX) + data->gain_limit = pdata->limit_gain; + else + data->gain_limit = TPA6130A2_GAIN_MAX; break; case TPA6140A2: for (i = 0; i < ARRAY_SIZE(data->supplies); i++) data->supplies[i].supply = tpa6140a2_supply_names[i];; + if (pdata->limit_gain > 0 && + pdata->limit_gain < TPA6140A2_GAIN_MAX) + data->gain_limit = pdata->limit_gain; + else + data->gain_limit = TPA6140A2_GAIN_MAX; break; default: dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", pdata->id); for (i = 0; i < ARRAY_SIZE(data->supplies); i++) data->supplies[i].supply = tpa6130a2_supply_names[i]; + if (pdata->limit_gain > 0 && + pdata->limit_gain < TPA6130A2_GAIN_MAX) + data->gain_limit = pdata->limit_gain; + else + data->gain_limit = TPA6130A2_GAIN_MAX; } ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), -- 1.7.1

3 4

[alsa-devel] [Patches_v2] Reforming S3C I2S towards supporting I2Sv4
by Jassi Brar 07 May '10

07 May '10

I am going to resubmit pending last 11 patches of the original series http://mailman.alsa-project.org/pipermail/alsa-devel/2010-March/025920.html No changes have been made other than moving dma_data from snd_soc_dai to snd_soc_pcm_stream in the new I2Sv4 controller driver.

4 5

Re: [alsa-devel] USB transfer_buffer allocations on 64bit systems
by Daniel Mack 07 May '10

07 May '10

On Fri, May 07, 2010 at 01:47:56PM +0200, Oliver Neukum wrote: > Am Freitag, 7. Mai 2010 13:42:34 schrieb Oliver Neukum: > > Am Freitag, 7. Mai 2010 11:47:37 schrieb Clemens Ladisch: > > > > In the particular case of audio drivers, though, the contents of the > > > > buffers are likely to change after the submission. What we do here > > > > is that we map the audio stream buffers which are used by ALSA to > > > > the output URBs, so they're filled asychronously. Once the buffer is > > > > actually sent out on the bus, it is believed to contain proper audio > > > > date. If it doesn't, that's due to too tight audio timing or other > > > > problems. This breaks once buffers are magically bounced in the > > > > background. > > > > > > At least the audio class and ua101 drivers don't do this and fill the > > > buffers before they are submitted. > > > > > > > So - long story short: these audio buffers need to be DMA coherent. > > > > > > Does the USB API actually guarantee that all controllers use DMA, i.e., > > > that the buffers can be filled after submission? > > > > No, you must not touch buffers after submission. This does not even > > work if we use DMA, because on some architectures this violates > > guarantees to the dma primitives. It cannot be done. We do such tricks on other OS where IRQ latency as high as some tens of milliseconds. So we queue EHCI transfers well in advance and update access their contents (for both input and output) from audio context. I considered implementing this idea to the Linux USB audio driver as well. > Or to be precise it could be done with coherent memory, but you'd risk > transfering partially updated buffers, as you cannot know when DMA will > be done and we don't guarantee that DMA will be done right as we transfer. If the memory is coherent, it should be possible. And if buffers are only partially updated, the audio buffer scheduling is too tight. IOW, the configured buffer size of audio clients in userspace is too small. This is OT now I believe, I will start another thread for this topic once I have something to show. Thanks, Daniel

1 0

Re: [alsa-devel] USB transfer_buffer allocations on 64bit systems
by Daniel Mack 07 May '10

07 May '10

On Tue, Apr 20, 2010 at 01:16:58AM +0100, Pedro Ribeiro wrote: > On 15 April 2010 16:20, Alan Stern <stern(a)rowland.harvard.edu> wrote: > > On Thu, 15 Apr 2010, Pedro Ribeiro wrote: > > > >> I enabled CONFIG_DMAR_BROKEN_GFX_WA=y and the result is the same. A > >> delay in the boot process and usb devices don't work properly, > >> including my USB mouse. > >> > >> Strange, since you have the same platform as me. The extra usb devices > >> you were seeing are because of my docking station - but it makes no > >> difference whether I'm docked or not for the purposes of the original > >> bug or this situation right now. The dmesg I'm attaching is without > >> the computer being docked. > > > > It's not possible to determine the reason for the timeout errors > > between timestamps 16 and 53 from the small amount of debugging > > information in the log. Clearly something is going wrong with the > > communication between the computer and the EHCI controller. And > > clearly the kernel config changes are responsible. > > > > But I don't know what to do to track it down any farther. > > > > Alan Stern > > > > > > I guess this is pretty much a dead end until anyone else can reproduce it! Hmm, I think I finally found the reason for all this confusion. No idea why I didn't come up with the following explanation any earlier. The problem is again (summarized): On 64bit machines, with 4GB or more, the allocated buffers for USB transfers might be beyond the 32bit boundary. In this case, the IOMMU should take care and install DMA bounce buffer to copy over the buffer before the transfer actually happens. The problem is, however, that this copy mechanism takes place when the URB with its associated buffer is submitted, not when the EHCI will actually do the transfer. In the particular case of audio drivers, though, the contents of the buffers are likely to change after the submission. What we do here is that we map the audio stream buffers which are used by ALSA to the output URBs, so they're filled asychronously. Once the buffer is actually sent out on the bus, it is believed to contain proper audio date. If it doesn't, that's due to too tight audio timing or other problems. This breaks once buffers are magically bounced in the background. So - long story short: these audio buffers need to be DMA coherent. The patch below does that, and Pedro excessively tested this patch for weeks now :) It was just the final explanation _why_ it does the right thing that was yet missing. If nobody has objections, can we still squeeze it into .34? Thanks, Daniel

2 2

[alsa-devel] [PATCH RFC v1] ASoC: core: Support for limiting the volume
by Peter Ujfalusi 07 May '10

07 May '10

Add support for the core to limit the maximum volume on an existing control. The function will modify the soc_mixer_control.max value of the given control. The new value must be lower than the original one (chip maximum) If there is a need for limiting a gain on a given control, than machine drivers can do the following in their snd_soc_dai_link.init function: snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21); This will modify the original 31 (chip maximum) to 21, so user space will not be able to set the gain higher than this. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello Mark, Liam, So this is my proposed way of doing the volume limiting within the core. By looking around the code, this should be safe. With this simple function all machine driver can limit gains on a given control in a way, that the codec/amp driver does not need to be involved. What do you think? --- Peter include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 38 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 40 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 01e9c66..9f306f0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -326,6 +326,8 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_limit_volume(struct snd_soc_codec *codec, + const char *name, int max); /** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d59076e..5968499 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2238,6 +2238,44 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); /** + * snd_soc_limit_volume - Set new limit to an existing volume control. + * + * @codec: where to look for the control + * @name: Name of the control + * @max: new maximum limit + * + * Return 0 for success, else error. + */ +int snd_soc_limit_volume(struct snd_soc_codec *codec, + const char *name, int max) +{ + struct snd_card *card = codec->card; + struct snd_kcontrol *kctl; + struct soc_mixer_control *mc; + int found = 0; + int ret = -EINVAL; + + /* Sanity check for name and max */ + if (unlikely(!name || max <= 0)) + return -EINVAL; + + list_for_each_entry(kctl, &card->controls, list) { + if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { + found = 1; + break; + } + } + if (found) { + mc = (struct soc_mixer_control *)kctl->private_value; + if (max < mc->max) + mc->max = max; + ret = 0; + } + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_limit_volume); + +/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID -- 1.7.1

2 4

[alsa-devel] [PATCH][BUGFIX for .33 and .34] hda - fix DG45ID SPDIF output
by Wu Fengguang 07 May '10

07 May '10

This reverts part of commit 52dc438606d1e, in order to fix a regression: broken SPDIF output on Intel DG45FC motherboard (IDT 92HD73E1X5 codec). --- DG45FC-IDT-codec-2.6.32 (SPDIF OK) +++ DG45FC-IDT-codec-2.6.33 (SPDIF broken) Node 0x22 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT - Pin Default 0x40f000f0: [N/A] Other at Ext N/A - Conn = Unknown, Color = Unknown - DefAssociation = 0xf, Sequence = 0x0 - Pin-ctls: 0x00: + Pin Default 0x014510a0: [Jack] SPDIF Out at Ext Rear + Conn = Optical, Color = Black + DefAssociation = 0xa, Sequence = 0x0 + Pin-ctls: 0x40: OUT Connection: 3 0x25* 0x20 0x21 Node 0x23 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT - Pin Default 0x01451140: [Jack] SPDIF Out at Ext Rear + Pin Default 0x074510b0: [Jack] SPDIF Out at Ext Rear Panel Conn = Optical, Color = Black - DefAssociation = 0x4, Sequence = 0x0 - Misc = NO_PRESENCE - Pin-ctls: 0x40: OUT + DefAssociation = 0xb, Sequence = 0x0 + Pin-ctls: 0x00: Connection: 3 0x26* 0x20 0x21 CC: <stable(a)kernel.org> CC: Alexey Fisher <bug-track(a)fisher-privat.net> Tested-by: David Härdeman <david(a)hardeman.nu> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_sigmatel.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_sigmatel.c 2010-05-06 18:10:30.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_sigmatel.c 2010-05-06 19:34:48.000000000 +0800 @@ -1544,11 +1544,9 @@ static unsigned int alienware_m17x_pin_c 0x904601b0, }; -static unsigned int intel_dg45id_pin_configs[14] = { +static unsigned int intel_dg45id_pin_configs[13] = { 0x02214230, 0x02A19240, 0x01013214, 0x01014210, - 0x01A19250, 0x01011212, 0x01016211, 0x40f000f0, - 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x014510A0, - 0x074510B0, 0x40f000f0 + 0x01A19250, 0x01011212, 0x01016211 }; static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {

2 1

[alsa-devel] [PATCH 2.6.34-rc5] HID: Prodikeys PC-MIDI HID Driver
by dhprince.devel＠yahoo 07 May '10

07 May '10

A specialised HID driver for the CreativeLabs Prodikeys PC-MIDI USB Keyboard. The Prodikeys PC-MIDI is a multifunction keyboard comprising a qwerty keyboard, multimedia keys and a touch sensitive musical keyboard. The specialised HID driver adds full support for the musical keyboard and extra multimedia keys which are not currently handled by the default HID driver. The specialised HID driver interfaces with Alsa, and presents the midi keyboard as a Rawmidi device. Sustain duration, octave shifting and the midi output channel can be read/written form userspace via sysfs. Signed-off-by: Don Prince <dhprince-devel(a)yahoo.co.uk> --- The keyboard can be viewed here http://www.prodikeys.com/products/Prodikeys_PC-MIDI/ The driver and a simple wxWidgets based UI application is available (in tar.gz format) at the Sourceforge project page http://sourceforge.net/projects/pc-midi-linux/ Apply patch with patch -p1 < <path to the driver patch> --- diff -X /opt/scripts/diff-exclude.txt -Npaur a/Documentation/ABI/testing/sysfs-driver-hid-prodikeys b/Documentation/ABI/testing/sysfs-driver-hid-prodikeys --- a/Documentation/ABI/testing/sysfs-driver-hid-prodikeys 1970-01-01 01:00:00.000000000 +0100 +++ b/Documentation/ABI/testing/sysfs-driver-hid-prodikeys 2010-05-05 12:54:37.224487960 +0100 @@ -0,0 +1,29 @@ +What: /sys/bus/hid/drivers/prodikeys/.../channel +Date: April 2010 +KernelVersion: 2.6.34 +Contact: Don Prince <dhprince.devel(a)yahoo.co.uk> +Descripts/checkpatch.plscription: + Allows control (via software) the midi channel to which + that the pc-midi keyboard will output.midi data. + Range: 0..15 + Type: Read/write +What: /sys/bus/hid/drivers/prodikeys/.../sustain +Date: April 2010 +KernelVersion: 2.6.34 +Contact: Don Prince <dhprince.devel(a)yahoo.co.uk> +Description: + Allows control (via software) the sustain duration of a + note held by the pc-midi driver. + 0 means sustain mode is disabled. + Range: 0..5000 (milliseconds) + Type: Read/write +What: /sys/bus/hid/drivers/prodikeys/.../octave +Date: April 2010 +KernelVersion: 2.6.34 +Contact: Don Prince <dhprince.devel(a)yahoo.co.uk> +Description: + Controls the octave shift modifier in the pc-midi driver. + The octave can be shifted via software up/down 2 octaves. + 0 means the no ocatve shift. + Range: -2..2 (minus 2 to plus 2) + Type: Read/Write diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/hid-core.c b/drivers/hid/hid-core.c --- a/drivers/hid/hid-core.c 2010-04-22 18:52:46.445384754 +0100 +++ b/drivers/hid/hid-core.c 2010-04-25 19:10:31.317248975 +0100 @@ -1297,6 +1297,7 @@ static const struct hid_device_id hid_bl { HID_USB_DEVICE(USB_VENDOR_ID_BELKIN, USB_DEVICE_ID_FLIP_KVM) }, { HID_USB_DEVICE(USB_VENDOR_ID_CHERRY, USB_DEVICE_ID_CHERRY_CYMOTION) }, { HID_USB_DEVICE(USB_VENDOR_ID_CHICONY, USB_DEVICE_ID_CHICONY_TACTICAL_PAD) }, + { HID_USB_DEVICE(USB_VENDOR_ID_CREATIVELABS, USB_DEVICE_ID_PRODIKEYS_PCMIDI) }, { HID_USB_DEVICE(USB_VENDOR_ID_CYPRESS, USB_DEVICE_ID_CYPRESS_BARCODE_1) }, { HID_USB_DEVICE(USB_VENDOR_ID_CYPRESS, USB_DEVICE_ID_CYPRESS_BARCODE_2) }, { HID_USB_DEVICE(USB_VENDOR_ID_CYPRESS, USB_DEVICE_ID_CYPRESS_BARCODE_3) }, diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/hid-ids.h b/drivers/hid/hid-ids.h --- a/drivers/hid/hid-ids.h 2010-04-22 18:52:46.455385039 +0100 +++ b/drivers/hid/hid-ids.h 2010-04-25 19:10:31.319248829 +0100 @@ -147,6 +147,9 @@ #define USB_DEVICE_ID_CODEMERCS_IOW_FIRST 0x1500 #define USB_DEVICE_ID_CODEMERCS_IOW_LAST 0x15ff +#define USB_VENDOR_ID_CREATIVELABS 0x041e +#define USB_DEVICE_ID_PRODIKEYS_PCMIDI 0x2801 + #define USB_VENDOR_ID_CYGNAL 0x10c4 #define USB_DEVICE_ID_CYGNAL_RADIO_SI470X 0x818a diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/hid-prodikeys.c b/drivers/hid/hid-prodikeys.c --- a/drivers/hid/hid-prodikeys.c 1970-01-01 01:00:00.000000000 +0100 +++ b/drivers/hid/hid-prodikeys.c 2010-05-05 16:18:08.896362471 +0100 @@ -0,0 +1,970 @@ +/* + * HID driver for the Prodikeys PC-MIDI Keyboard + * providing midi & extra multimedia keys functionality + * + * Copyright (c) 2009 Don Prince <dhprince.devel(a)yahoo.co.uk> + * + * Controls for Octave Shift Up/Down, Channel, and + * Sustain Duration available via sysfs. + * + */ + +/* + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + */ + +#include <linux/device.h> +#include <linux/module.h> +#include <linux/usb.h> +#include <linux/mutex.h> +#include <linux/hid.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> +#include "usbhid/usbhid.h" +#include "hid-ids.h" + + +#define pk_debug(format, arg...) \ + pr_debug("hid-prodikeys: " format "\n" , ## arg) +#define pk_error(format, arg...) \ + pr_err("hid-prodikeys: " format "\n" , ## arg) + +struct pcmidi_snd; + +struct pk_device { + unsigned long quirks; + + struct hid_device *hdev; + struct pcmidi_snd *pm; /* pcmidi device context */ +}; + +struct pcmidi_snd; + +struct pcmidi_sustain { + atomic_t in_use; + struct pcmidi_snd *pm; + struct timer_list timer; + unsigned char status; + unsigned char note; + unsigned char velocity; +}; + +#define PCMIDI_SUSTAINED_MAX 32 +struct pcmidi_snd { + struct pk_device *pk; + unsigned short ifnum; + struct hid_report *pcmidi_report6; + struct input_dev *input_ep82; + unsigned short midi_mode; + unsigned short midi_sustain_mode; + unsigned short midi_sustain; + unsigned short midi_channel; + short midi_octave; + struct pcmidi_sustain sustained_notes[PCMIDI_SUSTAINED_MAX]; + unsigned short fn_state; + unsigned short last_key[24]; + spinlock_t rawmidi_in_lock; + struct snd_card *card; + struct snd_rawmidi *rwmidi; + struct snd_rawmidi_substream *in_substream; + struct snd_rawmidi_substream *out_substream; + unsigned long in_triggered; + unsigned long out_active; + struct tasklet_struct tasklet; +}; + +#define PK_QUIRK_NOGET 0x00010000 +#define PCMIDI_MIDDLE_C 60 +#define PCMIDI_CHANNEL_MIN 0 +#define PCMIDI_CHANNEL_MAX 15 +#define PCMIDI_OCTAVE_MIN (-2) +#define PCMIDI_OCTAVE_MAX 2 +#define PCMIDI_SUSTAIN_MIN 0 +#define PCMIDI_SUSTAIN_MAX 5000 + +static const char shortname[] = "PC-MIDI"; +static const char longname[] = "Prodikeys PC-MIDI Keyboard"; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +module_param_array(id, charp, NULL, 0444); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for the PC-MIDI virtual audio driver"); +MODULE_PARM_DESC(id, "ID string for the PC-MIDI virtual audio driver"); +MODULE_PARM_DESC(enable, "Enable for the PC-MIDI virtual audio driver"); + + +/* Output routine for the sysfs channel file */ +static ssize_t show_channel(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + dbg_hid("pcmidi sysfs read channel=%u\n", pk->pm->midi_channel); + + return sprintf(buf, "%u (min:%u, max:%u)\n", pk->pm->midi_channel, + PCMIDI_CHANNEL_MIN, PCMIDI_CHANNEL_MAX); +} + +/* Input routine for the sysfs channel file */ +static ssize_t store_channel(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + unsigned channel = 0; + + if (sscanf(buf, "%u", &channel) > 0 && channel <= PCMIDI_CHANNEL_MAX) { + dbg_hid("pcmidi sysfs write channel=%u\n", channel); + pk->pm->midi_channel = channel; + return strlen(buf); + } + return -EINVAL; +} + +static DEVICE_ATTR(channel, S_IRUGO | S_IWUGO, show_channel, + store_channel); + +static struct device_attribute *sysfs_device_attr_channel = { + &dev_attr_channel, + }; + +/* Output routine for the sysfs sustain file */ +static ssize_t show_sustain(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + dbg_hid("pcmidi sysfs read sustain=%u\n", pk->pm->midi_sustain); + + return sprintf(buf, "%u (off:%u, max:%u (ms))\n", pk->pm->midi_sustain, + PCMIDI_SUSTAIN_MIN, PCMIDI_SUSTAIN_MAX); +} + +/* Input routine for the sysfs sustain file */ +static ssize_t store_sustain(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + unsigned sustain = 0; + + if (sscanf(buf, "%u", &sustain) > 0 && sustain <= PCMIDI_SUSTAIN_MAX) { + dbg_hid("pcmidi sysfs write sustain=%u\n", sustain); + pk->pm->midi_sustain = sustain; + pk->pm->midi_sustain_mode = + (0 == sustain || !pk->pm->midi_mode) ? 0 : 1; + return strlen(buf); + } + return -EINVAL; +} + +static DEVICE_ATTR(sustain, S_IRUGO | S_IWUGO, show_sustain, + store_sustain); + +static struct device_attribute *sysfs_device_attr_sustain = { + &dev_attr_sustain, + }; + +/* Output routine for the sysfs octave file */ +static ssize_t show_octave(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + dbg_hid("pcmidi sysfs read octave=%d\n", pk->pm->midi_octave); + + return sprintf(buf, "%d (min:%d, max:%d)\n", pk->pm->midi_octave, + PCMIDI_OCTAVE_MIN, PCMIDI_OCTAVE_MAX); +} + +/* Input routine for the sysfs octave file */ +static ssize_t store_octave(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) +{ + struct hid_device *hdev = container_of(dev, struct hid_device, dev); + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + + int octave = 0; + + if (sscanf(buf, "%d", &octave) > 0 && + octave >= PCMIDI_OCTAVE_MIN && octave <= PCMIDI_OCTAVE_MAX) { + dbg_hid("pcmidi sysfs write octave=%d\n", octave); + pk->pm->midi_octave = octave; + return strlen(buf); + } + return -EINVAL; +} + +static DEVICE_ATTR(octave, S_IRUGO | S_IWUGO, show_octave, + store_octave); + +static struct device_attribute *sysfs_device_attr_octave = { + &dev_attr_octave, + }; + + +static void pcmidi_send_note(struct pcmidi_snd *pm, + unsigned char status, unsigned char note, unsigned char velocity) +{ + unsigned long flags; + unsigned char buffer[3]; + int res = 0; + + buffer[0] = status; + buffer[1] = note; + buffer[2] = velocity; + + spin_lock_irqsave(&pm->rawmidi_in_lock, flags); + + if (!pm->in_substream) + goto drop_note; + if (!test_bit(pm->in_substream->number, &pm->in_triggered)) + goto drop_note; + + res = snd_rawmidi_receive(pm->in_substream, buffer, 3); + +drop_note: + spin_unlock_irqrestore(&pm->rawmidi_in_lock, flags); + + return; +} + +void pcmidi_sustained_note_release(unsigned long data) +{ + struct pcmidi_sustain *pms = (struct pcmidi_sustain *)data; + + pcmidi_send_note(pms->pm, pms->status, pms->note, pms->velocity); + atomic_set(&pms->in_use, 0); +} + +void init_sustain_timers(struct pcmidi_snd *pm) +{ + struct pcmidi_sustain *pms; + unsigned i; + + for (i = 0; i < PCMIDI_SUSTAINED_MAX; i++) { + pms = &pm->sustained_notes[i]; + atomic_set(&pms->in_use, 0); + pms->pm = pm; + setup_timer(&pms->timer, pcmidi_sustained_note_release, + (unsigned long)pms); + } +} + +void stop_sustain_timers(struct pcmidi_snd *pm) +{ + struct pcmidi_sustain *pms; + unsigned i; + + for (i = 0; i < PCMIDI_SUSTAINED_MAX; i++) { + pms = &pm->sustained_notes[i]; + atomic_set(&pms->in_use, 1); + del_timer_sync(&pms->timer); + } +} + +static int pcmidi_get_output_report(struct pcmidi_snd *pm) +{ + struct hid_device *hdev = pm->pk->hdev; + struct hid_report *report; + + list_for_each_entry(report, + &hdev->report_enum[HID_OUTPUT_REPORT].report_list, list) { + if (!(6 == report->id)) + continue; + + if (report->maxfield < 1) { + dev_err(&hdev->dev, "output report is empty\n"); + break; + } + if (report->field[0]->report_count != 2) { + dev_err(&hdev->dev, "field count too low\n"); + break; + } + pm->pcmidi_report6 = report; + return 0; + } + /* should never get here */ + return -ENODEV; +} + +static void pcmidi_submit_output_report(struct pcmidi_snd *pm, int state) +{ + struct hid_device *hdev = pm->pk->hdev; + struct hid_report *report = pm->pcmidi_report6; + report->field[0]->value[0] = 0x01; + report->field[0]->value[1] = state; + + usbhid_submit_report(hdev, report, USB_DIR_OUT); +} + + +static int pcmidi_handle_report1(struct pcmidi_snd *pm, u8 *data) +{ + u32 bit_mask; + + bit_mask = data[1]; + bit_mask = (bit_mask << 8) | data[2]; + bit_mask = (bit_mask << 8) | data[3]; + + dbg_hid("pcmidi mode: %d\n", pm->midi_mode); + + /*KEY_MAIL or octave down*/ + if (pm->midi_mode && bit_mask == 0x004000) { + /* octave down */ + pm->midi_octave--; + if (pm->midi_octave < -2) + pm->midi_octave = -2; + dbg_hid("pcmidi mode: %d octave: %d\n", + pm->midi_mode, pm->midi_octave); + return 1; + } + /*KEY_WWW or sustain*/ + else if (pm->midi_mode && bit_mask == 0x000004) { + /* sustain on/off*/ + pm->midi_sustain_mode ^= 0x1; + return 1; + } + + return 0; /* continue key processing */ +} + +static int pcmidi_handle_report3(struct pcmidi_snd *pm, u8 *data, int size) +{ + struct pcmidi_sustain *pms; + unsigned i, j; + unsigned char status, note, velocity; + + unsigned num_notes = (size-1)/2; + for (j = 0; j < num_notes; j++) { + note = data[j*2+1]; + velocity = data[j*2+2]; + + if (note < 0x81) { /* note on */ + status = 128 + 16 + pm->midi_channel; /* 1001nnnn */ + note = note - 0x54 + PCMIDI_MIDDLE_C + + (pm->midi_octave * 12); + } else { /* note off */ + status = 128 + pm->midi_channel; /* 1000nnnn */ + note = note - 0x94 + PCMIDI_MIDDLE_C + + (pm->midi_octave*12); + + if (pm->midi_sustain_mode) { + for (i = 0; i < PCMIDI_SUSTAINED_MAX; i++) { + pms = &pm->sustained_notes[i]; + if (!atomic_read(&pms->in_use)) { + pms->status = status; + pms->note = note; + pms->velocity = velocity; + atomic_set(&pms->in_use, 1); + + mod_timer(&pms->timer, + jiffies + + msecs_to_jiffies(pm->midi_sustain)); + return 1; + } + } + } + } + pcmidi_send_note(pm, status, note, velocity); + } + + return 1; +} + +static int pcmidi_handle_report4(struct pcmidi_snd *pm, u8 *data) +{ + unsigned key; + u32 bit_mask; + u32 bit_index; + + bit_mask = data[1]; + bit_mask = (bit_mask << 8) | data[2]; + bit_mask = (bit_mask << 8) | data[3]; + + /* break keys */ + for (bit_index = 0; bit_index < 24; bit_index++) { + key = pm->last_key[bit_index]; + if (!((0x01 << bit_index) & bit_mask)) { + input_event(pm->input_ep82, EV_KEY, + pm->last_key[bit_index], 0); + pm->last_key[bit_index] = 0; + } + } + + /* make keys */ + for (bit_index = 0; bit_index < 24; bit_index++) { + key = 0; + switch ((0x01 << bit_index) & bit_mask) { + case 0x000010: /* Fn lock*/ + pm->fn_state ^= 0x000010; + if (pm->fn_state) + pcmidi_submit_output_report(pm, 0xc5); + else + pcmidi_submit_output_report(pm, 0xc6); + continue; + case 0x020000: /* midi launcher..send a key (qwerty) or not? */ + pcmidi_submit_output_report(pm, 0xc1); + pm->midi_mode ^= 0x01; + + dbg_hid("pcmidi mode: %d\n", pm->midi_mode); + continue; + case 0x100000: /* KEY_MESSENGER or octave up */ + dbg_hid("pcmidi mode: %d\n", pm->midi_mode); + if (pm->midi_mode) { + pm->midi_octave++; + if (pm->midi_octave > 2) + pm->midi_octave = 2; + dbg_hid("pcmidi mode: %d octave: %d\n", + pm->midi_mode, pm->midi_octave); + continue; + } else + key = KEY_MESSENGER; + break; + case 0x400000: + key = KEY_CALENDAR; + break; + case 0x080000: + key = KEY_ADDRESSBOOK; + break; + case 0x040000: + key = KEY_DOCUMENTS; + break; + case 0x800000: + key = KEY_WORDPROCESSOR; + break; + case 0x200000: + key = KEY_SPREADSHEET; + break; + case 0x010000: + key = KEY_COFFEE; + break; + case 0x000100: + key = KEY_HELP; + break; + case 0x000200: + key = KEY_SEND; + break; + case 0x000400: + key = KEY_REPLY; + break; + case 0x000800: + key = KEY_FORWARDMAIL; + break; + case 0x001000: + key = KEY_NEW; + break; + case 0x002000: + key = KEY_OPEN; + break; + case 0x004000: + key = KEY_CLOSE; + break; + case 0x008000: + key = KEY_SAVE; + break; + case 0x000001: + key = KEY_UNDO; + break; + case 0x000002: + key = KEY_REDO; + break; + case 0x000004: + key = KEY_SPELLCHECK; + break; + case 0x000008: + key = KEY_PRINT; + break; + } + if (key) { + input_event(pm->input_ep82, EV_KEY, key, 1); + pm->last_key[bit_index] = key; + } + } + + return 1; +} + +int pcmidi_handle_report( + struct pcmidi_snd *pm, unsigned report_id, u8 *data, int size) +{ + int ret = 0; + + switch (report_id) { + case 0x01: /* midi keys (qwerty)*/ + ret = pcmidi_handle_report1(pm, data); + break; + case 0x03: /* midi keyboard (musical)*/ + ret = pcmidi_handle_report3(pm, data, size); + break; + case 0x04: /* multimedia/midi keys (qwerty)*/ + ret = pcmidi_handle_report4(pm, data); + break; + } + return ret; +} + +void pcmidi_setup_extra_keys(struct pcmidi_snd *pm, struct input_dev *input) +{ + /* reassigned functionality for N/A keys + MY PICTURES => KEY_WORDPROCESSOR + MY MUSIC=> KEY_SPREADSHEET + */ + unsigned int keys[] = { + KEY_FN, + KEY_MESSENGER, KEY_CALENDAR, + KEY_ADDRESSBOOK, KEY_DOCUMENTS, + KEY_WORDPROCESSOR, + KEY_SPREADSHEET, + KEY_COFFEE, + KEY_HELP, KEY_SEND, + KEY_REPLY, KEY_FORWARDMAIL, + KEY_NEW, KEY_OPEN, + KEY_CLOSE, KEY_SAVE, + KEY_UNDO, KEY_REDO, + KEY_SPELLCHECK, KEY_PRINT, + 0 + }; + + unsigned int *pkeys = &keys[0]; + unsigned short i; + + if (pm->ifnum != 1) /* only set up ONCE for interace 1 */ + return; + + pm->input_ep82 = input; + + for (i = 0; i < 24; i++) + pm->last_key[i] = 0; + + while (*pkeys != 0) { + set_bit(*pkeys, pm->input_ep82->keybit); + ++pkeys; + } +} + + +static int pcmidi_set_operational(struct pcmidi_snd *pm) +{ + if (pm->ifnum != 1) + return 0; /* only set up ONCE for interace 1 */ + + pcmidi_get_output_report(pm); + pcmidi_submit_output_report(pm, 0xc1); + return 0; +} + + +static int pcmidi_snd_free(struct snd_device *dev) +{ + return 0; +} + +static void pcmidi_out_tasklet(unsigned long data) +{ + struct pcmidi_snd *pm = (struct pcmidi_snd *)data; + + if (!pm->out_active) + return; + if (!pm->out_substream) + return; + + while (1) { + /* just read them and drop them */ + u8 b; + if (snd_rawmidi_transmit(pm->out_substream, &b, 1) != 1) { + pm->out_active = 0; + break; + } + } +} + +static int pcmidi_out_open(struct snd_rawmidi_substream *substream) +{ + struct pcmidi_snd *pm = substream->rmidi->private_data; + + dbg_hid("pcmidi out open\n"); + pm->out_substream = substream; + return 0; +} + +static int pcmidi_out_close(struct snd_rawmidi_substream *substream) +{ + dbg_hid("pcmidi out close\n"); + return 0; +} + +static void pcmidi_out_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct pcmidi_snd *pm = substream->rmidi->private_data; + + dbg_hid("pcmidi out trigger %d\n", up); + + pm->out_active = up; + if (up) + tasklet_hi_schedule(&pm->tasklet); +} + +static int pcmidi_in_open(struct snd_rawmidi_substream *substream) +{ + struct pcmidi_snd *pm = substream->rmidi->private_data; + + dbg_hid("pcmidi in open\n"); + pm->in_substream = substream; + return 0; +} + +static int pcmidi_in_close(struct snd_rawmidi_substream *substream) +{ + dbg_hid("pcmidi in close\n"); + return 0; +} + +static void pcmidi_in_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct pcmidi_snd *pm = substream->rmidi->private_data; + + dbg_hid("pcmidi in trigger %d\n", up); + + if (up) + set_bit(substream->number, &pm->in_triggered); + else + clear_bit(substream->number, &pm->in_triggered); + +} + +static struct snd_rawmidi_ops pcmidi_in_ops = { + .open = pcmidi_in_open, + .close = pcmidi_in_close, + .trigger = pcmidi_in_trigger +}; + +static struct snd_rawmidi_ops pcmidi_out_ops = { + .open = pcmidi_out_open, + .close = pcmidi_out_close, + .trigger = pcmidi_out_trigger +}; + +int pcmidi_snd_initialise(struct pcmidi_snd *pm) +{ + static int dev; + struct snd_card *card; + struct snd_rawmidi *rwmidi; + int err; + int out_ports = 1; + int in_ports = 1; + + static struct snd_device_ops ops = { + .dev_free = pcmidi_snd_free, + }; + + if (pm->ifnum != 1) + return 0; /* only set up midi device ONCE for interace 1 */ + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* Setup sound card */ + tasklet_init(&pm->tasklet, pcmidi_out_tasklet, (unsigned long)pm); + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) { + pk_error("failed to create pc-midi sound card\n"); + err = -ENOMEM; + goto fail; + } + pm->card = card; + + /* Setup sound device */ + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, pm, &ops); + if (err < 0) { + pk_error("failed to create pc-midi sound device: error %d\n", + err); + goto fail; + } + + strncpy(card->driver, shortname, sizeof(card->driver)); + strncpy(card->shortname, shortname, sizeof(card->shortname)); + strncpy(card->longname, longname, sizeof(card->longname)); + + /* Set up rawmidi */ + snd_component_add(card, "MIDI"); + err = snd_rawmidi_new(card, card->shortname, 0, + out_ports, in_ports, &rwmidi); + if (err < 0) { + pk_error("failed to create pc-midi rawmidi device: error %d\n", + err); + goto fail; + } + pm->rwmidi = rwmidi; + strncpy(rwmidi->name, card->shortname, sizeof(rwmidi->name)); + rwmidi->info_flags = SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_OUTPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; + rwmidi->private_data = pm; + + snd_rawmidi_set_ops(rwmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &pcmidi_out_ops); + snd_rawmidi_set_ops(rwmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &pcmidi_in_ops); + + snd_card_set_dev(card, &pm->pk->hdev->dev); + + /* register it */ + err = snd_card_register(card); + if (err < 0) { + pk_error("failed to register pc-midi sound card: error %d\n", + err); + goto fail; + } + + /* create sysfs variables */ + err = device_create_file(&pm->pk->hdev->dev, + sysfs_device_attr_channel); + if (err < 0) { + pk_error("failed to create sysfs attribute channel: error %d\n", + err); + goto fail; + } + + err = device_create_file(&pm->pk->hdev->dev, + sysfs_device_attr_sustain); + if (err < 0) { + pk_error("failed to create sysfs attribute sustain: error %d\n", + err); + goto fail_attr_sustain; + } + + err = device_create_file(&pm->pk->hdev->dev, + sysfs_device_attr_octave); + if (err < 0) { + pk_error("failed to create sysfs attribute octave: error %d\n", + err); + goto fail_attr_octave; + } + + spin_lock_init(&pm->rawmidi_in_lock); + + init_sustain_timers(pm); + pcmidi_set_operational(pm); + + dbg_hid("pcmidi_snd_initialise finished ok\n"); + return 0; + +fail_attr_sustain: + device_remove_file(&pm->pk->hdev->dev, sysfs_device_attr_sustain); +fail_attr_octave: + device_remove_file(&pm->pk->hdev->dev, sysfs_device_attr_channel); +fail: + if (pm->card) { + snd_card_free(pm->card); + pm->card = NULL; + } + return err; +} + +int pcmidi_snd_terminate(struct pcmidi_snd *pm) +{ + if (pm->card) { + tasklet_disable(&pm->tasklet); + tasklet_kill(&pm->tasklet); + + device_remove_file(&pm->pk->hdev->dev, + sysfs_device_attr_channel); + device_remove_file(&pm->pk->hdev->dev, + sysfs_device_attr_sustain); + device_remove_file(&pm->pk->hdev->dev, + sysfs_device_attr_octave); + + stop_sustain_timers(pm); + + snd_card_disconnect(pm->card); + snd_card_free_when_closed(pm->card); + } + + return 0; +} + +/* + * PC-MIDI report descriptor for report id is wrong. + */ +static void pk_report_fixup(struct hid_device *hdev, __u8 *rdesc, + unsigned int rsize) +{ + if (rsize == 178 && + rdesc[111] == 0x06 && rdesc[112] == 0x00 && + rdesc[113] == 0xff) { + dev_info(&hdev->dev, "fixing up pc-midi keyboard report " + "descriptor\n"); + + rdesc[144] = 0x18; /* report 4: was 0x10 report count */ + } +} + +static int pk_input_mapping(struct hid_device *hdev, struct hid_input *hi, + struct hid_field *field, struct hid_usage *usage, + unsigned long **bit, int *max) +{ + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + struct pcmidi_snd *pm; + + pm = pk->pm; + + if (HID_UP_MSVENDOR == (usage->hid & HID_USAGE_PAGE) && + 1 == pm->ifnum) { + pcmidi_setup_extra_keys(pm, hi->input); + return 0; + } + + return 0; +} + + +static int pk_raw_event(struct hid_device *hdev, struct hid_report *report, + u8 *data, int size) +{ + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + int ret = 0; + + if (1 == pk->pm->ifnum) { + if (report->id == data[0]) + switch (report->id) { + case 0x01: /* midi keys (qwerty)*/ + case 0x03: /* midi keyboard (musical)*/ + case 0x04: /* extra/midi keys (qwerty)*/ + ret = pcmidi_handle_report(pk->pm, + report->id, data, size); + break; + } + } + + return ret; +} + +static int pk_probe(struct hid_device *hdev, const struct hid_device_id *id) +{ + int ret; + struct usb_interface *intf = to_usb_interface(hdev->dev.parent); + unsigned short ifnum = intf->cur_altsetting->desc.bInterfaceNumber; + unsigned long quirks = id->driver_data; + struct pk_device *pk; + struct pcmidi_snd *pm = NULL; + + pk = kzalloc(sizeof(*pk), GFP_KERNEL); + if (pk == NULL) { + dev_err(&hdev->dev, "prodikeys: can't alloc descriptor\n"); + return -ENOMEM; + } + + pk->hdev = hdev; + + pm = kzalloc(sizeof(*pm), GFP_KERNEL); + if (pm == NULL) { + dev_err(&hdev->dev, + "prodikeys: can't alloc descriptor\n"); + ret = -ENOMEM; + goto err_free; + } + pm->pk = pk; + pk->pm = pm; + pm->ifnum = ifnum; + + hid_set_drvdata(hdev, pk); + + ret = hid_parse(hdev); + if (ret) { + dev_err(&hdev->dev, "prodikeys hid parse failed\n"); + goto err_free; + } + + if (quirks & PK_QUIRK_NOGET) { /* hid_parse cleared all the quirks */ + hdev->quirks |= HID_QUIRK_NOGET; + } + + ret = hid_hw_start(hdev, HID_CONNECT_DEFAULT); + if (ret) { + dev_err(&hdev->dev, "prodikeys hw start failed\n"); + goto err_free; + } + + + ret = pcmidi_snd_initialise(pm); + if (ret < 0) + goto err_stop; + + return 0; +err_stop: + hid_hw_stop(hdev); +err_free: + if (pm != NULL) + kfree(pm); + + kfree(pk); + return ret; +} + +static void pk_remove(struct hid_device *hdev) +{ + struct pk_device *pk = (struct pk_device *)hid_get_drvdata(hdev); + struct pcmidi_snd *pm; + + hid_hw_stop(hdev); + + pm = pk->pm; + if (pm) { + pcmidi_snd_terminate(pm); + kfree(pm); + } + + kfree(pk); +} + +static const struct hid_device_id pk_devices[] = { + {HID_USB_DEVICE(USB_VENDOR_ID_CREATIVELABS, + USB_DEVICE_ID_PRODIKEYS_PCMIDI), + .driver_data = PK_QUIRK_NOGET}, + { } +}; +MODULE_DEVICE_TABLE(hid, pk_devices); + +static struct hid_driver pk_driver = { + .name = "prodikeys", + .id_table = pk_devices, + .report_fixup = pk_report_fixup, + .input_mapping = pk_input_mapping, + .raw_event = pk_raw_event, + .probe = pk_probe, + .remove = pk_remove, +}; + +static int pk_init(void) +{ + return hid_register_driver(&pk_driver); +} + +static void pk_exit(void) +{ + hid_unregister_driver(&pk_driver); +} + +module_init(pk_init); +module_exit(pk_exit); +MODULE_LICENSE("GPL"); + diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/Kconfig b/drivers/hid/Kconfig --- a/drivers/hid/Kconfig 2010-03-09 13:53:05.930183131 +0000 +++ b/drivers/hid/Kconfig 2010-05-05 12:57:35.691487442 +0100 @@ -100,6 +100,22 @@ config HID_CHICONY ---help--- Support for Chicony Tactical pad. +config HID_PRODIKEYS + tristate "Prodikeys PC-MIDI Keyboard support" if EMBEDDED + depends on USB_HID && SND + select SND_RAWMIDI + default !EMBEDDED + ---help--- + Support for Prodikeys PC-MIDI Keyboard device support. + Say Y here to enable support for this device. + - Prodikeys PC-MIDI keyboard. + The Prodikeys PC-MIDI acts as a USB Audio device, with one MIDI + input and one MIDI output. These MIDI jacks appear as + a sound "card" in the ALSA sound system. + Note: if you say N here, this device will still function as a basic + multimedia keyboard, but will lack support for the musical keyboard + and some additional multimedia keys. + config HID_CYPRESS tristate "Cypress" if EMBEDDED depends on USB_HID diff -X /opt/scripts/diff-exclude.txt -Npaur a/drivers/hid/Makefile b/drivers/hid/Makefile --- a/drivers/hid/Makefile 2010-03-09 13:53:06.036326844 +0000 +++ b/drivers/hid/Makefile 2010-05-05 12:57:45.986487233 +0100 @@ -41,6 +41,7 @@ obj-$(CONFIG_HID_MONTEREY) += hid-monter obj-$(CONFIG_HID_MOSART) += hid-mosart.o obj-$(CONFIG_HID_NTRIG) += hid-ntrig.o obj-$(CONFIG_HID_ORTEK) += hid-ortek.o +obj-$(CONFIG_HID_PRODIKEYS) += hid-prodikeys.o obj-$(CONFIG_HID_QUANTA) += hid-quanta.o obj-$(CONFIG_HID_PANTHERLORD) += hid-pl.o obj-$(CONFIG_HID_PETALYNX) += hid-petalynx.o ___________________________________________________________ The all-new Yahoo! Mail goes wherever you go - free your email address from your Internet provider. http://uk.docs.yahoo.com/nowyoucan.html

3 3

[alsa-devel] [PATCH 1/2] snd-maestro3: Make hardware volume buttons an input device (rev2)
by Hans de Goede 07 May '10

07 May '10

While working on the sound suspend / resume problems with my laptop I noticed that the hardware volume handling code in essence just detects key presses, and then does some hardcoded modification of the master volume based on which key is pressed. This made me think that clearly the right thing to do here is just report these keypresses to userspace and let userspace decide what to with them. This patch adds a Kconfig option which when enabled reports the volume buttons as keypresses using an input device. When enabled this option also gets rid of the ugly direct ac97 writes from the tasklet, the ac97lock and the need for using a tasklet in general. As an added bonus the keys now work identical to volume keys on a (usb) keyboard with multimedia keys, providing visual feedback of the volume level change, and a better range of the volume control (with a properly configured desktop environment). Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/Kconfig | 10 ++++ sound/pci/maestro3.c | 116 +++++++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 124 insertions(+), 2 deletions(-) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 351654c..2bca3d6 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -655,6 +655,16 @@ config SND_MAESTRO3 To compile this driver as a module, choose M here: the module will be called snd-maestro3. +config SND_MAESTRO3_INPUT + bool "Enable input device for maestro3 volume buttons" + depends on SND_MAESTRO3 + depends on INPUT=y || INPUT=SND_MAESTRO3 + help + If you say Y here, you will get an input device which reports + keypresses for the volume buttons connected to the maestro3 chip. + If you say N the buttons will directly control the master volume. + It is recommended to say Y. + config SND_MIXART tristate "Digigram miXart" select SND_HWDEP diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 0ed634a..1070250 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -41,6 +41,7 @@ #include <linux/vmalloc.h> #include <linux/moduleparam.h> #include <linux/firmware.h> +#include <linux/input.h> #include <sound/core.h> #include <sound/info.h> #include <sound/control.h> @@ -844,11 +845,17 @@ struct snd_m3 { struct m3_dma *substreams; spinlock_t reg_lock; - spinlock_t ac97_lock; +#ifdef CONFIG_SND_MAESTRO3_INPUT + struct input_dev *input_dev; + char phys[64]; /* physical device path */ +#else + spinlock_t ac97_lock; struct snd_kcontrol *master_switch; struct snd_kcontrol *master_volume; struct tasklet_struct hwvol_tq; +#endif + unsigned int in_suspend; #ifdef CONFIG_PM @@ -1606,7 +1613,9 @@ static void snd_m3_update_hw_volume(unsigned long private_data) { struct snd_m3 *chip = (struct snd_m3 *) private_data; int x, val; +#ifndef CONFIG_SND_MAESTRO3_INPUT unsigned long flags; +#endif /* Figure out which volume control button was pushed, based on differences from the default register @@ -1632,6 +1641,7 @@ static void snd_m3_update_hw_volume(unsigned long private_data) if (chip->in_suspend) return; +#ifndef CONFIG_SND_MAESTRO3_INPUT if (!chip->master_switch || !chip->master_volume) return; @@ -1677,6 +1687,35 @@ static void snd_m3_update_hw_volume(unsigned long private_data) break; } spin_unlock_irqrestore(&chip->ac97_lock, flags); +#else + if (!chip->input_dev) + return; + + val = 0; + switch (x) { + case 0x88: + /* The counters have not changed, yet we've received a HV + interrupt. According to tests run by various people this + happens when pressing the mute button. */ + val = KEY_MUTE; + break; + case 0xaa: + /* counters increased by 1 -> volume up */ + val = KEY_VOLUMEUP; + break; + case 0x66: + /* counters decreased by 1 -> volume down */ + val = KEY_VOLUMEDOWN; + break; + } + + if (val) { + input_report_key(chip->input_dev, val, 1); + input_sync(chip->input_dev); + input_report_key(chip->input_dev, val, 0); + input_sync(chip->input_dev); + } +#endif } static irqreturn_t snd_m3_interrupt(int irq, void *dev_id) @@ -1691,7 +1730,11 @@ static irqreturn_t snd_m3_interrupt(int irq, void *dev_id) return IRQ_NONE; if (status & HV_INT_PENDING) +#ifdef CONFIG_SND_MAESTRO3_INPUT + snd_m3_update_hw_volume((unsigned long)chip); +#else tasklet_schedule(&chip->hwvol_tq); +#endif /* * ack an assp int if its running @@ -1957,18 +2000,24 @@ static unsigned short snd_m3_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct snd_m3 *chip = ac97->private_data; +#ifndef CONFIG_SND_MAESTRO3_INPUT unsigned long flags; +#endif unsigned short data = 0xffff; if (snd_m3_ac97_wait(chip)) goto fail; +#ifndef CONFIG_SND_MAESTRO3_INPUT spin_lock_irqsave(&chip->ac97_lock, flags); +#endif snd_m3_outb(chip, 0x80 | (reg & 0x7f), CODEC_COMMAND); if (snd_m3_ac97_wait(chip)) goto fail_unlock; data = snd_m3_inw(chip, CODEC_DATA); fail_unlock: +#ifndef CONFIG_SND_MAESTRO3_INPUT spin_unlock_irqrestore(&chip->ac97_lock, flags); +#endif fail: return data; } @@ -1977,14 +2026,20 @@ static void snd_m3_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct snd_m3 *chip = ac97->private_data; +#ifndef CONFIG_SND_MAESTRO3_INPUT unsigned long flags; +#endif if (snd_m3_ac97_wait(chip)) return; +#ifndef CONFIG_SND_MAESTRO3_INPUT spin_lock_irqsave(&chip->ac97_lock, flags); +#endif snd_m3_outw(chip, val, CODEC_DATA); snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND); +#ifndef CONFIG_SND_MAESTRO3_INPUT spin_unlock_irqrestore(&chip->ac97_lock, flags); +#endif } @@ -2091,7 +2146,9 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) { struct snd_ac97_bus *pbus; struct snd_ac97_template ac97; +#ifndef CONFIG_SND_MAESTRO3_INPUT struct snd_ctl_elem_id elem_id; +#endif int err; static struct snd_ac97_bus_ops ops = { .write = snd_m3_ac97_write, @@ -2111,6 +2168,7 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) schedule_timeout_uninterruptible(msecs_to_jiffies(100)); snd_ac97_write(chip->ac97, AC97_PCM, 0); +#ifndef CONFIG_SND_MAESTRO3_INPUT memset(&elem_id, 0, sizeof(elem_id)); elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; strcpy(elem_id.name, "Master Playback Switch"); @@ -2119,6 +2177,7 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; strcpy(elem_id.name, "Master Playback Volume"); chip->master_volume = snd_ctl_find_id(chip->card, &elem_id); +#endif return 0; } @@ -2398,6 +2457,11 @@ static int snd_m3_free(struct snd_m3 *chip) struct m3_dma *s; int i; +#ifdef CONFIG_SND_MAESTRO3_INPUT + if (chip->input_dev) + input_unregister_device(chip->input_dev); +#endif + if (chip->substreams) { spin_lock_irq(&chip->reg_lock); for (i = 0; i < chip->num_substreams; i++) { @@ -2524,6 +2588,41 @@ static int m3_resume(struct pci_dev *pci) } #endif /* CONFIG_PM */ +#ifdef CONFIG_SND_MAESTRO3_INPUT +static int __devinit snd_m3_input_register(struct snd_m3 *chip) +{ + struct input_dev *input_dev; + int err; + + input_dev = input_allocate_device(); + if (!input_dev) + return -ENOMEM; + + snprintf(chip->phys, sizeof(chip->phys), "pci-%s/input0", + pci_name(chip->pci)); + + input_dev->name = chip->card->driver; + input_dev->phys = chip->phys; + input_dev->id.bustype = BUS_PCI; + input_dev->id.vendor = chip->pci->vendor; + input_dev->id.product = chip->pci->device; + input_dev->dev.parent = &chip->pci->dev; + + __set_bit(EV_KEY, input_dev->evbit); + __set_bit(KEY_MUTE, input_dev->keybit); + __set_bit(KEY_VOLUMEDOWN, input_dev->keybit); + __set_bit(KEY_VOLUMEUP, input_dev->keybit); + + err = input_register_device(input_dev); + if (err) { + input_free_device(input_dev); + return err; + } + + chip->input_dev = input_dev; + return 0; +} +#endif /* CONFIG_INPUT */ /* */ @@ -2567,7 +2666,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, } spin_lock_init(&chip->reg_lock); +#ifndef CONFIG_SND_MAESTRO3_INPUT spin_lock_init(&chip->ac97_lock); +#endif switch (pci->device) { case PCI_DEVICE_ID_ESS_ALLEGRO: @@ -2650,7 +2751,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, snd_m3_hv_init(chip); +#ifndef CONFIG_SND_MAESTRO3_INPUT tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); +#endif if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED, card->driver, chip)) { @@ -2682,7 +2785,16 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, if ((err = snd_m3_pcm(chip, 0)) < 0) return err; - + +#ifdef CONFIG_SND_MAESTRO3_INPUT + if (chip->hv_config & HV_CTRL_ENABLE) { + err = snd_m3_input_register(chip); + if (err) + snd_printk(KERN_WARNING "Input device registration " + "failed with error %i", err); + } +#endif + snd_m3_enable_ints(chip); snd_m3_assp_continue(chip); -- 1.7.0.1

2 3

Re: [alsa-devel] iec958 regression with alsa 1.0.22 and kernel 2.6.33 on Intel DG45FC motherboard
by Wu Fengguang 06 May '10

06 May '10

Hi David, On Thu, May 06, 2010 at 07:05:16AM +0800, David Härdeman wrote: > A recent upgrade to kernel 2.6.33.x and alsa 1.0.22 broke spdif (iec958) Upgraded from which kernel? > output on my Intel DG45FC motherboard (IDT 92HD73E1X5 codec, > snd-hda-intel driver). > > After playing around a bit with module params, it turns out that loading > snd-hda-intel with model=no-jd fixes this problem (previously the module > worked without any model option at all, note also that model=intel does > not work which is kinda weird). You mean model=intel never worked? > Any suggestions on a proper fix? The current situation is not really > user-friendly... Thanks, Fengguang

2 4

[alsa-devel] SPDIF out with HDA/Conexant 20561?
by pl bossart 06 May '10

06 May '10

Has anyone had any luck enabling S/PDIF output with the Conexant 20561/ctx5051? On my Lenovo W500 I see a digital output device, I can play on it but I can't hear any sound on my receiver. There was a patch by Takashi earlier this year to correct a wrong nodeID (03b515c45e2a93331d2ac9dc1c16440f7fa313ec, ALSA: hda - Fix SPDIF output widget for Cxt5051 codec), so I figure some amount of testing has been done already? I wonder if this comes from a non-initialized pin configuration as reported in /proc/asound/card0/codec#0? Node 0x1c [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x12 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="Conexant Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Thanks - Pierre

2 4

Re: [alsa-devel] ALSA 1.0.23 Kernel Panic in snd_hda_codec_idt:stac92xx_add_jack
by Robert Krakora 06 May '10

06 May '10

All: I am seeing a kernel panic with ALSA Drivers 1.0.23 in snd_hda_codec_idt:stac92xx_add_jack on my Dell Studio 1737 running stock CentOS 5.4. Before I file a bug report I just wanted to make sure that you did not already have a patch for this problem. Please see the attached .jpg for the trace. I took a picture as this is a hard failure with a backtrace (keyboard blinks). Best Regards, -- Rob Krakora Senior Software Engineer MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext. 206 (317)663-0808 Fax

2 2

[alsa-devel] [GIT PULL} ASoC updates for 2.6.35
by Liam Girdwood 06 May '10

06 May '10

The following changes since commit cb7b76961f73e4ae934d44f7b2e7ba974442f2fe: Takashi Iwai (1): Merge branch 'for-2.6.35' of git://git.kernel.org/.../lrg/asoc-2.6 into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.35 Jarkko Nikula (2): ASoC: omap: Add basic audio support for Nokia RX-51/N900 ASoC: tlv320aic3x: Add platform data and reset gpio handling Liam Girdwood (3): ASoC: pandora - update DAPM pins ASoC: zoom2 - update DAPM pins ASoC: tlv320dac33 - disable regulators at i2c remove() Peter Ujfalusi (9): ASoC: TWL4030: AIF/APLL fix in DAPM domain ASoC: TWL4030: Remove OUTL/R outputs ASoC: tlv320dac33: Optimize power up, and restore ASoC: tlv320dac33: Revised module loading, and DAC33 ID read ASoC: tlv320dac33: Manage a pointer for snd_pcm_substream in private structure ASoC: tlv320dac33: Support for turning off the codec ASoC: tpa6130a2: TLV mapping for tpa6140a2 ASoC: tpa6130a2: Support for limiting gain ASoC: tlv320dac33: Use codec defaults for LOM/LOP and DAC power include/sound/tlv320aic3x.h | 17 +++ include/sound/tpa6130a2-plat.h | 1 + sound/soc/codecs/tlv320aic3x.c | 25 ++++ sound/soc/codecs/tlv320dac33.c | 223 ++++++++++++++++-------------- sound/soc/codecs/tpa6130a2.c | 99 +++++++++++++- sound/soc/codecs/twl4030.c | 86 ++++++++---- sound/soc/omap/Kconfig | 10 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3pandora.c | 2 - sound/soc/omap/rx51.c | 294 ++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/zoom2.c | 3 - 11 files changed, 619 insertions(+), 143 deletions(-) create mode 100644 include/sound/tlv320aic3x.h create mode 100644 sound/soc/omap/rx51.c

2 1

[alsa-devel] [PATCHv2] ASoC: tlv320aic3x: Add platform data and reset gpio handling
by Jarkko Nikula 06 May '10

06 May '10

Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- v2: - Removed added regulator_bulk_disable brain fart from aic3x_i2c_probe - Added comment to gpio_reset number in aic3x_pdata --- include/sound/tlv320aic3x.h | 17 +++++++++++++++++ sound/soc/codecs/tlv320aic3x.c | 25 +++++++++++++++++++++++++ 2 files changed, 42 insertions(+), 0 deletions(-) create mode 100644 include/sound/tlv320aic3x.h diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h new file mode 100644 index 0000000..b1a5f34 --- /dev/null +++ b/include/sound/tlv320aic3x.h @@ -0,0 +1,17 @@ +/* + * Platform data for Texas Instruments TLV320AIC3x codec + * + * Author: Jarkko Nikula <jhnikula(a)gmail.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. + */ +#ifndef __TLV320AIC3x_H__ +#define __TLV320AIC3x_H__ + +struct aic3x_pdata { + int gpio_reset; /* < 0 if not used */ +}; + +#endif \ No newline at end of file diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b2c2794..435adf1 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -38,6 +38,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/gpio.h> #include <linux/regulator/consumer.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -48,6 +49,7 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> +#include <sound/tlv320aic3x.h> #include "tlv320aic3x.h" @@ -65,6 +67,7 @@ struct aic3x_priv { struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES]; unsigned int sysclk; int master; + int gpio_reset; }; /* @@ -1279,6 +1282,10 @@ static int aic3x_unregister(struct aic3x_priv *aic3x) snd_soc_unregister_dai(&aic3x_dai); snd_soc_unregister_codec(&aic3x->codec); + if (aic3x->gpio_reset >= 0) { + gpio_set_value(aic3x->gpio_reset, 0); + gpio_free(aic3x->gpio_reset); + } regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); @@ -1303,6 +1310,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, { struct snd_soc_codec *codec; struct aic3x_priv *aic3x; + struct aic3x_pdata *pdata = i2c->dev.platform_data; int ret, i; aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); @@ -1319,6 +1327,15 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, aic3x); + aic3x->gpio_reset = -1; + if (pdata && pdata->gpio_reset >= 0) { + ret = gpio_request(pdata->gpio_reset, "tlv320aic3x reset"); + if (ret != 0) + goto err_gpio; + aic3x->gpio_reset = pdata->gpio_reset; + gpio_direction_output(aic3x->gpio_reset, 0); + } + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) aic3x->supplies[i].supply = aic3x_supply_names[i]; @@ -1336,11 +1353,19 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, goto err_enable; } + if (aic3x->gpio_reset >= 0) { + udelay(1); + gpio_set_value(aic3x->gpio_reset, 1); + } + return aic3x_register(codec); err_enable: regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); err_get: + if (aic3x->gpio_reset >= 0) + gpio_free(aic3x->gpio_reset); +err_gpio: kfree(aic3x); return ret; } -- 1.7.0

4 8

[alsa-devel] [PATCH] ASoC: tlv320dac33: Use codec defaults for LOM/LOP and DAC power
by Peter Ujfalusi 06 May '10

06 May '10

Do not change the codec defaults for the following registers: 0x40, 0x41: Line output gains, do not use amplification 0x42: LOM/LOP Voltage hold, and selection 0x44: LOM inversion control It has been found, that the values configured to these registers can cause amplification, which can make the output of DAC33 distorted. The codec reset values are considered safe in all environmnts. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- If there is a need for user control over these, than additional kcontrols can be invented for the settings. But sorting out the meaning, and naming of such controls could take considerable time. At the moment, it is safer to use the codec defaults for these instead of the original values. sound/soc/codecs/tlv320dac33.c | 11 ----------- 1 files changed, 0 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 68b7ccb..ad5e263 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -303,21 +303,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec) /* C : (defaults) */ dac33_write(codec, DAC33_DAC_CTRL_C, 0x00); - /* 64-65 : L&R DAC power control - Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/ - dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); - dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); - /* 73 : volume soft stepping control, clock source = internal osc (?) */ dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN); - /* 66 : LOP/LOM Modes */ - dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff); - - /* 68 : LOM inverted from LOP */ - dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2)); - dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB); /* Restore only selected registers (gains mostly) */ -- 1.7.1

3 2

[alsa-devel] [PATCHv2] ASoC: omap: Add basic audio support for Nokia RX-51/N900
by Jarkko Nikula 06 May '10

06 May '10

This patch adds support for integrated stereo speakers and digital microphone found on Nokia RX-51 hardware. This is a cut down version based on Maemo kernel sources and earlier patchset by Eduardo Valentin et al. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.ht… Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Eduardo Valentin <eduardo.valentin(a)nokia.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- This is for sound-2.6.git. This is independent and boot safe patch but audio codec doesn't probe without a few patches to arch/arm/mach-omap2/board-rx51-peripherals.c. See http://marc.info/?l=linux-omap&m=127290212826770&w=2 Some audio features would need more work and some, even being trivial to add, need more testing are there support for them in mainline. Thus this heavily simplified and little bit cleaned up version. --- sound/soc/omap/Kconfig | 10 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/rx51.c | 294 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 306 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/rx51.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index f11963c..83be4a7 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -18,6 +18,16 @@ config SND_OMAP_SOC_N810 help Say Y if you want to add support for SoC audio on Nokia N810. +config SND_OMAP_SOC_RX51 + tristate "SoC Audio support for Nokia RX-51" + depends on SND_OMAP_SOC && MACH_NOKIA_RX51 + select OMAP_MCBSP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on Nokia RX-51 + hardware. This is also known as Nokia N900 product. + config SND_OMAP_SOC_AMS_DELTA tristate "SoC Audio support for Amstrad E3 (Delta) videophone" depends on SND_OMAP_SOC && MACH_AMS_DELTA diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0bc00ca..3a75755 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -9,6 +9,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-rx51-objs := rx51.o snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o @@ -22,6 +23,7 @@ snd-soc-zoom2-objs := zoom2.o snd-soc-igep0020-objs := igep0020.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c new file mode 100644 index 0000000..47d831e --- /dev/null +++ b/sound/soc/omap/rx51.c @@ -0,0 +1,294 @@ +/* + * rx51.c -- SoC audio for Nokia RX-51 + * + * Copyright (C) 2008 - 2009 Nokia Corporation + * + * Contact: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * Eduardo Valentin <eduardo.valentin(a)nokia.com> + * Jarkko Nikula <jhnikula(a)gmail.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/delay.h> +#include <linux/gpio.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 "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic3x.h" + +/* + * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This + * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c + */ +#define RX51_SPEAKER_AMP_TWL_GPIO (192 + 7) + +static int rx51_spk_func; +static int rx51_dmic_func; + +static void rx51_ext_control(struct snd_soc_codec *codec) +{ + if (rx51_spk_func) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); + if (rx51_dmic_func) + snd_soc_dapm_enable_pin(codec, "DMic"); + else + snd_soc_dapm_disable_pin(codec, "DMic"); + + snd_soc_dapm_sync(codec); +} + +static int rx51_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->card->codec; + + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); + rx51_ext_control(codec); + + return 0; +} + +static int rx51_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 err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) + return err; + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) + return err; + + /* Set the codec system clock for DAC and ADC */ + return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000, + SND_SOC_CLOCK_IN); +} + +static struct snd_soc_ops rx51_ops = { + .startup = rx51_startup, + .hw_params = rx51_hw_params, +}; + +static int rx51_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = rx51_spk_func; + + return 0; +} + +static int rx51_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (rx51_spk_func == ucontrol->value.integer.value[0]) + return 0; + + rx51_spk_func = ucontrol->value.integer.value[0]; + rx51_ext_control(codec); + + return 1; +} + +static int rx51_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 1); + else + gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 0); + + return 0; +} + +static int rx51_get_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = rx51_dmic_func; + + return 0; +} + +static int rx51_set_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (rx51_dmic_func == ucontrol->value.integer.value[0]) + return 0; + + rx51_dmic_func = ucontrol->value.integer.value[0]; + rx51_ext_control(codec); + + return 1; +} + +static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), + SND_SOC_DAPM_MIC("DMic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Ext Spk", NULL, "HPLOUT"}, + {"Ext Spk", NULL, "HPROUT"}, + + {"DMic Rate 64", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "DMic"}, +}; + +static const char *spk_function[] = {"Off", "On"}; +static const char *input_function[] = {"ADC", "Digital Mic"}; + +static const struct soc_enum rx51_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function), +}; + +static const struct snd_kcontrol_new aic34_rx51_controls[] = { + SOC_ENUM_EXT("Speaker Function", rx51_enum[0], + rx51_get_spk, rx51_set_spk), + SOC_ENUM_EXT("Input Select", rx51_enum[1], + rx51_get_input, rx51_set_input), +}; + +static int rx51_aic34_init(struct snd_soc_codec *codec) +{ + int err; + + /* Set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "MIC3L"); + snd_soc_dapm_nc_pin(codec, "MIC3R"); + snd_soc_dapm_nc_pin(codec, "LINE1R"); + + /* Add RX-51 specific controls */ + err = snd_soc_add_controls(codec, aic34_rx51_controls, + ARRAY_SIZE(aic34_rx51_controls)); + if (err < 0) + return err; + + /* Add RX-51 specific widgets */ + snd_soc_dapm_new_controls(codec, aic34_dapm_widgets, + ARRAY_SIZE(aic34_dapm_widgets)); + + /* Set up RX-51 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link rx51_dai[] = { + { + .name = "TLV320AIC34", + .stream_name = "AIC34", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &aic3x_dai, + .init = rx51_aic34_init, + .ops = &rx51_ops, + }, +}; + +/* Audio private data */ +static struct aic3x_setup_data rx51_aic34_setup = { + .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED, + .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT, +}; + +/* Audio card */ +static struct snd_soc_card rx51_sound_card = { + .name = "RX-51", + .dai_link = rx51_dai, + .num_links = ARRAY_SIZE(rx51_dai), + .platform = &omap_soc_platform, +}; + +/* Audio subsystem */ +static struct snd_soc_device rx51_snd_devdata = { + .card = &rx51_sound_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &rx51_aic34_setup, +}; + +static struct platform_device *rx51_snd_device; + +static int __init rx51_soc_init(void) +{ + int err; + + if (!machine_is_nokia_rx51()) + return -ENODEV; + + rx51_snd_device = platform_device_alloc("soc-audio", -1); + if (!rx51_snd_device) { + err = -ENOMEM; + goto err1; + } + + platform_set_drvdata(rx51_snd_device, &rx51_snd_devdata); + rx51_snd_devdata.dev = &rx51_snd_device->dev; + *(unsigned int *)rx51_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ + + err = platform_device_add(rx51_snd_device); + if (err) + goto err2; + + return 0; +err2: + platform_device_put(rx51_snd_device); +err1: + + return err; +} + +static void __exit rx51_soc_exit(void) +{ + platform_device_unregister(rx51_snd_device); +} + +module_init(rx51_soc_init); +module_exit(rx51_soc_exit); + +MODULE_AUTHOR("Nokia Corporation"); +MODULE_DESCRIPTION("ALSA SoC Nokia RX-51"); +MODULE_LICENSE("GPL"); -- 1.7.0

5 4

[alsa-devel] [PATCH] es1688: allocate snd_es1688 structure as a part of snd_card structure
by Krzysztof Helt 06 May '10

06 May '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Allocate the snd_es1688 during the snd_card allocation. This allows to remove the card pointer from the snd_es1688 structure. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the first part of changes to merge the es968 and the es1688 drivers. include/sound/es1688.h | 10 ++++---- sound/isa/es1688/es1688.c | 23 ++++++++++++--------- sound/isa/es1688/es1688_lib.c | 43 +++++++++++++--------------------------- sound/isa/gus/gusextreme.c | 26 +++++++++++++----------- 4 files changed, 46 insertions(+), 56 deletions(-) diff --git a/include/sound/es1688.h b/include/sound/es1688.h index 10fcf14..4c29572 100644 --- a/include/sound/es1688.h +++ b/include/sound/es1688.h @@ -44,7 +44,6 @@ struct snd_es1688 { unsigned char pad; unsigned int dma_size; - struct snd_card *card; struct snd_pcm *pcm; struct snd_pcm_substream *playback_substream; struct snd_pcm_substream *capture_substream; @@ -108,14 +107,15 @@ struct snd_es1688 { void snd_es1688_mixer_write(struct snd_es1688 *chip, unsigned char reg, unsigned char data); int snd_es1688_create(struct snd_card *card, + struct snd_es1688 *chip, unsigned long port, unsigned long mpu_port, int irq, int mpu_irq, int dma8, - unsigned short hardware, - struct snd_es1688 ** rchip); -int snd_es1688_pcm(struct snd_es1688 *chip, int device, struct snd_pcm ** rpcm); -int snd_es1688_mixer(struct snd_es1688 *chip); + unsigned short hardware); +int snd_es1688_pcm(struct snd_card *card, struct snd_es1688 *chip, int device, + struct snd_pcm **rpcm); +int snd_es1688_mixer(struct snd_card *card, struct snd_es1688 *chip); #endif /* __SOUND_ES1688_H */ diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 07df201..2816794 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -79,8 +79,8 @@ static int __devinit snd_es1688_match(struct device *dev, unsigned int n) return enable[n]; } -static int __devinit snd_es1688_legacy_create(struct snd_card *card, - struct device *dev, unsigned int n, struct snd_es1688 **rchip) +static int __devinit snd_es1688_legacy_create(struct snd_card *card, + struct snd_es1688 *chip, struct device *dev, unsigned int n) { static long possible_ports[] = {0x220, 0x240, 0x260}; static int possible_irqs[] = {5, 9, 10, 7, -1}; @@ -104,14 +104,14 @@ static int __devinit snd_es1688_legacy_create(struct snd_card *card, } if (port[n] != SNDRV_AUTO_PORT) - return snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip); + return snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_AUTO); i = 0; do { port[n] = possible_ports[i]; - error = snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip); + error = snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_AUTO); } while (error < 0 && ++i < ARRAY_SIZE(possible_ports)); return error; @@ -125,19 +125,22 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) struct snd_pcm *pcm; int error; - error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + error = snd_card_create(index[n], id[n], THIS_MODULE, + sizeof(struct snd_es1688), &card); if (error < 0) return error; - error = snd_es1688_legacy_create(card, dev, n, &chip); + chip = card->private_data; + + error = snd_es1688_legacy_create(card, chip, dev, n); if (error < 0) goto out; - error = snd_es1688_pcm(chip, 0, &pcm); + error = snd_es1688_pcm(card, chip, 0, &pcm); if (error < 0) goto out; - error = snd_es1688_mixer(chip); + error = snd_es1688_mixer(card, chip); if (error < 0) goto out; diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index c76bb00..80c11fd 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -638,23 +638,20 @@ static const char *snd_es1688_chip_id(struct snd_es1688 *chip) } int snd_es1688_create(struct snd_card *card, + struct snd_es1688 *chip, unsigned long port, unsigned long mpu_port, int irq, int mpu_irq, int dma8, - unsigned short hardware, - struct snd_es1688 **rchip) + unsigned short hardware) { static struct snd_device_ops ops = { .dev_free = snd_es1688_dev_free, }; - struct snd_es1688 *chip; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (chip == NULL) return -ENOMEM; chip->irq = -1; @@ -662,25 +659,21 @@ int snd_es1688_create(struct snd_card *card, if ((chip->res_port = request_region(port + 4, 12, "ES1688")) == NULL) { snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4); - snd_es1688_free(chip); return -EBUSY; } if (request_irq(irq, snd_es1688_interrupt, IRQF_DISABLED, "ES1688", (void *) chip)) { snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq); - snd_es1688_free(chip); return -EBUSY; } chip->irq = irq; if (request_dma(dma8, "ES1688")) { snd_printk(KERN_ERR "es1688: can't grab DMA8 %d\n", dma8); - snd_es1688_free(chip); return -EBUSY; } chip->dma8 = dma8; spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); - chip->card = card; chip->port = port; mpu_port &= ~0x000f; if (mpu_port < 0x300 || mpu_port > 0x330) @@ -689,23 +682,16 @@ int snd_es1688_create(struct snd_card *card, chip->mpu_irq = mpu_irq; chip->hardware = hardware; - if ((err = snd_es1688_probe(chip)) < 0) { - snd_es1688_free(chip); + err = snd_es1688_probe(chip); + if (err < 0) return err; - } - if ((err = snd_es1688_init(chip, 1)) < 0) { - snd_es1688_free(chip); - return err; - } - /* Register device */ - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_es1688_free(chip); + err = snd_es1688_init(chip, 1); + if (err < 0) return err; - } - *rchip = chip; - return 0; + /* Register device */ + return snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); } static struct snd_pcm_ops snd_es1688_playback_ops = { @@ -730,12 +716,14 @@ static struct snd_pcm_ops snd_es1688_capture_ops = { .pointer = snd_es1688_capture_pointer, }; -int snd_es1688_pcm(struct snd_es1688 * chip, int device, struct snd_pcm ** rpcm) +int snd_es1688_pcm(struct snd_card *card, struct snd_es1688 *chip, + int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "ESx688", device, 1, 1, &pcm)) < 0) + err = snd_pcm_new(card, "ESx688", device, 1, 1, &pcm); + if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_es1688_playback_ops); @@ -1009,18 +997,15 @@ static unsigned char snd_es1688_init_table[][2] = { { ES1688_REC_DEV, 0x17 } }; -int snd_es1688_mixer(struct snd_es1688 *chip) +int snd_es1688_mixer(struct snd_card *card, struct snd_es1688 *chip) { - struct snd_card *card; unsigned int idx; int err; unsigned char reg, val; - if (snd_BUG_ON(!chip || !chip->card)) + if (snd_BUG_ON(!chip || !card)) return -EINVAL; - card = chip->card; - strcpy(card->mixername, snd_es1688_chip_id(chip)); for (idx = 0; idx < ARRAY_SIZE(snd_es1688_controls); idx++) { diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 65e4b18..008e8e5 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -95,7 +95,7 @@ static int __devinit snd_gusextreme_match(struct device *dev, unsigned int n) } static int __devinit snd_gusextreme_es1688_create(struct snd_card *card, - struct device *dev, unsigned int n, struct snd_es1688 **rchip) + struct snd_es1688 *chip, struct device *dev, unsigned int n) { static long possible_ports[] = {0x220, 0x240, 0x260}; static int possible_irqs[] = {5, 9, 10, 7, -1}; @@ -119,14 +119,14 @@ static int __devinit snd_gusextreme_es1688_create(struct snd_card *card, } if (port[n] != SNDRV_AUTO_PORT) - return snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_1688, rchip); + return snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_1688); i = 0; do { port[n] = possible_ports[i]; - error = snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_1688, rchip); + error = snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_1688); } while (error < 0 && ++i < ARRAY_SIZE(possible_ports)); return error; @@ -206,9 +206,8 @@ static int __devinit snd_gusextreme_detect(struct snd_gus_card *gus, return 0; } -static int __devinit snd_gusextreme_mixer(struct snd_es1688 *chip) +static int __devinit snd_gusextreme_mixer(struct snd_card *card) { - struct snd_card *card = chip->card; struct snd_ctl_elem_id id1, id2; int error; @@ -241,17 +240,20 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) struct snd_opl3 *opl3; int error; - error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + error = snd_card_create(index[n], id[n], THIS_MODULE, + sizeof(struct snd_es1688), &card); if (error < 0) return error; + es1688 = card->private_data; + if (mpu_port[n] == SNDRV_AUTO_PORT) mpu_port[n] = 0; if (mpu_irq[n] > 15) mpu_irq[n] = -1; - error = snd_gusextreme_es1688_create(card, dev, n, &es1688); + error = snd_gusextreme_es1688_create(card, es1688, dev, n); if (error < 0) goto out; @@ -280,11 +282,11 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) } gus->codec_flag = 1; - error = snd_es1688_pcm(es1688, 0, NULL); + error = snd_es1688_pcm(card, es1688, 0, NULL); if (error < 0) goto out; - error = snd_es1688_mixer(es1688); + error = snd_es1688_mixer(card, es1688); if (error < 0) goto out; @@ -300,7 +302,7 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) if (error < 0) goto out; - error = snd_gusextreme_mixer(es1688); + error = snd_gusextreme_mixer(card); if (error < 0) goto out; -- 1.6.4 -------------------------------------------------- Odkryj tajemnice TAROTA! Sprawdz >> http://link.interia.pl/f2695

2 2

[alsa-devel] [PATCH 1/2] powerpc/8610: add probing for individual DMA channels, not just DMA controllers
by Timur Tabi 05 May '10

05 May '10

A future version of the MPC8610 HPCD's ASoC DMA driver will probe on individual DMA channel nodes, so the DMA controller nodes' compatible string must be listed in mpc8610_ids[] for the probe to work. Also remove the "gianfar" compatible from mpc8610_ids[], since there is no gianfar (or any other networking device) on the 8610. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Kumar, the ASoC mainters are willing to pick up this patch, but they want an ACK from you first. Or, you could pick it up, since by itself it's harmless. arch/powerpc/platforms/86xx/mpc8610_hpcd.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c index 5abe137..018cc67 100644 --- a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c +++ b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c @@ -83,7 +83,8 @@ static struct of_device_id __initdata mpc8610_ids[] = { { .compatible = "fsl,mpc8610-immr", }, { .compatible = "fsl,mpc8610-guts", }, { .compatible = "simple-bus", }, - { .compatible = "gianfar", }, + /* So that the DMA channel nodes can be probed individually: */ + { .compatible = "fsl,eloplus-dma", }, {} }; -- 1.6.5

3 8

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Add platform data and reset gpio handling
by Jarkko Nikula 05 May '10

05 May '10

Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- include/sound/tlv320aic3x.h | 17 +++++++++++++++++ sound/soc/codecs/tlv320aic3x.c | 26 ++++++++++++++++++++++++++ 2 files changed, 43 insertions(+), 0 deletions(-) create mode 100644 include/sound/tlv320aic3x.h diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h new file mode 100644 index 0000000..e493a14 --- /dev/null +++ b/include/sound/tlv320aic3x.h @@ -0,0 +1,17 @@ +/* + * Platform data for Texas Instruments TLV320AIC3x codec + * + * Author: Jarkko Nikula <jhnikula(a)gmail.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. + */ +#ifndef __TLV320AIC3x_H__ +#define __TLV320AIC3x_H__ + +struct aic3x_pdata { + int gpio_reset; +}; + +#endif \ No newline at end of file diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b2c2794..cdcfb6b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -38,6 +38,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/gpio.h> #include <linux/regulator/consumer.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -48,6 +49,7 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> +#include <sound/tlv320aic3x.h> #include "tlv320aic3x.h" @@ -65,6 +67,7 @@ struct aic3x_priv { struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES]; unsigned int sysclk; int master; + int gpio_reset; }; /* @@ -1279,6 +1282,10 @@ static int aic3x_unregister(struct aic3x_priv *aic3x) snd_soc_unregister_dai(&aic3x_dai); snd_soc_unregister_codec(&aic3x->codec); + if (aic3x->gpio_reset >= 0) { + gpio_set_value(aic3x->gpio_reset, 0); + gpio_free(aic3x->gpio_reset); + } regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); @@ -1303,6 +1310,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, { struct snd_soc_codec *codec; struct aic3x_priv *aic3x; + struct aic3x_pdata *pdata = i2c->dev.platform_data; int ret, i; aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); @@ -1319,6 +1327,15 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, aic3x); + aic3x->gpio_reset = -1; + if (pdata && pdata->gpio_reset >= 0) { + ret = gpio_request(pdata->gpio_reset, "tlv320aic3x reset"); + if (ret != 0) + goto err_gpio; + aic3x->gpio_reset = pdata->gpio_reset; + gpio_direction_output(aic3x->gpio_reset, 0); + } + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) aic3x->supplies[i].supply = aic3x_supply_names[i]; @@ -1336,11 +1353,20 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, goto err_enable; } + if (aic3x->gpio_reset >= 0) { + udelay(1); + gpio_set_value(aic3x->gpio_reset, 1); + } + return aic3x_register(codec); err_enable: regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); err_get: + regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); + if (aic3x->gpio_reset >= 0) + gpio_free(aic3x->gpio_reset); +err_gpio: kfree(aic3x); return ret; } -- 1.7.0

2 2

[alsa-devel] [PATCH] ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite P500-PSPGSC-01800T
by Daniel Chen 05 May '10

05 May '10

Hi, Attached is another patch against sound-2.6 master HEAD (based on the previous patch from yesterday) that makes the OR's machine have audible sound by default. Please apply. Best, -Dan

2 1

[alsa-devel] [PATCH] ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite Pro T130-15F
by Daniel Chen 05 May '10

05 May '10

Hi, Attached is a patch against sound-2.6 master HEAD that makes the OR's machine have audible sound by default. Please apply. Best, -Dan

2 1

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 05 May '10

05 May '10

The following changes since commit cb7b76961f73e4ae934d44f7b2e7ba974442f2fe: Takashi Iwai (1): Merge branch 'for-2.6.35' of git://git.kernel.org/.../lrg/asoc-2.6 into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Mark Brown (2): ASoC: Remove redundant WM8960 SYSCLKSEL clkdiv option ASoC: Add WM9090 amplifier driver include/sound/wm9090.h | 28 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8960.c | 4 - sound/soc/codecs/wm8960.h | 1 - sound/soc/codecs/wm9090.c | 773 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9090.h | 715 +++++++++++++++++++++++++++++++++++++++++ 7 files changed, 1522 insertions(+), 5 deletions(-) create mode 100644 include/sound/wm9090.h create mode 100644 sound/soc/codecs/wm9090.c create mode 100644 sound/soc/codecs/wm9090.h

2 1

[alsa-devel] [PATCH RFC] ALSA: hda - Add quirk for Dell Inspiron 19T using a Conexant CX20582
by Anisse Astier 05 May '10

05 May '10

Add a quirk for all-in-one computer Dell Inspiron One 19 Touch to have proper HP and Mic support. Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- Hi, I have a Dell 19T all-in-one computer, which has audio problems by default: - speakers works OK - no jack detection, no audio in HP - internal or external Mic not working If I specify model=ideapad, the problems go away. But Mic boost appears as an "Analog Mic Boost Capture Enum" between 0 and 40dB instead of a normal bar control in alsamixer. I also tested model=dell-vostro, laptop, and dell-laptop, and they give worse results. There's no BIOS update available from vendor ATM. I'm using kernel tree sound-2.6 on branch master. I'm attaching alsa-info.sh output with and without model=ideapad in addition to the patch. Regards, Anisse sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 509a104..8446273 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2967,6 +2967,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), + SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), {} -- 1.7.0.6

2 1

[alsa-devel] sound/core/timer.c: inconsistent lock state reported for tu->qlock
by Stefan Richter 05 May '10

05 May '10

Hi, I got the following log from a rarely used PC that I just updated from 2.6.34-rc4 to 2.6.34-rc5. I don't know though under which kernel I last used the soundcard of it; could have been 2.6.33, could have been a .34-rc. It is an x86 Mac mini with 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Kernel is vanilla except for various firewire patches, which should not matter to this issue. ================================= [ INFO: inconsistent lock state ] 2.6.34-rc5 #5 --------------------------------- inconsistent {HARDIRQ-ON-W} -> {IN-HARDIRQ-W} usage. dolphin/4003 [HC1[1]:SC0[0]:HE0:SE1] takes: (&(&tu->qlock)->rlock){?.+...}, at: [<f84ec472>] snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] {HARDIRQ-ON-W} state was registered at: [<c1048de9>] __lock_acquire+0x654/0x1482 [<c1049c73>] lock_acquire+0x5c/0x73 [<c125ac3e>] _raw_spin_lock+0x25/0x34 [<f84ec370>] snd_timer_user_ccallback+0x55/0x95 [snd_timer] [<f84ecc4b>] snd_timer_notify1+0x53/0xca [snd_timer] [<f84ecdbe>] _snd_timer_stop+0xfc/0x106 [snd_timer] [<f84eceab>] snd_timer_stop+0x13/0x42 [snd_timer] [<f84ee2dc>] snd_timer_user_ioctl+0xa27/0xaf5 [snd_timer] [<c1092c2f>] vfs_ioctl+0x22/0x87 [<c10931aa>] do_vfs_ioctl+0x472/0x4ac [<c1093210>] sys_ioctl+0x2c/0x45 [<c1002850>] sysenter_do_call+0x12/0x36 irq event stamp: 2042234 hardirqs last enabled at (2042233): [<c1002883>] sysenter_exit+0xf/0x18 hardirqs last disabled at (2042234): [<c1002d67>] common_interrupt+0x27/0x34 softirqs last enabled at (2042226): [<c102cfc3>] __do_softirq+0x102/0x10a softirqs last disabled at (2042221): [<c102cff6>] do_softirq+0x2b/0x43 other info that might help us debug this: 2 locks held by dolphin/4003: #0: (snd_pcm_link_rwlock){.-....}, at: [<f8eb0a35>] snd_pcm_period_elapsed+0x2a/0xaf [snd_pcm] #1: (&(&substream->self_group.lock)->rlock){-.....}, at: [<f8eb0a3f>] snd_pcm_period_elapsed+0x34/0xaf [snd_pcm] stack backtrace: Pid: 4003, comm: dolphin Tainted: G W 2.6.34-rc5 #5 Call Trace: [<c125882d>] ? printk+0xf/0x12 [<c104761b>] print_usage_bug+0x14e/0x15a [<c10478c8>] mark_lock+0x2a1/0x4a7 [<c1048066>] ? check_usage_forwards+0x0/0xb0 [<c1048d69>] __lock_acquire+0x5d4/0x1482 [<c1049c08>] ? __lock_acquire+0x1473/0x1482 [<c1049c73>] lock_acquire+0x5c/0x73 [<f84ec472>] ? snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] [<c125ac3e>] _raw_spin_lock+0x25/0x34 [<f84ec472>] ? snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] [<f84ec472>] snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] [<f84ec44a>] ? snd_timer_user_tinterrupt+0x0/0x132 [snd_timer] [<f84ed2e1>] snd_timer_interrupt+0x213/0x264 [snd_timer] [<c125ac45>] ? _raw_spin_lock+0x2c/0x34 [<f8eb0a7d>] snd_pcm_period_elapsed+0x72/0xaf [snd_pcm] [<f8f733bb>] azx_interrupt+0x8d/0x109 [snd_hda_intel] [<c1057fb0>] handle_IRQ_event+0x1d/0xa2 [<c105980b>] handle_fasteoi_irq+0x79/0xb2 [<c10046ff>] handle_irq+0x3b/0x48 [<c1003f3d>] do_IRQ+0x45/0x9f [<c1002d6e>] common_interrupt+0x2e/0x34 [<c1250000>] ? piix_init_one+0x1d2/0x628 -- Stefan Richter -=====-==-=- -=-- ==-== http://arcgraph.de/sr/

3 2

[alsa-devel] [PATCH] ASoC: omap: Add basic audio support for Nokia RX-51/N900
by Jarkko Nikula 05 May '10

05 May '10

This patch adds support for integrated stereo speakers and digital microphone found on Nokia RX-51 hardware. This is a cut down version based on Maemo kernel sources and earlier patchset by Eduardo Valentin et al. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.ht… Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Eduardo Valentin <eduardo.valentin(a)nokia.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- This is for sound-2.6.git. This is independent and boot safe patch but audio codec doesn't probe without a few patches to arch/arm/mach-omap2/board-rx51-peripherals.c. See http://marc.info/?l=linux-omap&m=127290212826770&w=2 Some audio features would need more work and some, even being trivial to add, need more testing are there support for them in mainline. Thus this heavily simplified and little bit cleaned up version. --- sound/soc/omap/Kconfig | 10 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/rx51.c | 305 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 317 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/rx51.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index f11963c..83be4a7 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -18,6 +18,16 @@ config SND_OMAP_SOC_N810 help Say Y if you want to add support for SoC audio on Nokia N810. +config SND_OMAP_SOC_RX51 + tristate "SoC Audio support for Nokia RX-51" + depends on SND_OMAP_SOC && MACH_NOKIA_RX51 + select OMAP_MCBSP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on Nokia RX-51 + hardware. This is also known as Nokia N900 product. + config SND_OMAP_SOC_AMS_DELTA tristate "SoC Audio support for Amstrad E3 (Delta) videophone" depends on SND_OMAP_SOC && MACH_AMS_DELTA diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0bc00ca..3a75755 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -9,6 +9,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-rx51-objs := rx51.o snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o @@ -22,6 +23,7 @@ snd-soc-zoom2-objs := zoom2.o snd-soc-igep0020-objs := igep0020.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c new file mode 100644 index 0000000..425362b --- /dev/null +++ b/sound/soc/omap/rx51.c @@ -0,0 +1,305 @@ +/* + * rx51.c -- SoC audio for Nokia RX-51 + * + * Copyright (C) 2008 - 2009 Nokia Corporation + * + * Contact: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * Eduardo Valentin <eduardo.valentin(a)nokia.com> + * Jarkko Nikula <jhnikula(a)gmail.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/delay.h> +#include <linux/gpio.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 "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic3x.h" + +#define RX51_CODEC_RESET_GPIO 60 +/* + * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This + * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c + */ +#define RX51_SPEAKER_AMP_TWL_GPIO (192 + 7) + +static int rx51_spk_func; +static int rx51_dmic_func; + +static void rx51_ext_control(struct snd_soc_codec *codec) +{ + if (rx51_spk_func) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); + if (rx51_dmic_func) + snd_soc_dapm_enable_pin(codec, "DMic"); + else + snd_soc_dapm_disable_pin(codec, "DMic"); + + snd_soc_dapm_sync(codec); +} + +static int rx51_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->card->codec; + + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); + rx51_ext_control(codec); + + return 0; +} + +static int rx51_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 err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) + return err; + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) + return err; + + /* Set the codec system clock for DAC and ADC */ + return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000, + SND_SOC_CLOCK_IN); +} + +static struct snd_soc_ops rx51_ops = { + .startup = rx51_startup, + .hw_params = rx51_hw_params, +}; + +static int rx51_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = rx51_spk_func; + + return 0; +} + +static int rx51_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (rx51_spk_func == ucontrol->value.integer.value[0]) + return 0; + + rx51_spk_func = ucontrol->value.integer.value[0]; + rx51_ext_control(codec); + + return 1; +} + +static int rx51_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 1); + else + gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 0); + + return 0; +} + +static int rx51_get_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = rx51_dmic_func; + + return 0; +} + +static int rx51_set_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (rx51_dmic_func == ucontrol->value.integer.value[0]) + return 0; + + rx51_dmic_func = ucontrol->value.integer.value[0]; + rx51_ext_control(codec); + + return 1; +} + +static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), + SND_SOC_DAPM_MIC("DMic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Ext Spk", NULL, "HPLOUT"}, + {"Ext Spk", NULL, "HPROUT"}, + + {"DMic Rate 64", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "DMic"}, +}; + +static const char *spk_function[] = {"Off", "On"}; +static const char *input_function[] = {"ADC", "Digital Mic"}; + +static const struct soc_enum rx51_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function), +}; + +static const struct snd_kcontrol_new aic34_rx51_controls[] = { + SOC_ENUM_EXT("Speaker Function", rx51_enum[0], + rx51_get_spk, rx51_set_spk), + SOC_ENUM_EXT("Input Select", rx51_enum[1], + rx51_get_input, rx51_set_input), +}; + +static int rx51_aic34_init(struct snd_soc_codec *codec) +{ + int err; + + /* Set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "MIC3L"); + snd_soc_dapm_nc_pin(codec, "MIC3R"); + snd_soc_dapm_nc_pin(codec, "LINE1R"); + + /* Add RX-51 specific controls */ + err = snd_soc_add_controls(codec, aic34_rx51_controls, + ARRAY_SIZE(aic34_rx51_controls)); + if (err < 0) + return err; + + /* Add RX-51 specific widgets */ + snd_soc_dapm_new_controls(codec, aic34_dapm_widgets, + ARRAY_SIZE(aic34_dapm_widgets)); + + /* Set up RX-51 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link rx51_dai[] = { + { + .name = "TLV320AIC34", + .stream_name = "AIC34", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &aic3x_dai, + .init = rx51_aic34_init, + .ops = &rx51_ops, + }, +}; + +/* Audio private data */ +static struct aic3x_setup_data rx51_aic34_setup = { + .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED, + .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT, +}; + +/* Audio card */ +static struct snd_soc_card rx51_sound_card = { + .name = "RX-51", + .dai_link = rx51_dai, + .num_links = ARRAY_SIZE(rx51_dai), + .platform = &omap_soc_platform, +}; + +/* Audio subsystem */ +static struct snd_soc_device rx51_snd_devdata = { + .card = &rx51_sound_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &rx51_aic34_setup, +}; + +static struct platform_device *rx51_snd_device; + +static int __init rx51_soc_init(void) +{ + int err; + + if (!machine_is_nokia_rx51()) + return -ENODEV; + + BUG_ON(gpio_request(RX51_CODEC_RESET_GPIO, NULL) < 0); + gpio_direction_output(RX51_CODEC_RESET_GPIO, 0); + + gpio_set_value(RX51_CODEC_RESET_GPIO, 0); + udelay(1); + gpio_set_value(RX51_CODEC_RESET_GPIO, 1); + msleep(1); + + rx51_snd_device = platform_device_alloc("soc-audio", -1); + if (!rx51_snd_device) { + err = -ENOMEM; + goto err1; + } + + platform_set_drvdata(rx51_snd_device, &rx51_snd_devdata); + rx51_snd_devdata.dev = &rx51_snd_device->dev; + *(unsigned int *)rx51_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ + + err = platform_device_add(rx51_snd_device); + if (err) + goto err2; + + return 0; +err2: + platform_device_put(rx51_snd_device); +err1: + gpio_free(RX51_CODEC_RESET_GPIO); + + return err; +} + +static void __exit rx51_soc_exit(void) +{ + platform_device_unregister(rx51_snd_device); + gpio_free(RX51_CODEC_RESET_GPIO); +} + +module_init(rx51_soc_init); +module_exit(rx51_soc_exit); + +MODULE_AUTHOR("Nokia Corporation"); +MODULE_DESCRIPTION("ALSA SoC Nokia RX-51"); +MODULE_LICENSE("GPL"); -- 1.7.0

2 3

[alsa-devel] timestamps
by Jonas Holmberg 04 May '10

04 May '10

How can I get the CLOCK_MONOTONIC timestamp for the first sample in the data I get from snd_pcm_readi()? /Jonas

2 1

[alsa-devel] [PATCH] ASoC: tpa6130a2: TLV mapping for tpa6140a2
by Peter Ujfalusi 04 May '10

04 May '10

Both tpa6130a2, and tpa6140a2 is supported by the same driver, but the gain dB scaling is different on the amplifiers. Provide different mixer control for the chips with correct TLV mapping. User space will see: "TPA6130A2 Headphone Playback Volume" in case of 6130 "TPA6140A2 Headphone Playback Volume" in case of 6140 The way machine drivers are using this amplifier remained the same. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tpa6130a2.c | 31 ++++++++++++++++++++++++++++--- 1 files changed, 28 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 958d49c..0cf3e38 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -53,6 +53,7 @@ struct tpa6130a2_data { struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES]; int power_gpio; unsigned char power_state; + enum tpa_model id; }; static int tpa6130a2_i2c_read(int reg) @@ -263,6 +264,20 @@ static const struct snd_kcontrol_new tpa6130a2_controls[] = { tpa6130_tlv), }; +static const unsigned int tpa6140_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 8, TLV_DB_SCALE_ITEM(-5900, 400, 0), + 9, 16, TLV_DB_SCALE_ITEM(-2500, 200, 0), + 17, 31, TLV_DB_SCALE_ITEM(-1000, 100, 0), +}; + +static const struct snd_kcontrol_new tpa6140a2_controls[] = { + SOC_SINGLE_EXT_TLV("TPA6140A2 Headphone Playback Volume", + TPA6130A2_REG_VOL_MUTE, 1, 0x1f, 0, + tpa6130a2_get_reg, tpa6130a2_set_reg, + tpa6140_tlv), +}; + /* * Enable or disable channel (left or right) * The bit number for mute and amplifier are the same per channel: @@ -368,13 +383,22 @@ static const struct snd_soc_dapm_route audio_map[] = { int tpa6130a2_add_controls(struct snd_soc_codec *codec) { + struct tpa6130a2_data *data; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets, ARRAY_SIZE(tpa6130a2_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - return snd_soc_add_controls(codec, tpa6130a2_controls, - ARRAY_SIZE(tpa6130a2_controls)); + if (data->id == TPA6140A2) + return snd_soc_add_controls(codec, tpa6140a2_controls, + ARRAY_SIZE(tpa6140a2_controls)); + else + return snd_soc_add_controls(codec, tpa6130a2_controls, + ARRAY_SIZE(tpa6130a2_controls)); } EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); @@ -407,6 +431,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, pdata = client->dev.platform_data; data->power_gpio = pdata->power_gpio; + data->id = pdata->id; mutex_init(&data->mutex); @@ -425,7 +450,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, gpio_direction_output(data->power_gpio, 0); } - switch (pdata->id) { + switch (data->id) { case TPA6130A2: for (i = 0; i < ARRAY_SIZE(data->supplies); i++) data->supplies[i].supply = tpa6130a2_supply_names[i]; -- 1.7.1

4 3

[alsa-devel] Playing multiple sounds at the same time
by Andreas Falkenhahn 04 May '10

04 May '10

Hi, I'm new to ALSA programming so please bear with me if the following question is all too obvious. I did some research on my question before but I didn't come across a clear answer. What I want to do is really simple: Playing multiple sounds in different formats at the same time. E.g. Sound 1: 12345 hz, 8-bit signed, mono Sound 2: 44100 hz, 16-bit unsigned, stereo Sound 3: 8000 hz, 8-bit unsigned, stereo Now I want to play sounds 1 to 3 at the same time. My question is now: Can ALSA do automatic mixing for me here or do I have to do the mixing on my own? In other words, can I simply feed all these three sound streams to ALSA or do I have to mix them into a fourth audio stream first and feeding this new fourth stream to ALSA then? Thanks, Andreas

3 4

[alsa-devel] safe support for rewind in ALSA
by pl bossart 04 May '10

04 May '10

All, I'd like to reopen a thread on concerns with the current implementation of rewind in ALSA. This was already discussed on the mailing list last year without any progress. (see http://thread.gmane.org/gmane.linux.alsa.devel/65759/focus=65773 and http://article.gmane.org/gmane.linux.alsa.devel/66902) The problem is that snd_pcm_rewind() allows in theory to go back to the next sample to be rendered. This however isn't possible on any existing hardware. DMAs do prefetch samples ahead of time, and output interfaces have FIFOs on top of this. This isn't too bad in the PC world with HDAudio, since the prefetching is limited. However in the embedded space it has become a headache. To reduce power consumption, most of the recent embedded interfaces do embed some local internal memory (5kB for TI's McBSP3, a lot more for others), and we should provide a means for driver developers to set a bound on how much can be rewound safely when power-saving features are enabled. Note that there already are a set of definitions for FIFOs, but they are either obsolete or no one uses them (actually HDAudio makes use of this value internally in azx_dev but doesn't export it). My suggestion would be to add a 'fifo' parameter to the runtime structure (so that it's visible to the core), let the driver developer fill the value in the .open routine, and use this parameter in rewind to prevent rewinding too much. This parameter would include both hardware FIFOs and DMA buffers. Comments welcome. - Pierre

7 41

[alsa-devel] asoc: fsl_dma_hw_free being called twice at end of playback
by Timur Tabi 04 May '10

04 May '10

In my 8610 ASoC driver, I'm seeing the DMA driver's .hw_free function being called twice whenever playback ends. I've added some printk's to demonstrate: snd_pcm_common_ioctl1:2543 snd_pcm_hw_free:543 soc_pcm_hw_free:761 c023983c fsl_dma_hw_free:720 substream=df029d00 dma_private=df440000 stream=0 snd_pcm_release_substream:2022 soc_pcm_hw_free:761 c023983c fsl_dma_hw_free:720 substream=df029d00 dma_private=df440000 stream=0 fsl_dma_close:759 substream=df029d00 dma_private=df440000 So the ioctl is being called only once, but the both snd_pcm_hw_free() and snd_pcm_release_substream() are calling the .hw_free function. Is this normal? I don't know how to handle this, other than to add some boolean to my private data structure that indicates whether the resources have been freed or not. -- Timur Tabi Linux kernel developer at Freescale

3 3

[alsa-devel] [PATCH 2/2] [v2] powerpc/alsa: update MPC8610 HPCD audio drivers to support multiple components
by Timur Tabi 04 May '10

04 May '10

Update the Freescale MPC8610 HPCD audio drivers to support the new multiple- component architecture in ASoC. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 299 ++++++----- sound/soc/fsl/fsl_dma.h | 19 +- sound/soc/fsl/fsl_ssi.c | 211 +++++---- sound/soc/fsl/fsl_ssi.h | 25 - sound/soc/fsl/mpc8610_hpcd.c | 1174 ++++++++++++++++++++---------------------- 5 files changed, 844 insertions(+), 884 deletions(-) rewrite sound/soc/fsl/mpc8610_hpcd.c (62%) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 857e9f7..ff0064c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -3,10 +3,11 @@ * * Author: Timur Tabi <timur(a)freescale.com> * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. * * This driver implements ASoC support for the Elo DMA controller, which is * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms, @@ -15,10 +16,11 @@ #include <linux/module.h> #include <linux/init.h> -#include <linux/platform_device.h> #include <linux/dma-mapping.h> #include <linux/interrupt.h> #include <linux/delay.h> +#include <linux/of_platform.h> +#include <linux/list.h> #include <sound/core.h> #include <sound/pcm.h> @@ -51,26 +53,16 @@ #define FSLDMA_PCM_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ SNDRV_PCM_RATE_CONTINUOUS) -/* DMA global data. This structure is used by fsl_dma_open() to determine - * which DMA channels to assign to a substream. Unfortunately, ASoC V1 does - * not allow the machine driver to provide this information to the PCM - * driver in advance, and there's no way to differentiate between the two - * DMA controllers. So for now, this driver only supports one SSI device - * using two DMA channels. We cannot support multiple DMA devices. - * - * ssi_stx_phys: bus address of SSI STX register - * ssi_srx_phys: bus address of SSI SRX register - * dma_channel: pointer to the DMA channel's registers - * irq: IRQ for this DMA channel - * assigned: set to 1 if that DMA channel is assigned to a substream - */ -static struct { +struct dma_object { + struct list_head list; + struct snd_soc_platform_driver dai; dma_addr_t ssi_stx_phys; dma_addr_t ssi_srx_phys; - struct ccsr_dma_channel __iomem *dma_channel[2]; - unsigned int irq[2]; - unsigned int assigned[2]; -} dma_global_data; + struct ccsr_dma_channel __iomem *channel; + unsigned int irq; + bool assigned; + char path[1]; +}; /* * The number of DMA links to use. Two is the bare minimum, but if you @@ -87,8 +79,6 @@ static struct { * structure. * * @link[]: array of link descriptors - * @controller_id: which DMA controller (0, 1, ...) - * @channel_id: which DMA channel on the controller (0, 1, 2, ...) * @dma_channel: pointer to the DMA channel's registers * @irq: IRQ for this DMA channel * @substream: pointer to the substream object, needed by the ISR @@ -103,8 +93,6 @@ static struct { */ struct fsl_dma_private { struct fsl_dma_link_descriptor link[NUM_DMA_LINKS]; - unsigned int controller_id; - unsigned int channel_id; struct ccsr_dma_channel __iomem *dma_channel; unsigned int irq; struct snd_pcm_substream *substream; @@ -211,6 +199,9 @@ static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private) static irqreturn_t fsl_dma_isr(int irq, void *dev_id) { struct fsl_dma_private *dma_private = dev_id; + struct snd_pcm_substream *substream = dma_private->substream; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; irqreturn_t ret = IRQ_NONE; u32 sr, sr2 = 0; @@ -221,11 +212,8 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) sr = in_be32(&dma_channel->sr); if (sr & CCSR_DMA_SR_TE) { - dev_err(dma_private->substream->pcm->card->dev, - "DMA transmit error (controller=%u channel=%u irq=%u\n", - dma_private->controller_id, - dma_private->channel_id, irq); - fsl_dma_abort_stream(dma_private->substream); + dev_err(dev, "dma transmit error\n"); + fsl_dma_abort_stream(substream); sr2 |= CCSR_DMA_SR_TE; ret = IRQ_HANDLED; } @@ -234,11 +222,8 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) ret = IRQ_HANDLED; if (sr & CCSR_DMA_SR_PE) { - dev_err(dma_private->substream->pcm->card->dev, - "DMA%u programming error (channel=%u irq=%u)\n", - dma_private->controller_id, - dma_private->channel_id, irq); - fsl_dma_abort_stream(dma_private->substream); + dev_err(dev, "dma programming error\n"); + fsl_dma_abort_stream(substream); sr2 |= CCSR_DMA_SR_PE; ret = IRQ_HANDLED; } @@ -252,8 +237,6 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) ret = IRQ_HANDLED; if (sr & CCSR_DMA_SR_EOSI) { - struct snd_pcm_substream *substream = dma_private->substream; - /* Tell ALSA we completed a period. */ snd_pcm_period_elapsed(substream); @@ -304,10 +287,8 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); if (ret) { - dev_err(card->dev, - "Can't allocate playback DMA buffer (size=%u)\n", - fsl_dma_hardware.buffer_bytes_max); - return -ENOMEM; + dev_err(card->dev, "can't allocate playback dma buffer\n"); + return ret; } ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, @@ -315,10 +296,8 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, &pcm->streams[1].substream->dma_buffer); if (ret) { snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); - dev_err(card->dev, - "Can't allocate capture DMA buffer (size=%u)\n", - fsl_dma_hardware.buffer_bytes_max); - return -ENOMEM; + dev_err(card->dev, "can't allocate capture dma buffer\n"); + return ret; } return 0; @@ -389,6 +368,10 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, static int fsl_dma_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; + struct dma_object *dma = + container_of(rtd->platform->driver, struct dma_object, dai); struct fsl_dma_private *dma_private; struct ccsr_dma_channel __iomem *dma_channel; dma_addr_t ld_buf_phys; @@ -406,52 +389,44 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { - dev_err(substream->pcm->card->dev, "invalid buffer size\n"); + dev_err(dev, "invalid buffer size\n"); return ret; } channel = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - if (dma_global_data.assigned[channel]) { - dev_err(substream->pcm->card->dev, - "DMA channel already assigned\n"); + if (dma->assigned) { + dev_err(dev, "dma channel already assigned\n"); return -EBUSY; } - dma_private = dma_alloc_coherent(substream->pcm->card->dev, - sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL); + dma_private = dma_alloc_coherent(dev, sizeof(struct fsl_dma_private), + &ld_buf_phys, GFP_KERNEL); if (!dma_private) { - dev_err(substream->pcm->card->dev, - "can't allocate DMA private data\n"); + dev_err(dev, "can't allocate dma private data\n"); return -ENOMEM; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_private->ssi_sxx_phys = dma_global_data.ssi_stx_phys; + dma_private->ssi_sxx_phys = dma->ssi_stx_phys; else - dma_private->ssi_sxx_phys = dma_global_data.ssi_srx_phys; + dma_private->ssi_sxx_phys = dma->ssi_srx_phys; - dma_private->dma_channel = dma_global_data.dma_channel[channel]; - dma_private->irq = dma_global_data.irq[channel]; + dma_private->dma_channel = dma->channel; + dma_private->irq = dma->irq; dma_private->substream = substream; dma_private->ld_buf_phys = ld_buf_phys; dma_private->dma_buf_phys = substream->dma_buffer.addr; - /* We only support one DMA controller for now */ - dma_private->controller_id = 0; - dma_private->channel_id = channel; - ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private); if (ret) { - dev_err(substream->pcm->card->dev, - "can't register ISR for IRQ %u (ret=%i)\n", + dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n", dma_private->irq, ret); - dma_free_coherent(substream->pcm->card->dev, - sizeof(struct fsl_dma_private), + dma_free_coherent(dev, sizeof(struct fsl_dma_private), dma_private, dma_private->ld_buf_phys); return ret; } - dma_global_data.assigned[channel] = 1; + dma->assigned = 1; snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); @@ -545,6 +520,8 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; /* Number of bits per sample */ unsigned int sample_size = @@ -605,8 +582,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, break; default: /* We should never get here */ - dev_err(substream->pcm->card->dev, - "unsupported sample size %u\n", sample_size); + dev_err(dev, "unsupported sample size %u\n", sample_size); return -EINVAL; } @@ -688,6 +664,8 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; dma_addr_t position; snd_pcm_uframes_t frames; @@ -709,8 +687,7 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) if ((position < dma_private->dma_buf_phys) || (position > dma_private->dma_buf_end)) { - dev_err(substream->pcm->card->dev, - "dma pointer is out of range, halting stream\n"); + dev_err(dev, "dma pointer is out of range, halting stream\n"); return SNDRV_PCM_POS_XRUN; } @@ -771,26 +748,28 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; - int dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; + struct dma_object *dma = + container_of(rtd->platform->driver, struct dma_object, dai); if (dma_private) { if (dma_private->irq) free_irq(dma_private->irq, dma_private); if (dma_private->ld_buf_phys) { - dma_unmap_single(substream->pcm->card->dev, - dma_private->ld_buf_phys, - sizeof(dma_private->link), DMA_TO_DEVICE); + dma_unmap_single(dev, dma_private->ld_buf_phys, + sizeof(dma_private->link), + DMA_TO_DEVICE); } /* Deallocate the fsl_dma_private structure */ - dma_free_coherent(substream->pcm->card->dev, - sizeof(struct fsl_dma_private), - dma_private, dma_private->ld_buf_phys); + dma_free_coherent(dev, sizeof(struct fsl_dma_private), + dma_private, dma_private->ld_buf_phys); substream->runtime->private_data = NULL; } - dma_global_data.assigned[dir] = 0; + dma->assigned = 0; return 0; } @@ -813,6 +792,41 @@ static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm) } } +/* List of DMA nodes that we've probed */ +static LIST_HEAD(dma_list); + +/** + * fsl_soc_dma_to_dai - returning the DAI struct for a given DMA node + * + * When a machine driver registers itself with ASoC, it must provide the + * address of the DAI structures that it wants to connect. Most drivers simply + * reference global data structures in the various drivers, but we PowerPC + * developers are better than that. This driver keeps a list of every DMA + * node that it probes, and when the machine driver wants to reference a + * DMA DAI, this function provides that service. + * + * Hopefully one day, ASoC will just want the name of the DMA device, not + * some obscure pointer, and this function will go away. + */ +struct snd_soc_platform_driver *fsl_soc_dma_to_dai(const char *path, + dma_addr_t ssi_stx_phys, dma_addr_t ssi_srx_phys) +{ + struct list_head *ptr; + struct dma_object *dma; + + list_for_each(ptr, &dma_list) { + dma = list_entry(ptr, struct dma_object, list); + if (strcmp(path, dma->path) == 0) { + dma->ssi_stx_phys = ssi_stx_phys; + dma->ssi_srx_phys = ssi_srx_phys; + return &dma->dai; + } + } + + return NULL; +} +EXPORT_SYMBOL(fsl_soc_dma_to_dai); + static struct snd_pcm_ops fsl_dma_ops = { .open = fsl_dma_open, .close = fsl_dma_close, @@ -822,80 +836,91 @@ static struct snd_pcm_ops fsl_dma_ops = { .pointer = fsl_dma_pointer, }; -struct snd_soc_platform_driver fsl_soc_platform = { - .name = "fsl-dma", - .ops = &fsl_dma_ops, - .pcm_new = fsl_dma_new, - .pcm_free = fsl_dma_free_dma_buffers, -}; -EXPORT_SYMBOL_GPL(fsl_soc_platform); - -/** - * fsl_dma_configure: store the DMA parameters from the fabric driver. - * - * This function is called by the ASoC fabric driver to give us the DMA and - * SSI channel information. - * - * Unfortunately, ASoC V1 does make it possible to determine the DMA/SSI - * data when a substream is created, so for now we need to store this data - * into a global variable. This means that we can only support one DMA - * controller, and hence only one SSI. - */ -int fsl_dma_configure(struct fsl_dma_info *dma_info) +static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, + const struct of_device_id *match) { - static int initialized; + struct dma_object *dma; + struct device_node *np = of_dev->node; + int ret; - /* We only support one DMA controller for now */ - if (initialized) - return 0; + dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); + if (!dma) { + dev_err(&of_dev->dev, "could not allocate dma object\n"); + return -ENOMEM; + } - dma_global_data.ssi_stx_phys = dma_info->ssi_stx_phys; - dma_global_data.ssi_srx_phys = dma_info->ssi_srx_phys; - dma_global_data.dma_channel[0] = dma_info->dma_channel[0]; - dma_global_data.dma_channel[1] = dma_info->dma_channel[1]; - dma_global_data.irq[0] = dma_info->dma_irq[0]; - dma_global_data.irq[1] = dma_info->dma_irq[1]; - dma_global_data.assigned[0] = 0; - dma_global_data.assigned[1] = 0; - - initialized = 1; - return 1; -} -EXPORT_SYMBOL_GPL(fsl_dma_configure); + strcpy(dma->path, np->full_name); + dma->dai.name = dma->path; + dma->dai.ops = &fsl_dma_ops; + dma->dai.pcm_new = fsl_dma_new; + dma->dai.pcm_free = fsl_dma_free_dma_buffers; -static int __devinit fsl_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, -1, &fsl_soc_platform); + ret = snd_soc_register_platform(&of_dev->dev, 0, &dma->dai); + if (ret) { + dev_err(&of_dev->dev, "could not register platform\n"); + kfree(dma); + return ret; + } + + dma->channel = of_iomap(np, 0); + dma->irq = irq_of_parse_and_map(np, 0); + list_add(&dma->list, &dma_list); + + return 0; } -static int __devexit fsl_soc_platform_remove(struct platform_device *pdev) +static int __devexit fsl_soc_dma_remove(struct of_device *of_dev) { - snd_soc_unregister_platform(&pdev->dev, -1); + struct list_head *n, *ptr; + struct dma_object *dma; + + list_for_each_safe(ptr, n, &dma_list) { + dma = list_entry(ptr, struct dma_object, list); + list_del_init(ptr); + + snd_soc_unregister_platform(&of_dev->dev, 0); + iounmap(dma->channel); + irq_dispose_mapping(dma->irq); + kfree(dma); + } + return 0; } -static struct platform_driver fsl_pcm_driver = { - .driver = { - .name = "fsl-pcm-audio", - .owner = THIS_MODULE, - }, - .probe = fsl_soc_platform_probe, - .remove = __devexit_p(fsl_soc_platform_remove), +static const struct of_device_id fsl_soc_dma_ids[] = { + { .compatible = "fsl,ssi-dma-channel", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_soc_dma_ids); + +static struct of_platform_driver fsl_soc_dma_driver = { + .name = "fsl-soc-dma-dai", + .match_table = fsl_soc_dma_ids, + .probe = fsl_soc_dma_probe, + .remove = __devexit_p(fsl_soc_dma_remove), }; -static int __init snd_fsl_pcm_init(void) +static int __init fsl_soc_dma_init(void) { - return platform_driver_register(&fsl_pcm_driver); + pr_info("Freescale Elo DMA ASoC PCM Driver\n"); + + return of_register_platform_driver(&fsl_soc_dma_driver); } -module_init(snd_fsl_pcm_init); -static void __exit snd_fsl_pcm_exit(void) +static void __exit fsl_soc_dma_exit(void) { - platform_driver_unregister(&fsl_pcm_driver); + of_unregister_platform_driver(&fsl_soc_dma_driver); } -module_exit(snd_fsl_pcm_exit); + +/* We want the DMA driver to be initialized before the SSI driver, so that + * when the SSI driver calls fsl_soc_dma_dai_from_node(), the DMA driver + * will already have been probed. The easiest way to do that is to make the + * __init function called via arch_initcall(). + */ +arch_initcall(fsl_soc_dma_init); +module_exit(fsl_soc_dma_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module"); -MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index 253bd3c..88c686d 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -126,24 +126,9 @@ struct fsl_dma_link_descriptor { u8 res[4]; /* Reserved */ } __attribute__ ((aligned(32), packed)); -/* DMA information needed to create a snd_soc_dai object - * - * ssi_stx_phys: bus address of SSI STX register to use - * ssi_srx_phys: bus address of SSI SRX register to use - * dma[0]: points to the DMA channel to use for playback - * dma[1]: points to the DMA channel to use for capture - * dma_irq[0]: IRQ of the DMA channel to use for playback - * dma_irq[1]: IRQ of the DMA channel to use for capture - */ -struct fsl_dma_info { - dma_addr_t ssi_stx_phys; - dma_addr_t ssi_srx_phys; - struct ccsr_dma_channel __iomem *dma_channel[2]; - unsigned int dma_irq[2]; -}; - extern struct snd_soc_platform_driver fsl_soc_platform; -int fsl_dma_configure(struct fsl_dma_info *dma_info); +struct snd_soc_platform_driver *fsl_soc_dma_to_dai(const char *path, + dma_addr_t ssi_stx_phys, dma_addr_t ssi_srx_phys); #endif diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 780e243..56e66c2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -3,10 +3,11 @@ * * Author: Timur Tabi <timur(a)freescale.com> * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. */ #include <linux/init.h> @@ -14,6 +15,7 @@ #include <linux/interrupt.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/of_platform.h> #include <sound/core.h> #include <sound/pcm.h> @@ -70,33 +72,31 @@ /** * fsl_ssi_private: per-SSI private data * - * @name: short name for this device ("SSI0", "SSI1", etc) * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI * @first_stream: pointer to the stream that was opened first * @second_stream: pointer to second stream - * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened * @asynchronous: 0=synchronous mode, 1=asynchronous mode * @cpu_dai: the CPU DAI for this device * @dev_attr: the sysfs device attribute structure * @stats: SSI statistics + * @name: name for this device */ struct fsl_ssi_private { - char name[8]; struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; struct snd_pcm_substream *first_stream; struct snd_pcm_substream *second_stream; - struct device *dev; unsigned int playback; unsigned int capture; int asynchronous; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; + struct platform_device *pdev; struct { unsigned int rfrc; @@ -121,6 +121,8 @@ struct fsl_ssi_private { unsigned int tfe1; unsigned int tfe0; } stats; + + char name[1]; }; /** @@ -289,8 +291,10 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct ccsr_ssi __iomem *ssi = ssi_private->ssi; int ret; + /* The 'name' should not have any slashes in it. */ ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, - ssi_private->name, ssi_private); + strrchr(ssi_private->name, '/') + 1, + ssi_private); if (ret < 0) { dev_err(substream->pcm->card->dev, "could not claim irq %u\n", ssi_private->irq); @@ -522,56 +526,15 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, } } -/** - * fsl_ssi_set_sysclk: set the clock frequency and direction - * - * This function is called by the machine driver to tell us what the clock - * frequency and direction are. - * - * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN), - * and we don't care about the frequency. Return an error if the direction - * is not SND_SOC_CLOCK_IN. - * - * @clk_id: reserved, should be zero - * @freq: the frequency of the given clock ID, currently ignored - * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master) - */ -static int fsl_ssi_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - - return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL; -} - -/** - * fsl_ssi_set_fmt: set the serial format. - * - * This function is called by the machine driver to tell us what serial - * format to use. - * - * Currently, we only support I2S mode. Return an error if the format is - * not SND_SOC_DAIFMT_I2S. - * - * @format: one of SND_SOC_DAIFMT_xxx - */ -static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) -{ - return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; -} - -/** - * fsl_ssi_dai_template: template CPU DAI for the SSI - */ static struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, - .set_sysclk = fsl_ssi_set_sysclk, - .set_fmt = fsl_ssi_set_fmt, }; -struct snd_soc_dai_driver fsl_ssi_dai = { +/* Template for the CPU dai driver structure */ +static struct snd_soc_dai_driver fsl_ssi_dai_template = { .playback = { /* The SSI does not support monaural audio. */ .channels_min = 2, @@ -587,7 +550,6 @@ struct snd_soc_dai_driver fsl_ssi_dai = { }, .ops = &fsl_ssi_dai_ops, }; -EXPORT_SYMBOL_GPL(fsl_ssi_dai); /* Show the statistics of a flag only if its interrupt is enabled. The * compiler will optimze this code to a no-op if the interrupt is not @@ -639,92 +601,163 @@ static ssize_t fsl_sysfs_ssi_show(struct device *dev, return length; } -static int fsl_ssi_probe(struct platform_device *pdev) +/** + * Make every character in a string lower-case + */ +static void make_lowercase(char *s) +{ + char *p = s; + char c; + + while ((c = *p)) { + if ((c >= 'A') && (c <= 'Z')) + *p = c + ('a' - 'A'); + p++; + } +} + +static int __devinit fsl_ssi_probe(struct of_device *of_dev, + const struct of_device_id *match) { - struct fsl_ssi_info *ssi_info = dev_get_platdata(&pdev->dev); struct fsl_ssi_private *ssi_private; int ret = 0; struct device_attribute *dev_attr; + struct device_node *np = of_dev->node; + const char *sprop; + struct resource res; + char name[64]; + struct platform_device *pdev; + void *platform_data; + + /* We are only interested in SSIs with a codec phandle in them, so let's + * make sure this SSI has one. + */ + if (!of_get_property(np, "codec-handle", NULL)) + return -ENODEV; + + /* We only support the SSI in "I2S Slave" mode */ + sprop = of_get_property(np, "fsl,mode", NULL); + if (!sprop || strcmp(sprop, "i2s-slave")) { + dev_notice(&of_dev->dev, "mode %s is unsupported\n", sprop); + return -ENODEV; + } - ssi_private = kzalloc(sizeof(struct fsl_ssi_private), GFP_KERNEL); + ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + + strlen(np->full_name), GFP_KERNEL); if (!ssi_private) { - dev_err(ssi_info->dev, "could not allocate DAI object\n"); + dev_err(&of_dev->dev, "could not allocate DAI object\n"); return -ENOMEM; } - dev_attr = &ssi_private->dev_attr; + /* Initialize this copy of the CPU DAI driver structure */ + memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, + sizeof(fsl_ssi_dai_template)); + ssi_private->cpu_dai_drv.name = ssi_private->name; + + /* The name we register with ASoC is the full node name */ + strlcpy(ssi_private->name, np->full_name, sizeof(ssi_private->name)); + + /* Get the addresses and IRQ */ + ret = of_address_to_resource(np, 0, &res); + if (ret) { + dev_err(&of_dev->dev, "could not determine device resources\n"); + kfree(ssi_private); + return ret; + } + ssi_private->ssi = ioremap(res.start, 1 + res.end - res.start); + ssi_private->ssi_phys = res.start; + ssi_private->irq = irq_of_parse_and_map(np, 0); - sprintf(ssi_private->name, "ssi%u", (u8) ssi_info->id); - ssi_private->ssi = ssi_info->ssi; - ssi_private->ssi_phys = ssi_info->ssi_phys; - ssi_private->irq = ssi_info->irq; - ssi_private->dev = ssi_info->dev; - ssi_private->asynchronous = ssi_info->asynchronous; + /* Are the RX and the TX clocks locked? */ + if (of_find_property(np, "fsl,ssi-asynchronous", NULL)) + ssi_private->asynchronous = 1; + else + ssi_private->cpu_dai_drv.symmetric_rates = 1; /* Initialize the the device_attribute structure */ - dev_attr->attr.name = "ssi-stats"; + dev_attr = &ssi_private->dev_attr; + dev_attr->attr.name = "statistics"; dev_attr->attr.mode = S_IRUGO; dev_attr->show = fsl_sysfs_ssi_show; - ret = device_create_file(ssi_private->dev, dev_attr); + ret = device_create_file(&of_dev->dev, dev_attr); if (ret) { - dev_err(ssi_info->dev, "could not create sysfs %s file\n", + dev_err(&of_dev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); kfree(ssi_private); return ret; } - dev_set_drvdata(ssi_private->dev, ssi_private); - fsl_ssi_dai.name = ssi_private->name; - fsl_ssi_dai.symmetric_rates = 1; + /* Register with ASoC */ + dev_set_drvdata(&of_dev->dev, ssi_private); - ret = snd_soc_register_dai(ssi_private->dev, ssi_info->id, &fsl_ssi_dai); + ret = snd_soc_register_dai(&of_dev->dev, 0, &ssi_private->cpu_dai_drv); if (ret != 0) { - dev_err(ssi_info->dev, "failed to register DAI: %d\n", ret); + dev_err(&of_dev->dev, "failed to register DAI: %d\n", ret); kfree(ssi_private); return ret; } - return ret; + /* Trigger the machine driver's probe function. The platform driver + * name of the machine driver is taken from the /model node of the + * device tree. We also pass the address of the CPU DAI driver + * structure. + */ + sprop = of_get_property(of_find_node_by_path("/"), "model", NULL); + snprintf(name, sizeof(name), "snd-soc-%s", sprop); + make_lowercase(name); + + platform_data = &ssi_private->cpu_dai_drv; + pdev = platform_device_register_data(&of_dev->dev, name, -1, + &platform_data, sizeof(void *)); + + ssi_private->pdev = pdev; + + return 0; } /** * fsl_ssi_destroy_dai: destroy the snd_soc_dai object * - * This function undoes the operations of fsl_ssi_create_dai() + * This function undoes the operations of fsl_ssi_probe() */ -static int fsl_ssi_remove(struct platform_device *pdev) +static int fsl_ssi_remove(struct of_device *of_dev) { - struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev); - struct fsl_ssi_info *ssi_info = dev_get_platdata(&pdev->dev); - - device_remove_file(ssi_private->dev, &ssi_private->dev_attr); + struct fsl_ssi_private *ssi_private = dev_get_drvdata(&of_dev->dev); - snd_soc_unregister_dai(&pdev->dev, ssi_info->id); + platform_device_unregister(ssi_private->pdev); + snd_soc_unregister_dai(&of_dev->dev, 0); + device_remove_file(&of_dev->dev, &ssi_private->dev_attr); kfree(ssi_private); + dev_set_drvdata(&of_dev->dev, NULL); + return 0; } -static struct platform_driver fsl_ssi_driver = { +static const struct of_device_id fsl_ssi_ids[] = { + { .compatible = "fsl,mpc8610-ssi", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_ssi_ids); + +static struct of_platform_driver fsl_ssi_driver = { + .name = "fsl-ssi-dai", + .match_table = fsl_ssi_ids, .probe = fsl_ssi_probe, - .remove = __devexit_p(fsl_ssi_remove), - .driver = { - .name = "fsl-ssi-dai", - .owner = THIS_MODULE, - }, + .remove = fsl_ssi_remove, }; static int __init fsl_ssi_init(void) { printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n"); - return platform_driver_register(&fsl_ssi_driver); + return of_register_platform_driver(&fsl_ssi_driver); } static void __exit fsl_ssi_exit(void) { - platform_driver_unregister(&fsl_ssi_driver); + of_unregister_platform_driver(&fsl_ssi_driver); } module_init(fsl_ssi_init); @@ -732,4 +765,4 @@ module_exit(fsl_ssi_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 965fd86..2173000 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -196,30 +196,5 @@ struct ccsr_ssi { #define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT) #define CCSR_SSI_SOR_SYNRST 0x00000001 -/* Instantiation data for an SSI interface - * - * This structure contains all the information that the the SSI driver needs - * to instantiate an SSI interface with ALSA. The machine driver should - * create this structure, fill it in, call fsl_ssi_create_dai(), and then - * delete the structure. - * - * id: which SSI this is (0, 1, etc. ) - * ssi: pointer to the SSI's registers - * ssi_phys: physical address of the SSI registers - * irq: IRQ of this SSI - * dev: struct device, used to create the sysfs statistics file - * asynchronous: 0=synchronous mode, 1=asynchronous mode -*/ -struct fsl_ssi_info { - unsigned int id; - struct ccsr_ssi __iomem *ssi; - dma_addr_t ssi_phys; - unsigned int irq; - struct device *dev; - int asynchronous; -}; - -extern struct snd_soc_dai_driver fsl_ssi_dai; - #endif diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c dissimilarity index 62% index ad8d763..3e422f7 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -1,616 +1,558 @@ -/** - * Freescale MPC8610HPCD ALSA SoC Fabric driver - * - * Author: Timur Tabi <timur(a)freescale.com> - * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. - */ - -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/of_device.h> -#include <linux/of_platform.h> -#include <sound/soc.h> -#include <asm/immap_86xx.h> - -#include "../codecs/cs4270.h" -#include "fsl_dma.h" -#include "fsl_ssi.h" - -/** - * mpc8610_hpcd_data: fabric-specific ASoC device data - * - * This structure contains data for a single sound platform device on an - * MPC8610 HPCD. Some of the data is taken from the device tree. - */ -struct mpc8610_hpcd_data { - struct snd_soc_dai_link dai; - struct snd_soc_card machine; - unsigned int dai_format; - unsigned int codec_clk_direction; - unsigned int cpu_clk_direction; - unsigned int clk_frequency; - struct ccsr_guts __iomem *guts; - struct ccsr_ssi __iomem *ssi; - unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ - unsigned int ssi_irq; - unsigned int dma_id; /* 0 = DMA1, 1 = DMA2, etc */ - unsigned int dma_irq[2]; - struct ccsr_dma_channel __iomem *dma[2]; - unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ -}; - -/** - * mpc8610_hpcd_machine_probe: initalize the board - * - * This function is called when platform_device_add() is called. It is used - * to initialize the board-specific hardware. - * - * Here we program the DMACR and PMUXCR registers. - */ -static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) -{ - struct mpc8610_hpcd_data *machine_data = - sound_device->dev.platform_data; - - /* Program the signal routing between the SSI and the DMA */ - guts_set_dmacr(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI); - guts_set_dmacr(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI); - - guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[0], 0); - guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[1], 0); - - switch (machine_data->ssi_id) { - case 0: - clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); - break; - case 1: - clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); - break; - } - - return 0; -} - -/** - * mpc8610_hpcd_startup: program the board with various hardware parameters - * - * This function takes board-specific information, like clock frequencies - * and serial data formats, and passes that information to the codec and - * transport drivers. - */ -static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct mpc8610_hpcd_data *machine_data = - rtd->card->dev->platform_data; - int ret = 0; - - /* Tell the CPU driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver audio format\n"); - return ret; - } - - /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(codec_dai, machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver audio format\n"); - return ret; - } - - /* - * Tell the CPU driver what the clock frequency is, and whether it's a - * slave or master. - */ - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, - machine_data->clk_frequency, - machine_data->cpu_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver clock parameters\n"); - return ret; - } - - /* - * Tell the codec driver what the MCLK frequency is, and whether it's - * a slave or master. - */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - machine_data->clk_frequency, - machine_data->codec_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver clock params\n"); - return ret; - } - - return 0; -} - -/** - * mpc8610_hpcd_machine_remove: Remove the sound device - * - * This function is called to remove the sound device for one SSI. We - * de-program the DMACR and PMUXCR register. - */ -int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) -{ - struct mpc8610_hpcd_data *machine_data = - sound_device->dev.platform_data; - - /* Restore the signal routing */ - - guts_set_dmacr(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[0], 0); - guts_set_dmacr(machine_data->guts, machine_data->dma_id, - machine_data->dma_channel_id[1], 0); - - switch (machine_data->ssi_id) { - case 0: - clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); - break; - case 1: - clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA); - break; - } - - return 0; -} - -/** - * mpc8610_hpcd_ops: ASoC fabric driver operations - */ -static struct snd_soc_ops mpc8610_hpcd_ops = { - .startup = mpc8610_hpcd_startup, -}; - -/** - * mpc8610_hpcd_probe: OF probe function for the fabric driver - * - * This function gets called when an SSI node is found in the device tree. - * - * Although this is a fabric driver, the SSI node is the "master" node with - * respect to audio hardware connections. Therefore, we create a new ASoC - * device for each new SSI node that has a codec attached. - * - * FIXME: Currently, we only support one DMA controller, so if there are - * multiple SSI nodes with codecs, only the first will be supported. - * - * FIXME: Even if we did support multiple DMA controllers, we have no - * mechanism for assigning DMA controllers and channels to the individual - * SSI devices. We also probably aren't compatible with the generic Elo DMA - * device driver. - */ -static int mpc8610_hpcd_probe(struct of_device *ofdev, - const struct of_device_id *match) -{ - struct device_node *np = ofdev->node; - struct device_node *codec_np = NULL; - struct device_node *guts_np = NULL; - struct device_node *dma_np = NULL; - struct device_node *dma_channel_np = NULL; - const phandle *codec_ph; - const char *sprop; - const u32 *iprop; - struct resource res; - struct platform_device *sound_device = NULL; - struct mpc8610_hpcd_data *machine_data; - struct fsl_ssi_info ssi_info; - struct fsl_dma_info dma_info; - int ret = -ENODEV; - unsigned int playback_dma_channel; - unsigned int capture_dma_channel; - - machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); - if (!machine_data) - return -ENOMEM; - - memset(&ssi_info, 0, sizeof(ssi_info)); - memset(&dma_info, 0, sizeof(dma_info)); - - ssi_info.dev = &ofdev->dev; - - /* - * We are only interested in SSIs with a codec phandle in them, so let's - * make sure this SSI has one. - */ - codec_ph = of_get_property(np, "codec-handle", NULL); - if (!codec_ph) - goto error; - - codec_np = of_find_node_by_phandle(*codec_ph); - if (!codec_np) - goto error; - - /* The MPC8610 HPCD only knows about the CS4270 codec, so reject - anything else. */ - if (!of_device_is_compatible(codec_np, "cirrus,cs4270")) - goto error; - - /* Get the device ID */ - iprop = of_get_property(np, "cell-index", NULL); - if (!iprop) { - dev_err(&ofdev->dev, "cell-index property not found\n"); - ret = -EINVAL; - goto error; - } - machine_data->ssi_id = *iprop; - ssi_info.id = *iprop; - - /* Get the serial format and clock direction. */ - sprop = of_get_property(np, "fsl,mode", NULL); - if (!sprop) { - dev_err(&ofdev->dev, "fsl,mode property not found\n"); - ret = -EINVAL; - goto error; - } - - if (strcasecmp(sprop, "i2s-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - - /* - * In i2s-slave mode, the codec has its own clock source, so we - * need to get the frequency from the device tree and pass it to - * the codec driver. - */ - iprop = of_get_property(codec_np, "clock-frequency", NULL); - if (!iprop || !*iprop) { - dev_err(&ofdev->dev, "codec bus-frequency property " - "is missing or invalid\n"); - ret = -EINVAL; - goto error; - } - machine_data->clk_frequency = *iprop; - } else if (strcasecmp(sprop, "i2s-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "lj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "lj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "rj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "rj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "ac97-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "ac97-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else { - dev_err(&ofdev->dev, - "unrecognized fsl,mode property \"%s\"\n", sprop); - ret = -EINVAL; - goto error; - } - - if (!machine_data->clk_frequency) { - dev_err(&ofdev->dev, "unknown clock frequency\n"); - ret = -EINVAL; - goto error; - } - - /* Read the SSI information from the device tree */ - ret = of_address_to_resource(np, 0, &res); - if (ret) { - dev_err(&ofdev->dev, "could not obtain SSI address\n"); - goto error; - } - if (!res.start) { - dev_err(&ofdev->dev, "invalid SSI address\n"); - goto error; - } - ssi_info.ssi_phys = res.start; - - machine_data->ssi = ioremap(ssi_info.ssi_phys, sizeof(struct ccsr_ssi)); - if (!machine_data->ssi) { - dev_err(&ofdev->dev, "could not map SSI address %x\n", - ssi_info.ssi_phys); - ret = -EINVAL; - goto error; - } - ssi_info.ssi = machine_data->ssi; - - - /* Get the IRQ of the SSI */ - machine_data->ssi_irq = irq_of_parse_and_map(np, 0); - if (!machine_data->ssi_irq) { - dev_err(&ofdev->dev, "could not get SSI IRQ\n"); - ret = -EINVAL; - goto error; - } - ssi_info.irq = machine_data->ssi_irq; - - /* Do we want to use asynchronous mode? */ - ssi_info.asynchronous = - of_find_property(np, "fsl,ssi-asynchronous", NULL) ? 1 : 0; - if (ssi_info.asynchronous) - dev_info(&ofdev->dev, "using asynchronous mode\n"); - - /* Map the global utilities registers. */ - guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); - if (!guts_np) { - dev_err(&ofdev->dev, "could not obtain address of GUTS\n"); - ret = -EINVAL; - goto error; - } - machine_data->guts = of_iomap(guts_np, 0); - of_node_put(guts_np); - if (!machine_data->guts) { - dev_err(&ofdev->dev, "could not map GUTS\n"); - ret = -EINVAL; - goto error; - } - - /* Find the DMA channels to use. Both SSIs need to use the same DMA - * controller, so let's use DMA#1. - */ - for_each_compatible_node(dma_np, NULL, "fsl,mpc8610-dma") { - iprop = of_get_property(dma_np, "cell-index", NULL); - if (iprop && (*iprop == 0)) { - of_node_put(dma_np); - break; - } - } - if (!dma_np) { - dev_err(&ofdev->dev, "could not find DMA node\n"); - ret = -EINVAL; - goto error; - } - machine_data->dma_id = *iprop; - - /* SSI1 needs to use DMA Channels 0 and 1, and SSI2 needs to use DMA - * channels 2 and 3. This is just how the MPC8610 is wired - * internally. - */ - playback_dma_channel = (machine_data->ssi_id == 0) ? 0 : 2; - capture_dma_channel = (machine_data->ssi_id == 0) ? 1 : 3; - - /* - * Find the DMA channels to use. - */ - while ((dma_channel_np = of_get_next_child(dma_np, dma_channel_np))) { - iprop = of_get_property(dma_channel_np, "cell-index", NULL); - if (iprop && (*iprop == playback_dma_channel)) { - /* dma_channel[0] and dma_irq[0] are for playback */ - dma_info.dma_channel[0] = of_iomap(dma_channel_np, 0); - dma_info.dma_irq[0] = - irq_of_parse_and_map(dma_channel_np, 0); - machine_data->dma_channel_id[0] = *iprop; - continue; - } - if (iprop && (*iprop == capture_dma_channel)) { - /* dma_channel[1] and dma_irq[1] are for capture */ - dma_info.dma_channel[1] = of_iomap(dma_channel_np, 0); - dma_info.dma_irq[1] = - irq_of_parse_and_map(dma_channel_np, 0); - machine_data->dma_channel_id[1] = *iprop; - continue; - } - } - if (!dma_info.dma_channel[0] || !dma_info.dma_channel[1] || - !dma_info.dma_irq[0] || !dma_info.dma_irq[1]) { - dev_err(&ofdev->dev, "could not find DMA channels\n"); - ret = -EINVAL; - goto error; - } - - dma_info.ssi_stx_phys = ssi_info.ssi_phys + - offsetof(struct ccsr_ssi, stx0); - dma_info.ssi_srx_phys = ssi_info.ssi_phys + - offsetof(struct ccsr_ssi, srx0); - - /* We have the DMA information, so tell the DMA driver what it is */ - if (!fsl_dma_configure(&dma_info)) { - dev_err(&ofdev->dev, "could not instantiate DMA device\n"); - ret = -EBUSY; - goto error; - } - - /* - * Initialize our DAI data structure. We should probably get this - * information from the device tree. - */ - machine_data->dai.name = "CS4270"; - machine_data->dai.stream_name = "CS4270"; - - iprop = of_get_property(codec_np, "reg", NULL); - if (!iprop) { - dev_err(&ofdev->dev, "codec node is missing 'reg' property\n"); - goto error; - } - machine_data->dai.codec_id = *iprop; - - machine_data->dai.cpu_dai_drv = &fsl_ssi_dai; - machine_data->dai.codec_dai_drv = &cs4270_dai; /* The codec_dai we want */ - machine_data->dai.codec_drv = &soc_codec_device_cs4270; - machine_data->dai.ops = &mpc8610_hpcd_ops; - machine_data->dai.platform_drv = &fsl_soc_platform; - - machine_data->machine.probe = mpc8610_hpcd_machine_probe; - machine_data->machine.remove = mpc8610_hpcd_machine_remove; - machine_data->machine.name = "MPC8610 HPCD"; - machine_data->machine.num_links = 1; - machine_data->machine.dai_link = &machine_data->dai; - - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&ofdev->dev, "platform device allocation failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Set the platform device and ASoC device to point to each other */ - platform_set_drvdata(sound_device, &machine_data->machine); - - /* Tell ASoC to probe us. This will call mpc8610_hpcd_machine.probe(), - if it exists. */ - ret = platform_device_add(sound_device); - - if (ret) { - dev_err(&ofdev->dev, "platform device add failed\n"); - goto error; - } - - dev_set_drvdata(&ofdev->dev, sound_device); - - return 0; - -error: - of_node_put(codec_np); - of_node_put(guts_np); - of_node_put(dma_np); - of_node_put(dma_channel_np); - - if (sound_device) - platform_device_unregister(sound_device); - - if (ssi_info.ssi) - iounmap(ssi_info.ssi); - - if (ssi_info.irq) - irq_dispose_mapping(ssi_info.irq); - - if (dma_info.dma_channel[0]) - iounmap(dma_info.dma_channel[0]); - - if (dma_info.dma_channel[1]) - iounmap(dma_info.dma_channel[1]); - - if (dma_info.dma_irq[0]) - irq_dispose_mapping(dma_info.dma_irq[0]); - - if (dma_info.dma_irq[1]) - irq_dispose_mapping(dma_info.dma_irq[1]); - - if (machine_data->guts) - iounmap(machine_data->guts); - - kfree(machine_data); - - return ret; -} - -/** - * mpc8610_hpcd_remove: remove the OF device - * - * This function is called when the OF device is removed. - */ -static int mpc8610_hpcd_remove(struct of_device *ofdev) -{ - struct platform_device *sound_device = dev_get_drvdata(&ofdev->dev); - struct mpc8610_hpcd_data *machine_data = - sound_device->dev.platform_data; - - platform_device_unregister(sound_device); - - if (machine_data->ssi) - iounmap(machine_data->ssi); - - if (machine_data->dma[0]) - iounmap(machine_data->dma[0]); - - if (machine_data->dma[1]) - iounmap(machine_data->dma[1]); - - if (machine_data->dma_irq[0]) - irq_dispose_mapping(machine_data->dma_irq[0]); - - if (machine_data->dma_irq[1]) - irq_dispose_mapping(machine_data->dma_irq[1]); - - if (machine_data->guts) - iounmap(machine_data->guts); - - kfree(machine_data); - sound_device->dev.platform_data = NULL; - - dev_set_drvdata(&ofdev->dev, NULL); - - return 0; -} - -static struct of_device_id mpc8610_hpcd_match[] = { - { - .compatible = "fsl,mpc8610-ssi", - }, - {} -}; -MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match); - -static struct of_platform_driver mpc8610_hpcd_of_driver = { - .owner = THIS_MODULE, - .name = "mpc8610_hpcd", - .match_table = mpc8610_hpcd_match, - .probe = mpc8610_hpcd_probe, - .remove = mpc8610_hpcd_remove, -}; - -/** - * mpc8610_hpcd_init: fabric driver initialization. - * - * This function is called when this module is loaded. - */ -static int __init mpc8610_hpcd_init(void) -{ - int ret; - - printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n"); - - ret = of_register_platform_driver(&mpc8610_hpcd_of_driver); - - if (ret) - printk(KERN_ERR - "mpc8610-hpcd: failed to register platform driver\n"); - - return ret; -} - -/** - * mpc8610_hpcd_exit: fabric driver exit - * - * This function is called when this driver is unloaded. - */ -static void __exit mpc8610_hpcd_exit(void) -{ - of_unregister_platform_driver(&mpc8610_hpcd_of_driver); -} - -module_init(mpc8610_hpcd_init); -module_exit(mpc8610_hpcd_exit); - -MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC fabric driver"); -MODULE_LICENSE("GPL"); +/** + * Freescale MPC8610HPCD ALSA SoC Machine driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/of_device.h> +#include <sound/soc.h> +#include <asm/immap_86xx.h> + +#include "../codecs/cs4270.h" +#include "fsl_dma.h" +#include "fsl_ssi.h" + +/* There's only one global utilities register */ +static phys_addr_t guts_phys; + +/** + * mpc8610_hpcd_data: machine-specific ASoC device data + * + * This structure contains data for a single sound platform device on an + * MPC8610 HPCD. Some of the data is taken from the device tree. + */ +struct mpc8610_hpcd_data { + struct snd_soc_dai_link dai[2]; + struct snd_soc_card card; + unsigned int dai_format; + unsigned int codec_clk_direction; + unsigned int cpu_clk_direction; + unsigned int clk_frequency; + unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ + unsigned int dma_id[0]; /* 0 = DMA1, 1 = DMA2, etc */ + unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ + char dai_name[32]; +}; + +/** + * mpc8610_hpcd_machine_probe: initialize the board + * + * This function is used to initialize the board-specific hardware. + * + * Here we program the DMACR and PMUXCR registers. + */ +static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) +{ + struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct mpc8610_hpcd_data *machine_data = + container_of(card, struct mpc8610_hpcd_data, card); + struct ccsr_guts __iomem *guts; + + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + if (!guts) { + dev_err(card->dev, "could not map global utilities\n"); + return -ENOMEM; + } + + /* Program the signal routing between the SSI and the DMA */ + guts_set_dmacr(guts, machine_data->dma_id[0], + machine_data->dma_channel_id[0], + CCSR_GUTS_DMACR_DEV_SSI); + guts_set_dmacr(guts, machine_data->dma_id[1], + machine_data->dma_channel_id[1], + CCSR_GUTS_DMACR_DEV_SSI); + + guts_set_pmuxcr_dma(guts, machine_data->dma_id[0], + machine_data->dma_channel_id[0], 0); + guts_set_pmuxcr_dma(guts, machine_data->dma_id[1], + machine_data->dma_channel_id[1], 0); + + switch (machine_data->ssi_id) { + case 0: + clrsetbits_be32(&guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); + break; + case 1: + clrsetbits_be32(&guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); + break; + } + + iounmap(guts); + + return 0; +} + +/** + * mpc8610_hpcd_startup: program the board with various hardware parameters + * + * This function takes board-specific information, like clock frequencies + * and serial data formats, and passes that information to the codec and + * transport drivers. + */ +static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mpc8610_hpcd_data *machine_data = + container_of(rtd->card, struct mpc8610_hpcd_data, card); + struct device *dev = rtd->card->dev; + int ret = 0; + + /* Tell the codec driver what the serial protocol is. */ + ret = snd_soc_dai_set_fmt(rtd->codec_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(dev, "could not set codec driver audio format\n"); + return ret; + } + + /* + * Tell the codec driver what the MCLK frequency is, and whether it's + * a slave or master. + */ + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, + machine_data->clk_frequency, + machine_data->codec_clk_direction); + if (ret < 0) { + dev_err(dev, "could not set codec driver clock params\n"); + return ret; + } + + return 0; +} + +/** + * mpc8610_hpcd_machine_remove: Remove the sound device + * + * This function is called to remove the sound device for one SSI. We + * de-program the DMACR and PMUXCR register. + */ +static int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) +{ + struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct mpc8610_hpcd_data *machine_data = + container_of(card, struct mpc8610_hpcd_data, card); + struct ccsr_guts __iomem *guts; + + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + if (!guts) { + dev_err(card->dev, "could not map global utilities\n"); + return -ENOMEM; + } + + /* Restore the signal routing */ + + guts_set_dmacr(guts, machine_data->dma_id[0], + machine_data->dma_channel_id[0], 0); + guts_set_dmacr(guts, machine_data->dma_id[1], + machine_data->dma_channel_id[1], 0); + + switch (machine_data->ssi_id) { + case 0: + clrsetbits_be32(&guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + break; + case 1: + clrsetbits_be32(&guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA); + break; + } + + iounmap(guts); + + return 0; +} + +/** + * mpc8610_hpcd_ops: ASoC machine driver operations + */ +static struct snd_soc_ops mpc8610_hpcd_ops = { + .startup = mpc8610_hpcd_startup, +}; + +/** + * get_node_by_phandle_name - get a node by its phandle name + * + * This function takes a node, the name of a property in that node, and a + * compatible string. Assuming the property is a phandle to another node, + * it returns that node, (optionally) if that node is compatible. + * + * If the property is not a phandle, or the node it points to is not compatible + * with the specific string, then NULL is returned. + */ +static struct device_node *get_node_by_phandle_name(struct device_node *np, + const char *name, + const char *compatible) +{ + const phandle *ph; + int len; + + ph = of_get_property(np, name, &len); + if (!ph || (len != sizeof(phandle))) + return NULL; + + np = of_find_node_by_phandle(*ph); + if (!np) + return NULL; + + if (compatible && !of_device_is_compatible(np, compatible)) { + of_node_put(np); + return NULL; + } + + return np; +} + +/** + * get_parent_cell_index -- return the cell-index of the parent of a node + * + * Return the value of the cell-index property of the parent of the given + * node. This is used for DMA channel nodes that need to know the DMA ID + * of the controller they are on. + */ +static int get_parent_cell_index(struct device_node *np) +{ + struct device_node *parent = of_get_parent(np); + const u32 *iprop; + + if (!parent) + return -1; + + iprop = of_get_property(parent, "cell-index", NULL); + of_node_put(parent); + + if (!iprop) + return -1; + + return *iprop; +} + +static int get_dma_channel(struct device_node *ssi_np, + const char *compatible, + dma_addr_t ssi_stx_phys, + dma_addr_t ssi_srx_phys, + struct snd_soc_dai_link *dai, + unsigned int *dma_channel_id, + unsigned int *dma_id) +{ + struct device_node *dma_channel_np; + const u32 *iprop; + + dma_channel_np = get_node_by_phandle_name(ssi_np, compatible, + "fsl,ssi-dma-channel"); + if (!dma_channel_np) + return -EINVAL; + + dai->platform_drv = fsl_soc_dma_to_dai(dma_channel_np->full_name, + ssi_stx_phys, ssi_srx_phys); + + iprop = of_get_property(dma_channel_np, "cell-index", NULL); + if (!iprop) { + of_node_put(dma_channel_np); + return -EINVAL; + } + + *dma_channel_id = *iprop; + *dma_id = get_parent_cell_index(dma_channel_np); + of_node_put(dma_channel_np); + + return 0; +} +/** + * mpc8610_hpcd_probe: platform probe function for the machine driver + * + * Although this is a machine driver, the SSI node is the "master" node with + * respect to audio hardware connections. Therefore, we create a new ASoC + * device for each new SSI node that has a codec attached. + */ +static int mpc8610_hpcd_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_driver **platform_data = pdev->dev.platform_data; + struct snd_soc_dai_driver *cpu_dai_drv = *platform_data; + struct device *dev = pdev->dev.parent; + struct of_device *of_dev = container_of(dev, struct of_device, dev); + struct device_node *np = of_dev->node; + struct device_node *codec_np = NULL; + struct device_node *dma_np = NULL; + struct device_node *dma_channel_np = NULL; + struct platform_device *sound_device = NULL; + struct mpc8610_hpcd_data *machine_data; + int id; + dma_addr_t ssi_stx_phys; + dma_addr_t ssi_srx_phys; + int ret = -ENODEV; + const char *sprop; + const u32 *iprop; + struct resource res; + + /* We are only interested in SSIs with a codec phandle in them, + * so let's make sure this SSI has one. The MPC8610 HPCD only + * knows about the CS4270 codec, so reject anything else. + */ + codec_np = get_node_by_phandle_name(np, "codec-handle", + "cirrus,cs4270"); + if (!codec_np) { + dev_err(dev, "invalid codec node\n"); + return -EINVAL; + } + + machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); + if (!machine_data) + return -ENOMEM; + + machine_data->dai[0].cpu_dai_id = 0; + machine_data->dai[0].codec_dai_id = 0; + machine_data->dai[0].platform_id = 0; + + machine_data->dai[0].name = machine_data->dai_name; + + machine_data->dai[0].cpu_dai_drv = cpu_dai_drv; + machine_data->dai[0].codec_dai_drv = &cs4270_dai; + machine_data->dai[0].codec_drv = &soc_codec_device_cs4270; + machine_data->dai[0].ops = &mpc8610_hpcd_ops; + + /* Store the codec name, it will be used as the codec DAI name */ + of_modalias_node(codec_np, machine_data->dai_name, + sizeof(machine_data->dai_name)); + + memcpy(&machine_data->dai[1], &machine_data->dai[0], + sizeof(struct snd_soc_dai_link)); + + /* Get the device ID */ + iprop = of_get_property(np, "cell-index", NULL); + if (!iprop) { + dev_err(&pdev->dev, "cell-index property not found\n"); + ret = -EINVAL; + goto error; + } + machine_data->ssi_id = *iprop; + id = *iprop; + + /* Get the serial format and clock direction. */ + sprop = of_get_property(np, "fsl,mode", NULL); + if (!sprop) { + dev_err(&pdev->dev, "fsl,mode property not found\n"); + ret = -EINVAL; + goto error; + } + + if (strcasecmp(sprop, "i2s-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + + /* In i2s-slave mode, the codec has its own clock source, so we + * need to get the frequency from the device tree and pass it to + * the codec driver. + */ + iprop = of_get_property(codec_np, "clock-frequency", NULL); + if (!iprop || !*iprop) { + dev_err(&pdev->dev, "codec bus-frequency " + "property is missing or invalid\n"); + ret = -EINVAL; + goto error; + } + machine_data->clk_frequency = *iprop; + } else if (strcasecmp(sprop, "i2s-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "lj-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "lj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "rj-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "rj-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "ac97-slave") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "ac97-master") == 0) { + machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; + machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else { + dev_err(&pdev->dev, + "unrecognized fsl,mode property '%s'\n", sprop); + ret = -EINVAL; + goto error; + } + + if (!machine_data->clk_frequency) { + dev_err(&pdev->dev, "unknown clock frequency\n"); + ret = -EINVAL; + goto error; + } + + /* Read the SSI information from the device tree */ + ret = of_address_to_resource(np, 0, &res); + if (ret || !res.start) { + dev_err(&pdev->dev, "could not obtain SSI address\n"); + goto error; + } + ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); + ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); + + /* Find the playback DMA channel to use. */ + ret = get_dma_channel(np, "fsl,playback-dma", ssi_stx_phys, + ssi_srx_phys, &machine_data->dai[0], + &machine_data->dma_channel_id[0], + &machine_data->dma_id[0]); + if (ret) { + dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); + goto error; + } + + /* Find the capture DMA channel to use. */ + ret = get_dma_channel(np, "fsl,capture-dma", ssi_stx_phys, + ssi_srx_phys, &machine_data->dai[1], + &machine_data->dma_channel_id[1], + &machine_data->dma_id[1]); + if (ret) { + dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); + goto error; + } + + /* Initialize our DAI data structure. */ + iprop = of_get_property(codec_np, "reg", NULL); + if (!iprop) { + dev_err(&pdev->dev, "codec node is missing 'reg' property\n"); + goto error; + } + machine_data->dai[0].codec_id = *iprop; + machine_data->dai[1].codec_id = *iprop; + + machine_data->dai[0].stream_name = "playback"; + machine_data->dai[1].stream_name = "capture"; + + machine_data->card.probe = mpc8610_hpcd_machine_probe; + machine_data->card.remove = mpc8610_hpcd_machine_remove; + machine_data->card.name = "MPC8610 HPCD"; + machine_data->card.num_links = 2; + machine_data->card.dai_link = machine_data->dai; + + /* Allocate a new audio platform device structure */ + sound_device = platform_device_alloc("soc-audio", -1); + if (!sound_device) { + dev_err(&pdev->dev, "platform device alloc failed\n"); + ret = -ENOMEM; + goto error; + } + + /* Associate the card data with the sound device */ + platform_set_drvdata(sound_device, &machine_data->card); + + /* Register with ASoC */ + ret = platform_device_add(sound_device); + if (ret) { + dev_err(&pdev->dev, "platform device add failed\n"); + goto error; + } + + of_node_put(codec_np); + + return 0; + +error: + of_node_put(codec_np); + of_node_put(dma_np); + of_node_put(dma_channel_np); + + if (sound_device) + platform_device_unregister(sound_device); + + kfree(machine_data); + + return ret; +} + +/** + * mpc8610_hpcd_remove: remove the platform device + * + * This function is called when the platform device is removed. + */ +static int __devexit mpc8610_hpcd_remove(struct platform_device *pdev) +{ + struct platform_device *sound_device = dev_get_drvdata(&pdev->dev); + struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct mpc8610_hpcd_data *machine_data = + container_of(card, struct mpc8610_hpcd_data, card); + + platform_device_unregister(sound_device); + + kfree(machine_data); + sound_device->dev.platform_data = NULL; + + dev_set_drvdata(&pdev->dev, NULL); + + return 0; +} + +static struct platform_driver mpc8610_hpcd_driver = { + .probe = mpc8610_hpcd_probe, + .remove = __devexit_p(mpc8610_hpcd_remove), + .driver = { + /* The name must match the 'model' property in the device tree, + * in lowercase letters. + */ + .name = "snd-soc-mpc8610hpcd", + .owner = THIS_MODULE, + }, +}; + +/** + * mpc8610_hpcd_init: machine driver initialization. + * + * This function is called when this module is loaded. + */ +static int __init mpc8610_hpcd_init(void) +{ + struct device_node *guts_np; + struct resource res; + + pr_info("Freescale MPC8610 HPCD ALSA SoC machine driver\n"); + + /* Get the physical address of the global utilities registers */ + guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); + if (of_address_to_resource(guts_np, 0, &res)) { + pr_err("mpc8610-hpcd: missing/invalid global utilities node\n"); + return -EINVAL; + } + guts_phys = res.start; + + return platform_driver_register(&mpc8610_hpcd_driver); +} + +/** + * mpc8610_hpcd_exit: machine driver exit + * + * This function is called when this driver is unloaded. + */ +static void __exit mpc8610_hpcd_exit(void) +{ + platform_driver_unregister(&mpc8610_hpcd_driver); +} + +module_init(mpc8610_hpcd_init); +module_exit(mpc8610_hpcd_exit); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC machine driver"); +MODULE_LICENSE("GPL v2"); -- 1.6.5

1 0

[alsa-devel] How to play a sine wave in alsa?
by Timo Schneider 04 May '10

04 May '10

Hello! I am trying to use alsa to play a sine wave in async mode. Everything seems to work fine, except what I am hearing is _not_ a sine wave. I am checking for XRUNs, but none occur, so I guess the problem is that I am supplying the wrong sample data to alsa. I want the tone only on one channel, that seems to work fine the way i do it, so i guess my assumptions on how the interleaved mode works are not completely wrong. Can anyone give me a hint on what I am doing wrong in the example program below? I already checked the example in tests/pcm.c but it didn't really help me. Regards, Timo Example Program (compile with cc test.c -o test -lasound): #include <math.h> #include <stdlib.h> #include <inttypes.h> #include <alsa/asoundlib.h> struct sound_settings { // alsa parameters snd_pcm_t *pcm_handle; snd_pcm_hw_params_t *hw_params; snd_pcm_sw_params_t *sw_params; snd_async_handler_t *pcm_callback; snd_pcm_uframes_t buffer_size; snd_pcm_uframes_t period_size; unsigned int sample_rate; char *sample_buffer; size_t sample_buffer_size; // user settings made in gui char *cardname; char *modename; int startfreq; int endfreq; int duration; int channels; // keep track of gnerated samples long int actsample; long int lastsample; }; struct sound_settings sdata; void generate_samples(struct sound_settings *sdata) { int i = 0; if (sdata->channels == 1) i=0; if (sdata->channels == 2) i=1; if (sdata->channels == 3) i=0; if (strcmp(sdata->modename, "Sine")==0) { while (i<(sdata->sample_buffer_size / 8)) { int samplerate = 44100; int freq = sdata->startfreq; double sample = sin(sdata->actsample * (double) 2*M_PI / ((double) samplerate / (double)freq)); int16_t rsamp = sample * 65535; sdata->actsample++; *(((int16_t*) (sdata->sample_buffer))+i) = rsamp; i += 2; } } } void MyCallback(snd_async_handler_t *pcm_callback) { snd_pcm_t *pcm_handle = snd_async_handler_get_pcm(pcm_callback); snd_pcm_sframes_t avail; struct sound_settings *sdata = (struct sound_settings*) snd_async_handler_get_callback_private(pcm_callback); avail = snd_pcm_avail_update(pcm_handle); if (avail == -EPIPE) { snd_pcm_prepare (pcm_handle); printf("Underrun occured\n"); } while (avail >= sdata->period_size) { generate_samples(sdata); snd_pcm_writei(pcm_handle, sdata->sample_buffer, sdata->period_size); avail = snd_pcm_avail_update(pcm_handle); } } void init_alsa(struct sound_settings *sdata) { int rc; // set some sound_settings data (i.e., sample_rate) to sensible defaults sdata->buffer_size = 1024; sdata->period_size = 64; sdata->sample_rate = 44100; // Open PCM device for playback rc = snd_pcm_open(&(sdata->pcm_handle), "default", SND_PCM_STREAM_PLAYBACK, 0); if (rc < 0) fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); // Allocate a hardware parameters object snd_pcm_hw_params_malloc (&(sdata->hw_params)); snd_pcm_hw_params_any(sdata->pcm_handle, sdata->hw_params); snd_pcm_hw_params_set_access(sdata->pcm_handle, sdata->hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(sdata->pcm_handle, sdata->hw_params, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(sdata->pcm_handle, sdata->hw_params, 2); snd_pcm_hw_params_set_rate_near(sdata->pcm_handle, sdata->hw_params, &(sdata->sample_rate), NULL); snd_pcm_hw_params_set_buffer_size_near(sdata->pcm_handle, sdata->hw_params, &(sdata->buffer_size)); snd_pcm_hw_params_set_period_size_near (sdata->pcm_handle, sdata->hw_params, &(sdata->period_size), NULL); rc = snd_pcm_hw_params(sdata->pcm_handle, sdata->hw_params); if (rc < 0) fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); // tell alsa we want to work asynchronously snd_pcm_sw_params_malloc(&(sdata->sw_params)); snd_pcm_sw_params_current(sdata->pcm_handle, sdata->sw_params); snd_pcm_sw_params_set_start_threshold(sdata->pcm_handle, sdata->sw_params, sdata->buffer_size - sdata->period_size); snd_pcm_sw_params_set_avail_min(sdata->pcm_handle, sdata->sw_params, sdata->period_size); rc = snd_pcm_sw_params(sdata->pcm_handle, sdata->sw_params); if (rc < 0) fprintf(stderr, "unable to set sw parameters: %s\n", snd_strerror(rc)); // Use a buffer large enough to hold two periods snd_pcm_hw_params_get_period_size(sdata->hw_params, &(sdata->period_size), NULL); sdata->sample_buffer_size = sdata->period_size * 8; // 2 bytes/sample, 2 channels, two periods sdata->sample_buffer = (char *) malloc(sdata->sample_buffer_size); generate_samples(sdata); // prepare dev so that we can write snd_pcm_prepare(sdata->pcm_handle); /* fill buffer with some data */ generate_samples(sdata); /* write first chunk of data */ rc = snd_pcm_writei(sdata->pcm_handle, sdata->sample_buffer, sdata->period_size); if (rc == -EPIPE) snd_pcm_prepare(sdata->pcm_handle); /* register callback */ rc = snd_async_add_pcm_handler(&(sdata->pcm_callback), sdata->pcm_handle, MyCallback, sdata); snd_pcm_start(sdata->pcm_handle); } int main(int argc, char **argv) { sdata.channels = 1; sdata.modename = "Sine"; sdata.startfreq = 220; sdata.actsample = 0; init_alsa(&sdata); static int i=0; while (i<1000000) { i++; sleep(1); } return 0; }

1 0

Re: [alsa-devel] Ooops when working with USB MIDI (2.6.33.1)
by Takashi Iwai 03 May '10

03 May '10

At Fri, 9 Apr 2010 07:51:35 +0100, Tvrtko Ursulin wrote: > > On Thursday 08 Apr 2010 13:22:36 Takashi Iwai wrote: > > > Takashi, do you remember what the original problem was? > > > > Well, I have only a vague memory -- it's a similar scenario that some app > > still accessing after disconnection. The URB can't be handled after > > the disconnection is finished. > > > > I think the patch below might fix in this case. You can try it > > instead of reverting the commit above. > > > > > > thanks, > > > > Takashi > > > > --- > > diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c > > index 2c59afd..81c8d85 100644 > > --- a/sound/usb/usbmidi.c > > +++ b/sound/usb/usbmidi.c > > @@ -986,6 +986,8 @@ static void snd_usbmidi_output_drain(struct > > snd_rawmidi_substream *substream) DEFINE_WAIT(wait); > > long timeout = msecs_to_jiffies(50); > > > > + if (ep->umidi->disconnected) > > + return; > > /* > > * The substream buffer is empty, but some data might still be in the > > * currently active URBs, so we have to wait for those to complete. > > @@ -1275,6 +1277,11 @@ void snd_usbmidi_disconnect(struct list_head* p) > > snd_usbmidi_in_endpoint_delete(ep->in); > > ep->in = NULL; > > } > > + ep->active_urbs = 0; > > + if (ep->drain_urbs) { > > + ep->drain_urbs = 0; > > + wake_up(&ep->drain_wait); > > + } > > } > > del_timer_sync(&umidi->error_timer); > > } > > For the second hunk, do you think ep->out->... and so on? That would be more > in-line with code present in 2.6.33. Ah, crap. Sorry, that's just messing up. The revised (compiled but untested) patch is below. thanks, Takashi --- diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 2c59afd..9e28b20 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -986,6 +986,8 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) DEFINE_WAIT(wait); long timeout = msecs_to_jiffies(50); + if (ep->umidi->disconnected) + return; /* * The substream buffer is empty, but some data might still be in the * currently active URBs, so we have to wait for those to complete. @@ -1123,14 +1125,21 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, * Frees an output endpoint. * May be called when ep hasn't been initialized completely. */ -static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_out_endpoint_clear(struct snd_usb_midi_out_endpoint *ep) { unsigned int i; for (i = 0; i < OUTPUT_URBS; ++i) - if (ep->urbs[i].urb) + if (ep->urbs[i].urb) { free_urb_and_buffer(ep->umidi, ep->urbs[i].urb, ep->max_transfer); + ep->urbs[i].urb = NULL; + } +} + +static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint *ep) +{ + snd_usbmidi_out_endpoint_clear(ep); kfree(ep); } @@ -1262,15 +1271,18 @@ void snd_usbmidi_disconnect(struct list_head* p) usb_kill_urb(ep->out->urbs[j].urb); if (umidi->usb_protocol_ops->finish_out_endpoint) umidi->usb_protocol_ops->finish_out_endpoint(ep->out); + ep->out->active_urbs = 0; + if (ep->out->drain_urbs) { + ep->out->drain_urbs = 0; + wake_up(&ep->out->drain_wait); + } } if (ep->in) for (j = 0; j < INPUT_URBS; ++j) usb_kill_urb(ep->in->urbs[j]); /* free endpoints here; later call can result in Oops */ - if (ep->out) { - snd_usbmidi_out_endpoint_delete(ep->out); - ep->out = NULL; - } + if (ep->out) + snd_usbmidi_out_endpoint_clear(ep->out); if (ep->in) { snd_usbmidi_in_endpoint_delete(ep->in); ep->in = NULL;

2 3

[alsa-devel] [PATCHv2 0/4] ASoC: tlv320dac33: Support for turning off codec
by Peter Ujfalusi 03 May '10

03 May '10

Hello, Changes since v1: - patch 1: Removed extra register writes - patch 4: - Let core to bring the codec to BIAS_OFF (set idle_bias_off) - dac33_set_bias_level only takes care of power and minimum initialization - PRE_DAPM widget event is replacing the dac33_pcm_pointer call to take care of the initialization needed before _every_ playback. There are few scenarios, which has to be taken care:: 1. Analog bypass caused BIAS_OFF -> BIAS_ON We need to power on the codec, and do the chip init, but we does not need to execute the playback related configuration 2. Playback caused BIAS_OFF -> BIAS_ON We need to power on the codec, and do the chip init, and also we need to execute the playback related configuration. 3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is already on. 4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON) Nothing need to be done. 5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is still on. Since the power up, and the codec init is optimized, the added overhead in stream start is minimal. Withing this patch, the hard_power function is now only doing what it supposed to: only handle the powers, and GPIO reset line. The codec initialization and state restore has been moved out. --- Peter Ujfalusi (4): ASoC: tlv320dac33: Optimize power up, and restore ASoC: tlv320dac33: Revised module loading, and DAC33 ID read ASoC: tlv320dac33: Manage a pointer for snd_pcm_substream in private structure ASoC: tlv320dac33: Support for turning off the codec sound/soc/codecs/tlv320dac33.c | 235 ++++++++++++++++++++++------------------ 1 files changed, 129 insertions(+), 106 deletions(-)

3 6

[alsa-devel] [PATCHv3 0/2] ASoC: TWL4030: APLL/AIF ordering and DAPM updates
by Peter Ujfalusi 03 May '10

03 May '10

Hello, Changes since v2: apll_enable logic cleaned up. aif refcounting has been removed, aif handling simplified The new DAPM_OUTPUT and DAPM_INPUT widgets renamed Correct the APLL/AIF powering sequence. Also introducing DAPM outputs and input for keeping APLL/AIF enabled in case when no valid DAPM route selected during audio activity. The second patch removes the legacy OUTL/R. As a note: The APLL/AIF patch replaces the same patch in Liam's series. The OUTL/R removal can be applied after the OUTL/R reference has been removed from machine drivers (as in Lima's series). --- Peter Ujfalusi (2): ASoC: TWL4030: AIF/APLL fix in DAPM domain ASoC: TWL4030: Remove OUTL/R outputs sound/soc/codecs/twl4030.c | 86 ++++++++++++++++++++++++++++++------------- 1 files changed, 60 insertions(+), 26 deletions(-)

3 5

[alsa-devel] Changing volume of a single PCM channel
by Andreas Falkenhahn 03 May '10

03 May '10

Hi, how do I change the volume of single PCM channel with ALSA? I'm playing multiple sounds at the same time by opening a new channel for every sound using snd_pcm_open() and let PulseAudio do the mixing. Now assuming that I have allocated four channels using snd_pcm_open()... now I'd like to change the volume of the second channel, the other three channels' volume should not be affected. How can I do this? AFAICS, volume control is provided by the ALSA mixer interface but I don't seem to figure out how to connect the mixer with my PCM channels opened using snd_pcm_open(). Thanks for any help, Andreas

1 0

[alsa-devel] Reforming S3C I2S towards supporting I2Sv4
by Jassi Brar 02 May '10

02 May '10

Hello, I have prepared some patches that debug and modify the behaviour of extant S3C I2S support towards managing I2Sv4 controllers as well. Overall pic... For now the 'new' I2Sv4 CPU driver(s3c64xx-i2s-v4.c) is almost a copy of s3c64xx-i2s.c The driver will evolve as support for more features is added. Minor quirks in otherwise similar controllers are handled by defining a new field(feature) in 'struct s3c_i2sv2_info' that is meant to contain 1-bit flags for absence/presence of such quirks. The header with I2S register map and bit definitions has been copied to where the drivers are(sound/soc/s3c24xx/) since the header has nothing usable for platform code. Also, it will help avoid need for co-ordination between ASoC and S3C ARCH trees. For now, the header regs-s3c2412-iis.h is left intact but rendered useless by making ASoC drivers include the newly copied version of it (sound/soc/s3c24xx/regs-i2s-v2.h) Later the header could be dropped by patches to S3C PLAT tree. The patches apply against 'for-2.6.35' of http://www.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git Regards. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com>

6 16

[alsa-devel] [PATCH] Merge es1688 and es968 drivers
by Krzysztof Helt 01 May '10

01 May '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The ESS ES968 chip is nothing more then a PnP companion for non-PnP audio chips. It was paired with non-PnP ESS' chips: ES688 and ES1688. The ESS' audio chips are handled by the es1688 driver in native mode. The PnP cards are handled by the ES968 driver in SB compatible mode. Move the ES968 chip handling to the es1688 driver so the driver can handle both PnP and non-PnP cards. The es968 is removed. Also, a new PnP id is added for the card I acquired (the change was tested on this card). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Documentation/sound/alsa/ALSA-Configuration.txt | 21 +-- include/sound/es1688.h | 8 +- sound/isa/Kconfig | 16 +-- sound/isa/es1688/es1688.c | 204 ++++++++++++++++--- sound/isa/es1688/es1688_lib.c | 26 +-- sound/isa/sb/Makefile | 2 - sound/isa/sb/es968.c | 248 ----------------------- 7 files changed, 198 insertions(+), 327 deletions(-) delete mode 100644 sound/isa/sb/es968.c diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index bfcbbf8..448cff9 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -622,28 +622,23 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. The power-management is supported. - Module snd-es968 - ---------------- - - Module for sound cards based on ESS ES968 chip (PnP only). - - This module supports multiple cards, PnP and autoprobe. - - The power-management is supported. - Module snd-es1688 ----------------- Module for ESS AudioDrive ES-1688 and ES-688 sound cards. - port - port # for ES-1688 chip (0x220,0x240,0x260) - fm_port - port # for OPL3 (option; share the same port as default) + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) - irq - IRQ # for ES-1688 chip (5,7,9,10) mpu_irq - IRQ # for MPU-401 port (5,7,9,10) + fm_port - port # for OPL3 (option; share the same port as default) + + with isapnp=0, the following additional options are available: + port - port # for ES-1688 chip (0x220,0x240,0x260) + irq - IRQ # for ES-1688 chip (5,7,9,10) dma8 - DMA # for ES-1688 chip (0,1,3) - This module supports multiple cards and autoprobe (without MPU-401 port). + This module supports multiple cards and autoprobe (without MPU-401 port) + and PnP with the ES968 chip. Module snd-es18xx ----------------- diff --git a/include/sound/es1688.h b/include/sound/es1688.h index 10fcf14..10d7951 100644 --- a/include/sound/es1688.h +++ b/include/sound/es1688.h @@ -44,7 +44,6 @@ struct snd_es1688 { unsigned char pad; unsigned int dma_size; - struct snd_card *card; struct snd_pcm *pcm; struct snd_pcm_substream *playback_substream; struct snd_pcm_substream *capture_substream; @@ -113,9 +112,8 @@ int snd_es1688_create(struct snd_card *card, int irq, int mpu_irq, int dma8, - unsigned short hardware, - struct snd_es1688 ** rchip); -int snd_es1688_pcm(struct snd_es1688 *chip, int device, struct snd_pcm ** rpcm); -int snd_es1688_mixer(struct snd_es1688 *chip); + unsigned short hardware); +int snd_es1688_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm); +int snd_es1688_mixer(struct snd_card *card); #endif /* __SOUND_ES1688_H */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 755a0a5..c6990c6 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -128,26 +128,14 @@ config SND_CS4236 To compile this driver as a module, choose M here: the module will be called snd-cs4236. -config SND_ES968 - tristate "Generic ESS ES968 driver" - depends on PNP - select ISAPNP - select SND_MPU401_UART - select SND_SB8_DSP - help - Say Y here to include support for ESS AudioDrive ES968 chips. - - To compile this driver as a module, choose M here: the module - will be called snd-es968. - config SND_ES1688 - tristate "Generic ESS ES688/ES1688 driver" + tristate "Generic ESS ES688/ES1688 and ES968 PnP driver" select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help Say Y here to include support for ESS AudioDrive ES688 or - ES1688 chips. + ES1688 chips. Also, this module support cards with ES968 PnP chip. To compile this driver as a module, choose M here: the module will be called snd-es1688. diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 07df201..4b5cc8c 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -22,6 +22,7 @@ #include <linux/init.h> #include <linux/err.h> #include <linux/isa.h> +#include <linux/isapnp.h> #include <linux/time.h> #include <linux/wait.h> #include <linux/moduleparam.h> @@ -47,6 +48,9 @@ MODULE_SUPPORTED_DEVICE("{{ESS,ES688 PnP AudioDrive,pnp:ESS0100}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +#ifdef CONFIG_PNP +static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; +#endif static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */ static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* Usually 0x388 */ @@ -60,6 +64,10 @@ MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); +#ifdef CONFIG_PNP +module_param_array(isapnp, bool, NULL, 0444); +MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); +#endif MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); module_param_array(port, long, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); @@ -74,13 +82,19 @@ MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); module_param_array(dma8, int, NULL, 0444); MODULE_PARM_DESC(dma8, "8-bit DMA # for " CRD_NAME " driver."); +#ifdef CONFIG_PNP +#define is_isapnp_selected(dev) isapnp[dev] +#else +#define is_isapnp_selected(dev) 0 +#endif + static int __devinit snd_es1688_match(struct device *dev, unsigned int n) { - return enable[n]; + return enable[n] && !is_isapnp_selected(n); } -static int __devinit snd_es1688_legacy_create(struct snd_card *card, - struct device *dev, unsigned int n, struct snd_es1688 **rchip) +static int __devinit snd_es1688_legacy_create(struct snd_card *card, + struct device *dev, unsigned int n) { static long possible_ports[] = {0x220, 0x240, 0x260}; static int possible_irqs[] = {5, 9, 10, 7, -1}; @@ -105,41 +119,32 @@ static int __devinit snd_es1688_legacy_create(struct snd_card *card, if (port[n] != SNDRV_AUTO_PORT) return snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip); + mpu_irq[n], dma8[n], ES1688_HW_AUTO); i = 0; do { port[n] = possible_ports[i]; error = snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip); + mpu_irq[n], dma8[n], ES1688_HW_AUTO); } while (error < 0 && ++i < ARRAY_SIZE(possible_ports)); return error; } -static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) +static int __devinit snd_es1688_probe(struct snd_card *card, unsigned int n) { - struct snd_card *card; - struct snd_es1688 *chip; + struct snd_es1688 *chip = card->private_data; struct snd_opl3 *opl3; struct snd_pcm *pcm; int error; - error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + error = snd_es1688_pcm(card, 0, &pcm); if (error < 0) return error; - error = snd_es1688_legacy_create(card, dev, n, &chip); - if (error < 0) - goto out; - - error = snd_es1688_pcm(chip, 0, &pcm); - if (error < 0) - goto out; - - error = snd_es1688_mixer(chip); + error = snd_es1688_mixer(card); if (error < 0) - goto out; + return error; strcpy(card->driver, "ES1688"); strcpy(card->shortname, pcm->name); @@ -152,12 +157,12 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) if (fm_port[n] > 0) { if (snd_opl3_create(card, fm_port[n], fm_port[n] + 2, OPL3_HW_OPL3, 0, &opl3) < 0) - dev_warn(dev, + dev_warn(card->dev, "opl3 not detected at 0x%lx\n", fm_port[n]); else { error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (error < 0) - goto out; + return error; } } @@ -167,23 +172,41 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) chip->mpu_port, 0, mpu_irq[n], IRQF_DISABLED, NULL); if (error < 0) - goto out; + return error; } + return snd_card_register(card); +} + +static int __devinit snd_es1688_isa_probe(struct device *dev, unsigned int n) +{ + struct snd_card *card; + int error; + + error = snd_card_create(index[n], id[n], THIS_MODULE, + sizeof(struct snd_es1688), &card); + if (error < 0) + return error; + + error = snd_es1688_legacy_create(card, dev, n); + if (error < 0) + goto out; + snd_card_set_dev(card, dev); - error = snd_card_register(card); + error = snd_es1688_probe(card, n); if (error < 0) goto out; dev_set_drvdata(dev, card); - return 0; -out: snd_card_free(card); + return 0; +out: + snd_card_free(card); return error; } -static int __devexit snd_es1688_remove(struct device *dev, unsigned int n) +static int __devexit snd_es1688_isa_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); dev_set_drvdata(dev, NULL); @@ -192,8 +215,8 @@ static int __devexit snd_es1688_remove(struct device *dev, unsigned int n) static struct isa_driver snd_es1688_driver = { .match = snd_es1688_match, - .probe = snd_es1688_probe, - .remove = __devexit_p(snd_es1688_remove), + .probe = snd_es1688_isa_probe, + .remove = __devexit_p(snd_es1688_isa_remove), #if 0 /* FIXME */ .suspend = snd_es1688_suspend, .resume = snd_es1688_resume, @@ -203,14 +226,137 @@ static struct isa_driver snd_es1688_driver = { } }; +static int snd_es968_pnp_is_probed; + +#ifdef CONFIG_PNP +static int __devinit snd_card_es968_pnp(struct snd_card *card, unsigned int n, + struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + struct pnp_dev *pdev; + int error; + + pdev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); + if (pdev == NULL) + return -ENODEV; + + error = pnp_activate_dev(pdev); + if (error < 0) { + snd_printk(KERN_ERR "ES968 pnp configure failure\n"); + return error; + } + port[n] = pnp_port_start(pdev, 0); + dma8[n] = pnp_dma(pdev, 0); + irq[n] = pnp_irq(pdev, 0); + + return snd_es1688_create(card, port[n], mpu_port[n], irq[n], + mpu_irq[n], dma8[n], ES1688_HW_AUTO); +} + +static int __devinit snd_es968_pnp_detect(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + struct snd_card *card; + static unsigned int dev; + int error; + struct snd_es1688 *chip; + + if (snd_es968_pnp_is_probed) + return -EBUSY; + for ( ; dev < SNDRV_CARDS; dev++) { + if (enable[dev] && isapnp[dev]) + break; + } + + error = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_es1688), &card); + if (error < 0) + return error; + chip = card->private_data; + + error = snd_card_es968_pnp(card, dev, pcard, pid); + if (error < 0) { + snd_card_free(card); + return error; + } + snd_card_set_dev(card, &pcard->card->dev); + error = snd_es1688_probe(card, dev); + if (error < 0) + return error; + pnp_set_card_drvdata(pcard, card); + snd_es968_pnp_is_probed = 1; + return 0; +} + +static void __devexit snd_es968_pnp_remove(struct pnp_card_link * pcard) +{ + snd_card_free(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); + snd_es968_pnp_is_probed = 0; +} + +#ifdef CONFIG_PM +static int snd_es968_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_es1688 *chip = card->private_data; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(chip->pcm); + return 0; +} + +static int snd_es968_pnp_resume(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + +static struct pnp_card_device_id snd_es968_pnpids[] = { + { .id = "ESS0968", .devs = { { "@@@0968" }, } }, + { .id = "ESS0968", .devs = { { "ESS0968" }, } }, + { .id = "", } /* end */ +}; + +MODULE_DEVICE_TABLE(pnp_card, snd_es968_pnpids); + +static struct pnp_card_driver es968_pnpc_driver = { + .flags = PNP_DRIVER_RES_DISABLE, + .name = DEV_NAME " PnP", + .id_table = snd_es968_pnpids, + .probe = snd_es968_pnp_detect, + .remove = __devexit_p(snd_es968_pnp_remove), +#ifdef CONFIG_PM + .suspend = snd_es968_pnp_suspend, + .resume = snd_es968_pnp_resume, +#endif +}; +#endif + static int __init alsa_card_es1688_init(void) { +#ifdef CONFIG_PNP + pnp_register_card_driver(&es968_pnpc_driver); + if (snd_es968_pnp_is_probed) + return 0; + pnp_unregister_card_driver(&es968_pnpc_driver); +#endif return isa_register_driver(&snd_es1688_driver, SNDRV_CARDS); } static void __exit alsa_card_es1688_exit(void) { - isa_unregister_driver(&snd_es1688_driver); + if (!snd_es968_pnp_is_probed) { + isa_unregister_driver(&snd_es1688_driver); + return; + } +#ifdef CONFIG_PNP + pnp_unregister_card_driver(&es968_pnpc_driver); +#endif } module_init(alsa_card_es1688_init); diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index c76bb00..9632dda 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -643,20 +643,14 @@ int snd_es1688_create(struct snd_card *card, int irq, int mpu_irq, int dma8, - unsigned short hardware, - struct snd_es1688 **rchip) + unsigned short hardware) { static struct snd_device_ops ops = { .dev_free = snd_es1688_dev_free, }; - - struct snd_es1688 *chip; + struct snd_es1688 *chip = card->private_data; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; chip->irq = -1; chip->dma8 = -1; @@ -680,7 +674,6 @@ int snd_es1688_create(struct snd_card *card, spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); - chip->card = card; chip->port = port; mpu_port &= ~0x000f; if (mpu_port < 0x300 || mpu_port > 0x330) @@ -704,7 +697,6 @@ int snd_es1688_create(struct snd_card *card, return err; } - *rchip = chip; return 0; } @@ -730,12 +722,14 @@ static struct snd_pcm_ops snd_es1688_capture_ops = { .pointer = snd_es1688_capture_pointer, }; -int snd_es1688_pcm(struct snd_es1688 * chip, int device, struct snd_pcm ** rpcm) +int snd_es1688_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; + struct snd_es1688 *chip = card->private_data; int err; - if ((err = snd_pcm_new(chip->card, "ESx688", device, 1, 1, &pcm)) < 0) + err = snd_pcm_new(card, "ESx688", device, 1, 1, &pcm); + if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_es1688_playback_ops); @@ -1009,17 +1003,17 @@ static unsigned char snd_es1688_init_table[][2] = { { ES1688_REC_DEV, 0x17 } }; -int snd_es1688_mixer(struct snd_es1688 *chip) +int snd_es1688_mixer(struct snd_card *card) { - struct snd_card *card; + struct snd_es1688 *chip; unsigned int idx; int err; unsigned char reg, val; - if (snd_BUG_ON(!chip || !chip->card)) + if (snd_BUG_ON(!card || !card->private_data)) return -EINVAL; - card = chip->card; + chip = card->private_data; strcpy(card->mixername, snd_es1688_chip_id(chip)); diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index af36696..08b9fb9 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -11,7 +11,6 @@ snd-sb8-objs := sb8.o snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o -snd-es968-objs := es968.o snd-jazz16-objs := jazz16.o # Toplevel Module Dependency @@ -21,7 +20,6 @@ obj-$(CONFIG_SND_SB8_DSP) += snd-sb8-dsp.o obj-$(CONFIG_SND_SB8) += snd-sb8.o obj-$(CONFIG_SND_SB16) += snd-sb16.o obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o -obj-$(CONFIG_SND_ES968) += snd-es968.o obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c deleted file mode 100644 index ff18286..0000000 --- a/sound/isa/sb/es968.c +++ /dev/null @@ -1,248 +0,0 @@ - -/* - card-es968.c - driver for ESS AudioDrive ES968 based soundcards. - Copyright (C) 1999 by Massimo Piccioni <dafastidio(a)libero.it> - - Thanks to Pierfrancesco 'qM2' Passerini. - - This program is free software; you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation; either version 2 of the License, or - (at your option) any later version. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with this program; if not, write to the Free Software - Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -*/ - -#include <linux/init.h> -#include <linux/time.h> -#include <linux/pnp.h> -#include <linux/moduleparam.h> -#include <sound/core.h> -#include <sound/initval.h> -#include <sound/sb.h> - -#define PFX "es968: " - -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); -MODULE_DESCRIPTION("ESS AudioDrive ES968"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{ESS,AudioDrive ES968}}"); - -static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ -static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ -static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* Pnp setup */ -static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ - -module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for es968 based soundcard."); -module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for es968 based soundcard."); -module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable es968 based soundcard."); - -struct snd_card_es968 { - struct pnp_dev *dev; - struct snd_sb *chip; -}; - -static struct pnp_card_device_id snd_es968_pnpids[] = { - { .id = "ESS0968", .devs = { { "@@@0968" }, } }, - { .id = "", } /* end */ -}; - -MODULE_DEVICE_TABLE(pnp_card, snd_es968_pnpids); - -#define DRIVER_NAME "snd-card-es968" - -static irqreturn_t snd_card_es968_interrupt(int irq, void *dev_id) -{ - struct snd_sb *chip = dev_id; - - if (chip->open & SB_OPEN_PCM) { - return snd_sb8dsp_interrupt(chip); - } else { - return snd_sb8dsp_midi_interrupt(chip); - } -} - -static int __devinit snd_card_es968_pnp(int dev, struct snd_card_es968 *acard, - struct pnp_card_link *card, - const struct pnp_card_device_id *id) -{ - struct pnp_dev *pdev; - int err; - - acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) - return -ENODEV; - - pdev = acard->dev; - - err = pnp_activate_dev(pdev); - if (err < 0) { - snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); - return err; - } - port[dev] = pnp_port_start(pdev, 0); - dma8[dev] = pnp_dma(pdev, 0); - irq[dev] = pnp_irq(pdev, 0); - - return 0; -} - -static int __devinit snd_card_es968_probe(int dev, - struct pnp_card_link *pcard, - const struct pnp_card_device_id *pid) -{ - int error; - struct snd_sb *chip; - struct snd_card *card; - struct snd_card_es968 *acard; - - error = snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_es968), &card); - if (error < 0) - return error; - acard = card->private_data; - if ((error = snd_card_es968_pnp(dev, acard, pcard, pid))) { - snd_card_free(card); - return error; - } - snd_card_set_dev(card, &pcard->card->dev); - - if ((error = snd_sbdsp_create(card, port[dev], - irq[dev], - snd_card_es968_interrupt, - dma8[dev], - -1, - SB_HW_AUTO, &chip)) < 0) { - snd_card_free(card); - return error; - } - acard->chip = chip; - - if ((error = snd_sb8dsp_pcm(chip, 0, NULL)) < 0) { - snd_card_free(card); - return error; - } - - if ((error = snd_sbmixer_new(chip)) < 0) { - snd_card_free(card); - return error; - } - - if ((error = snd_sb8dsp_midi(chip, 0, NULL)) < 0) { - snd_card_free(card); - return error; - } - - strcpy(card->driver, "ES968"); - strcpy(card->shortname, "ESS ES968"); - sprintf(card->longname, "%s soundcard, %s at 0x%lx, irq %d, dma %d", - card->shortname, chip->name, chip->port, irq[dev], dma8[dev]); - - if ((error = snd_card_register(card)) < 0) { - snd_card_free(card); - return error; - } - pnp_set_card_drvdata(pcard, card); - return 0; -} - -static unsigned int __devinitdata es968_devices; - -static int __devinit snd_es968_pnp_detect(struct pnp_card_link *card, - const struct pnp_card_device_id *id) -{ - static int dev; - int res; - - for ( ; dev < SNDRV_CARDS; dev++) { - if (!enable[dev]) - continue; - res = snd_card_es968_probe(dev, card, id); - if (res < 0) - return res; - dev++; - es968_devices++; - return 0; - } - return -ENODEV; -} - -static void __devexit snd_es968_pnp_remove(struct pnp_card_link * pcard) -{ - snd_card_free(pnp_get_card_drvdata(pcard)); - pnp_set_card_drvdata(pcard, NULL); -} - -#ifdef CONFIG_PM -static int snd_es968_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_es968 *acard = card->private_data; - struct snd_sb *chip = acard->chip; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - snd_pcm_suspend_all(chip->pcm); - snd_sbmixer_suspend(chip); - return 0; -} - -static int snd_es968_pnp_resume(struct pnp_card_link *pcard) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_es968 *acard = card->private_data; - struct snd_sb *chip = acard->chip; - - snd_sbdsp_reset(chip); - snd_sbmixer_resume(chip); - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif - -static struct pnp_card_driver es968_pnpc_driver = { - .flags = PNP_DRIVER_RES_DISABLE, - .name = "es968", - .id_table = snd_es968_pnpids, - .probe = snd_es968_pnp_detect, - .remove = __devexit_p(snd_es968_pnp_remove), -#ifdef CONFIG_PM - .suspend = snd_es968_pnp_suspend, - .resume = snd_es968_pnp_resume, -#endif -}; - -static int __init alsa_card_es968_init(void) -{ - int err = pnp_register_card_driver(&es968_pnpc_driver); - if (err) - return err; - - if (!es968_devices) { - pnp_unregister_card_driver(&es968_pnpc_driver); -#ifdef MODULE - snd_printk(KERN_ERR "no ES968 based soundcards found\n"); -#endif - return -ENODEV; - } - return 0; -} - -static void __exit alsa_card_es968_exit(void) -{ - pnp_unregister_card_driver(&es968_pnpc_driver); -} - -module_init(alsa_card_es968_init) -module_exit(alsa_card_es968_exit) -- 1.6.4 ---------------------------------------------------------------------- Szukasz pracy? Zobacz ciekawe oferty w Twoim miescie Sprawdz >>> http://linkint.pl/f26b2

1 1

[alsa-devel] [PATCH 1/2] powerpc: add platform registration for ALSA SoC drivers
by Timur Tabi 01 May '10

01 May '10

An upcoming change in the architecture of ALSA SoC (ASoC) will require the MPC8610 HPCD's ASoC fabric driver to register as a standard platform driver. Therefore, we need to call platform_device_register_simple() from the board's platform code. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Kumar, the ASoC mainters are willing to pick up this patch, but they want an ACK from you first. Or, you could pick it up, since by itself it's harmless. arch/powerpc/platforms/86xx/mpc8610_hpcd.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c index 5abe137..66afff3 100644 --- a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c +++ b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c @@ -98,6 +98,9 @@ static int __init mpc8610_declare_of_platform_devices(void) /* Without this call, the SSI device driver won't get probed. */ of_platform_bus_probe(NULL, mpc8610_ids, NULL); + /* Register the platform device for the ASoC fabric driver */ + platform_device_register_simple("snd-soc-mpc8610-hpcd", 0, NULL, 0); + return 0; } machine_device_initcall(mpc86xx_hpcd, mpc8610_declare_of_platform_devices); -- 1.6.5

6 54

[alsa-devel] Relation between SND_PCM_IOPLUG_HW_PERIODS and SND_PCM_IOPLUG_HW_BUFFER_BYTES ?
by Stefan Schoenleitner 30 Apr '10

30 Apr '10

Hi, what is the relation between SND_PCM_IOPLUG_HW_PERIODS and SND_PCM_IOPLUG_HW_BUFFER_BYTES ? While SND_PCM_IOPLUG_HW_BUFFER_BYTES is the setting of the hardware buffer size, I'm not sure exactly what SND_PCM_IOPLUG_HW_PERIODS is. I suppose it is the number of periods the hardware buffer can take ? Hence the SND_PCM_IOPLUG_HW_PERIODS value is actually derived from the size of the hardware buffer ? Or is it rather the number of periods the ioplugin can receive at once ? Please enlighten me, cheers, Stefan

1 0

[alsa-devel] [PATCH 0/4] ASoC: tlv320dac33: Support for turning codec off
by Peter Ujfalusi 30 Apr '10

30 Apr '10

Hello, The following series adds support for turning the tlv320dac33 codec off in BIAS_STANDBY (and obviously in BIAS_OFF). Since the driver already had mechanics for supporting turned off codec in register writes, the needed changes are mainly in the BIAS/stream handling. As part of the series further optimization has been done to codec initialization, resume. The following scenarios has been tested, and verified (I think these are covering all cases): 1. Analog bypass caused BIAS_STANDBY -> BIAS_ON We need to power on the codec, and do the chip init, but we does not need to execute the playback related configuration 2. Playback caused BIAS_STANDBY -> BIAS_ON We need to power on the codec, and do the chip init, and also we need to execute the playback related configuration. 3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is already on. 4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON) Nothing need to be done. 5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is still on. 6. Analog bypass enable withing soc power down timeout (BIAS_ON -> BIAS_ON) Nothing need to be done. This series applies on top of Liam's: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.35 branch --- Peter Ujfalusi (4): ASoC: tlv320dac33: Optimize power up, and restore ASoC: tlv320dac33: Revised module loading, and DAC33 ID read ASoC: tlv320dac33: Manage a pointer for snd_pcm_substream in private structure ASoC: tlv320dac33: Support for turning off the codec in BIAS_STANDBY sound/soc/codecs/tlv320dac33.c | 236 +++++++++++++++++++++++----------------- 1 files changed, 137 insertions(+), 99 deletions(-)

2 13

[alsa-devel] PCM multi A/V Sync bug on SMP system
by Keith Chew 30 Apr '10

30 Apr '10

Hi I have a SB Audigy Value card with 5.1 channel output. This is defined in my asound.conf: --------------------------- pcm.multi { type multi slaves { a.pcm "dmixerrear" a.channels 2 b.pcm "dmixerfront" b.channels 2 c.pcm "dmixercenter" c.channels 2 } bindings { 0.slave a 0.channel 0 1.slave a 1.channel 1 2.slave b 2.channel 0 3.slave b 3.channel 1 4.slave c 4.channel 0 5.slave c 5.channel 1 } } pcm.dmixerfront { type dmix ipc_key 1024 slave { pcm "hw:0,0" channels 2 } } pcm.dmixerrear { type dmix ipc_key 1025 slave { pcm "hw:0,1" channels 2 } } pcm.dmixercenter { type dmix ipc_key 1026 slave { pcm "hw:0,2" channels 2 } } --------------------------- Playing a video with "mplayer -ao alsa:device=multi ..." works fine, I can hear the 6-channel audio without any A/V sync issues. However, this is on a uniprocessor system. On an SMP system, the A/V sync gets out of sync from the start and drifts quite quickly. I have narrowed it down to the pcm_multi component giving this problem. On an SMP system, using just "mplayer -ao alsa ..." gives no A/V sync issues (because it is not using multi) on the 6 channels. Moreover, I have also tried "mplayer -ao alsa:device=dmixerfront ...", "mplayer -ao alsa:device=dmixercenter ..." and "mplayer -ao alsa:device=dmixerrear ...", all giving no A/V sync issues on the individual devices. Please advise how can I track down this problem in code, happy to make changes and test. PS: I need multi to work, so that I can add additional options like softvol on top of it. Regards Keith

2 3

[alsa-devel] How to retrieve the final data being played by HW
by Yu Hang 30 Apr '10

30 Apr '10

I'm studying ALSA lib and trying to write a plugin that can perform AEC function. I have to retrieve the data being played and use it as reference. I've read pcm_dmix.c and found dmix just simply mixes data into HW mmap buffer (destination buffer). My problem is, there might be several dmix instances, which doesn't know if there is other dmix instance. So how can I know the mixed data is exactly the data being played. For example, the 1st instance had mixes the data into destination buffer, but 2nd instance haven't complete the mix operation. Would the data in destination buffer be playing by HW? Or there is some mechanism to guarantee the data will be played after all mix operation is done.

2 3

[alsa-devel] asoc: problem with snd_soc_dai_set_fmt()
by Timur Tabi 30 Apr '10

30 Apr '10

There is a problem with snd_soc_dai_set_fmt(): int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { if (dai->driver && dai->driver->ops->set_fmt) return dai->driver->ops->set_fmt(dai, fmt); else return -EINVAL; } Let's say the DAI driver has not defined a .set_fmt() function. This means that if the fabric driver does this: ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format); if (ret < 0) { dev_err(dev, "could not set CPU driver audio format\n"); return ret; } It's going to think that the DAI driver *rejected* the DAI format. What this means is that I cannot make this function optional. I have to define this function in my CPU driver. May I propose this: int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { if (dai->driver && dai->driver->ops->set_fmt) return dai->driver->ops->set_fmt(dai, fmt); else return 0; } -- Timur Tabi Linux kernel developer at Freescale

3 5

[alsa-devel] Wrote MIDIcat program, to connect ALSA MIDI to standard input/output
by Josh Lehan 30 Apr '10

30 Apr '10

Hi! A few months ago, I posted about a program to send untimed MIDI data to the ALSA sequencer. Using amidi(1) wasn't acceptable, since it only talks to "RawMIDI" devices. Using aplaymidi(1) wasn't acceptable either, because it required .MID files, and I simply wanted to send untimed MIDI data, similar to what would be produced if you hooked up a hardware MIDI keyboard and just started playing. I needed something more general, that would simply connect to the ALSA sequencer, and let me send arbitrary MIDI bytes to a sequencer destination of my choice. Unfortunately, no program existed that would simply do this. I'm pleased to announce that I've written such a program, It's called MIDIcat, as it works on the same principle as "cat" and other similar programs (Netcat, and so on). It hooks up standard input, and standard output, to the ALSA sequencer. This makes it easy to pipe data around. It's an ALSA sequencer client. Any data that is received on standard input will be forwarded into ALSA. You can choose another client as a destination, or you can just start the program passively and use ALSA's "subscription" mechanism to route the data later. Similarly, any MIDI data that comes from ALSA, will be forwarded along, and provided on standard output. This makes it easy to make many small "one-liner" command lines that are useful for testing and playing around with MIDI data in general. As an example, this plays Middle C: echo "90 3C 7F" | midicat --hex --port "TiMidity" The --hex option makes things more human-readable: standard input and output are hex digits, separated by spaces, instead of just raw binary data. The --port option has the usual ALSA sequencer meaning. Change it to target your synthesizer (I'm using the TiMidity softsynth). Here's another example for output (both of these commands block, so run them in two separate terminal windows): midicat --verbose --hex vkeybd --addr 129:0 The --verbose option prints out some more useful information at startup, such as the ALSA sequencer client:port number it got. That can later be passed into another program, such as vkeybd. If you play notes with vkeybd, you see the corresponding MIDI data appear on standard output. Standard input and standard output can both run at the same time (it's multithreaded). Here's some more elaborate examples: MIDI "panic button", sends "All Sound Off" to each channel: perl -e 'use bytes;for($i=0;$i<16;$i++){print chr(176+$i).chr(120).chr(0);}' | midicat --port "TiMidity" Another MIDI "panic button", sends individual "Note Off" commands: perl -e 'use bytes;for($i=0;$i<16;$i++){for($j=0;$j<128;$j++){print chr(128+$i).chr($j).chr(127);}}' | midicat --delay 10 --port "TiMidity" The --delay option inserts a slight delay between each MIDI command, to avoid flooding ALSA with too much data at once. Here's a fun one, sending random data into the MIDI subsystem and seeing what happens. DO NOT run this on a real synthesizer, there's the risk of composing a random SysEx command that could mess up your settings! It's fun to watch with aseqview(1): aseqview & cat /dev/urandom | midicat --delay 500 --port "MIDI Viewer" If you sent random data to a device that plays audio, you'll probably next want to use the "panic button" commands above...! Hope this illustrates some of the examples and possibilities. I'm working on writing some more documentation. The --help option is supported, there's a help output that shows you all of the available options. For now, the program is available here, on my Web server: http://krellan.com/midicat If you're interested, try it, and let me know what you think! Josh Lehan

2 1

[alsa-devel] My understanding of ALSA playback mechanism
by Yu Hang 30 Apr '10

30 Apr '10

Hi, I'm a newbie of ALSA. I've wrote some audio applications under Windows OS with DirectSound and Waveform Audio API. And I also have some knowledge about Sound Architecture of Windows Vista/Windows 7. All my experience about audio architecture was from Windows. I'm currently working on a ALSA plugin for AEC function. What I have to do is to retrieve audio data being played (after mixing) as reference to cancel the echo captured by MIC. According to my previous experience about Sound Architecture of Win7/Vista, the audio data is sending from OS component to my own component: no matter for capture or playback. All I have to do is just simply modify the input data and put them into output buffer. If my component is under OS mixer, the received audio data is after mixing. When I read source code of ALSA lib, I found the mechanism between Windows and ALSA lib are quite different. Following was my understanding about ALSA lib mechanism during playback. There is a hardware buffer. ALSA driver takes data from the buffer and sends it to HW. The driver also update the play position on the fly. dmix receive the data from application and mix them into a shared memory, and then copy it into hardware buffer. Each of dmix instance keep a pointer to indicate current mixing position which might different from other instance. [play pos] [1st mixed data] | | +----------------------------+ hardware buffer +----------------------------+ | [2nd mixed data] The figure shows, the data mixed from 1st dmix instance lasts to [1st mixed data] pos, and data mixed from 2nd dmix instance lasts to [2nd mixed data]. The buffer between [1st mixed data] and [2nd mixed data] contains only data of 2nd dmix instance, but it does not matter, ALSA just guarantees the data of 1st dmix instance can be mixed up before the [play pos] reaches [1st mixed data] pos. Of course the hardware buffer is not unlimited (it's a ring buffer), so before dmix mix data to hardware buffer, it has to check if there is enough space. Is my understanding of ALSA playback mechanism correct?

1 0

Re: [alsa-devel] [RFC] disabling ALSA period interrupts
by Raymond Yau 30 Apr '10

30 Apr '10

2010/4/30 pl bossart <bossart.nospam(a)gmail.com> > > do you mean that it only work on HDaudio and not intel8x0 ? > > > > How about other drivers ? > > It's driver/hardware specific, I guess everyone would have to enable > this in their drivers. I don't understand what 8x0 does...You're > welcome to test and do the same for 8x0 if you have the specs and know > how to disable interrupts. > > Any side effect of this disabling interrupt ? Do you mean that all applications must use pulse api instead of alsa-pulse plugin ? Did your test included those voip application which require low latency ? did your test only play one audio stream to the sound server ? Do those alsa application still work with "hw" device when pulseaudio is disabled ?

1 0

[alsa-devel] [PATCHv2 0/2] ASoC: TWL4030: APLL/AIF ordering and DAPM updates
by Peter Ujfalusi 29 Apr '10

29 Apr '10

Hello, Correct the APLL/AIF powering sequence. Also introducing DAPM outputs and input for keeping APLL/AIF enabled in case when no valid DAPM route selected during audio activity. The second patch removes the legacy OUTL/R. As a note: The APLL/AIF patch replaces the same patch in Liam's series. The OUTL/R removal can be applied after the OUTL/R reference has been removed from machine drivers (as in Lima's series). --- Peter Ujfalusi (2): ASoC: TWL4030: AIF/APLL fix in DAPM domain ASoC: TWL4030: Remove OUTL/R outputs sound/soc/codecs/twl4030.c | 91 ++++++++++++++++++++++++++++++++++---------- 1 files changed, 71 insertions(+), 20 deletions(-)

2 5

[alsa-devel] [PATCH] ASoC: TWL4030: AIF/APLL fix in DAPM domain
by Peter Ujfalusi 28 Apr '10

28 Apr '10

This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. The apll and aif reference counting code has been lifted, and reused from Liam Girdwood's earlier patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Some of the ideas (appl, aif refcounting) has been taken from Liam's patch: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-April/027217.html But I'm reusing the existing DAPM_SUPPLY widgets to handle the APLL and AIF enable. I have tested all possible scenarios, and the codec behaved correctly. The voice path was not verified due to missing HW, but that part has not been changed, so I expect no regressions related to that. sound/soc/codecs/twl4030.c | 91 ++++++++++++++++++++++++++++++++++---------- 1 files changed, 71 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2e025a3..816a995 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -123,7 +123,10 @@ struct twl4030_priv { struct snd_soc_codec codec; unsigned int codec_powered; + + /* reference counts of AIF/APLL users */ unsigned int apll_enabled; + unsigned int aif_enabled; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; @@ -259,22 +262,49 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int status; - - if (enable == twl4030->apll_enabled) - return; + int status = -1; if (enable) + twl4030->apll_enabled++; + else + twl4030->apll_enabled--; + + if (enable && twl4030->apll_enabled == 1) /* Enable PLL */ status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL); - else + else if (!enable && twl4030->apll_enabled == 0) /* Disable PLL */ status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL); if (status >= 0) twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); +} - twl4030->apll_enabled = enable; +static void twl4030_aif_enable(struct snd_soc_codec *codec, int enable) +{ + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); + u8 audio_if; + + /* update the user count */ + if (enable) + twl4030->aif_enabled++; + else + twl4030->aif_enabled--; + + audio_if = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + if (enable && twl4030->aif_enabled == 1) { + /* Enable AIF */ + /* enable the PLL before we use it to clock the DAI */ + twl4030_apll_enable(codec, 1); + + twl4030_write(codec, TWL4030_REG_AUDIO_IF, + audio_if | TWL4030_AIF_EN); + } else if (!enable && twl4030->aif_enabled == 0) { + /* disable the DAI before we stop it's source PLL */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, + audio_if & ~TWL4030_AIF_EN); + twl4030_apll_enable(codec, 0); + } } static void twl4030_power_up(struct snd_soc_codec *codec) @@ -672,6 +702,20 @@ static int apll_event(struct snd_soc_dapm_widget *w, return 0; } +static int aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + twl4030_aif_enable(w->codec, 1); + break; + case SND_SOC_DAPM_POST_PMD: + twl4030_aif_enable(w->codec, 0); + break; + } + return 0; +} + static void headset_ramp(struct snd_soc_codec *codec, int ramp) { struct snd_soc_device *socdev = codec->socdev; @@ -1167,8 +1211,6 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DIGIMIC1"), /* Outputs */ - SND_SOC_DAPM_OUTPUT("OUTL"), - SND_SOC_DAPM_OUTPUT("OUTR"), SND_SOC_DAPM_OUTPUT("EARPIECE"), SND_SOC_DAPM_OUTPUT("PREDRIVEL"), SND_SOC_DAPM_OUTPUT("PREDRIVER"), @@ -1180,6 +1222,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFR"), SND_SOC_DAPM_OUTPUT("VIBRA"), + /* AIF and APLL clocks for running DAIs (including loopback) */ + SND_SOC_DAPM_OUTPUT("AIF DAC"), + SND_SOC_DAPM_INPUT("AIF ADC"), + SND_SOC_DAPM_OUTPUT("APLL"), + /* DACs */ SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback", SND_SOC_NOPM, 0, 0), @@ -1243,7 +1290,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF Enable", SND_SOC_NOPM, 0, 0, aif_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), /* Output MIXER controls */ /* Earpiece */ @@ -1373,10 +1421,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital Voice Playback Mixer", NULL, "DAC Voice"}, /* Supply for the digital part (APLL) */ - {"Digital R1 Playback Mixer", NULL, "APLL Enable"}, - {"Digital L1 Playback Mixer", NULL, "APLL Enable"}, - {"Digital R2 Playback Mixer", NULL, "APLL Enable"}, - {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, @@ -1450,8 +1494,14 @@ static const struct snd_soc_dapm_route intercon[] = { {"Vibra Mux", "AudioR2", "DAC Right2"}, /* outputs */ - {"OUTL", NULL, "Analog L2 Playback Mixer"}, - {"OUTR", NULL, "Analog R2 Playback Mixer"}, + /* Must be always connected (for AIF and APLL) */ + {"AIF DAC", NULL, "Digital L1 Playback Mixer"}, + {"AIF DAC", NULL, "Digital R1 Playback Mixer"}, + {"AIF DAC", NULL, "Digital L2 Playback Mixer"}, + {"AIF DAC", NULL, "Digital R2 Playback Mixer"}, + /* Must be always connected (for APLL) */ + {"APLL", NULL, "Digital Voice Playback Mixer"}, + /* Physical outputs */ {"EARPIECE", NULL, "Earpiece PGA"}, {"PREDRIVEL", NULL, "PredriveL PGA"}, {"PREDRIVER", NULL, "PredriveR PGA"}, @@ -1465,6 +1515,12 @@ static const struct snd_soc_dapm_route intercon[] = { {"VIBRA", NULL, "Vibra Route"}, /* Capture path */ + /* Must be always connected (for AIF and APLL) */ + {"ADC Virtual Left1", NULL, "AIF ADC"}, + {"ADC Virtual Right1", NULL, "AIF ADC"}, + {"ADC Virtual Left2", NULL, "AIF ADC"}, + {"ADC Virtual Right2", NULL, "AIF ADC"}, + /* Physical inputs */ {"Analog Left", "Main Mic Capture Switch", "MAINMIC"}, {"Analog Left", "Headset Mic Capture Switch", "HSMIC"}, {"Analog Left", "AUXL Capture Switch", "AUXL"}, @@ -1497,11 +1553,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, - {"ADC Virtual Left1", NULL, "APLL Enable"}, - {"ADC Virtual Right1", NULL, "APLL Enable"}, - {"ADC Virtual Left2", NULL, "APLL Enable"}, - {"ADC Virtual Right2", NULL, "APLL Enable"}, - {"ADC Virtual Left1", NULL, "AIF Enable"}, {"ADC Virtual Right1", NULL, "AIF Enable"}, {"ADC Virtual Left2", NULL, "AIF Enable"}, -- 1.7.0.4

1 0

[alsa-devel] [PATCH 1/5] ASoC: twl4030 - AIF/APLL clock fix for all DAPM routes.
by Liam Girdwood 28 Apr '10

28 Apr '10

This allows the AIF and APLL clocks to continue running in loopback mode and when invalid DAPM routes are selected at stream runtime. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/codecs/twl4030.c | 123 +++++++++++++++++++++++++++++-------------- 1 files changed, 83 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index dc26b71..1a3e7f6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -123,7 +123,10 @@ struct twl4030_priv { struct snd_soc_codec codec; unsigned int codec_powered; + + /* reference counts of AIF/APLL users */ unsigned int apll_enabled; + unsigned int aif_enabled; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; @@ -259,22 +262,50 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int status; - - if (enable == twl4030->apll_enabled) - return; + int status = -1; + /* update the user count */ if (enable) + twl4030->apll_enabled++; + else + twl4030->apll_enabled--; + + if (enable && twl4030->apll_enabled == 1) /* Enable PLL */ status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL); - else + else if (!enable && twl4030->apll_enabled == 0) /* Disable PLL */ status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL); if (status >= 0) twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); +} - twl4030->apll_enabled = enable; +static void twl4030_aif_enable(struct snd_soc_codec *codec, int enable) +{ + struct twl4030_priv *twl4030 = codec->private_data; + + /* update the user count */ + if (enable) + twl4030->aif_enabled++; + else + twl4030->aif_enabled--; + + if (enable && twl4030->aif_enabled == 1) { + /* Enable AIF */ + /* enable the PLL before we use it to clock the DAI */ + twl4030_apll_enable(codec, 1); + + twl4030_write(codec, TWL4030_REG_AUDIO_IF, + twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF) + | TWL4030_AIF_EN); + } else if (!enable && twl4030->aif_enabled == 0) { + /* disable the DAI before we stop it's source PLL */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, + twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF) + & ~TWL4030_AIF_EN); + twl4030_apll_enable(codec, 0); + } } static void twl4030_power_up(struct snd_soc_codec *codec) @@ -672,6 +703,20 @@ static int apll_event(struct snd_soc_dapm_widget *w, return 0; } +static int aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + twl4030_aif_enable(w->codec, 1); + break; + case SND_SOC_DAPM_POST_PMD: + twl4030_aif_enable(w->codec, 0); + break; + } + return 0; +} + static void headset_ramp(struct snd_soc_codec *codec, int ramp) { struct snd_soc_device *socdev = codec->socdev; @@ -1178,6 +1223,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFR"), SND_SOC_DAPM_OUTPUT("VIBRA"), + /* AIF and APLL clocks for running DAIs (including loopback) */ + SND_SOC_DAPM_OUTPUT("AIF DAC"), + SND_SOC_DAPM_INPUT("AIF ADC"), + SND_SOC_DAPM_OUTPUT("APLL"), + /* DACs */ SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback", SND_SOC_NOPM, 0, 0), @@ -1215,16 +1265,21 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_dbypassv_control), /* Digital mixers, power control for the physical DACs */ - SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer", - TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Digital L1 Playback Mixer", - TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Digital R2 Playback Mixer", - TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Digital L2 Playback Mixer", - TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Digital Voice Playback Mixer", - TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Digital R1 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0, aif_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("Digital L1 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0, aif_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("Digital R2 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0, aif_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("Digital L2 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0, aif_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER_E("Digital Voice Playback Mixer", + TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0, apll_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), /* Analog mixers, power control for the physical PGAs */ SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", @@ -1238,11 +1293,6 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, - SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), - - SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0), - /* Output MIXER controls */ /* Earpiece */ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, @@ -1370,17 +1420,14 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital R2 Playback Mixer", NULL, "DAC Right2"}, {"Digital Voice Playback Mixer", NULL, "DAC Voice"}, - /* Supply for the digital part (APLL) */ - {"Digital R1 Playback Mixer", NULL, "APLL Enable"}, - {"Digital L1 Playback Mixer", NULL, "APLL Enable"}, - {"Digital R2 Playback Mixer", NULL, "APLL Enable"}, - {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, - {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, + /* Route for the digital part (APLL) */ + {"APLL", NULL, "Digital Voice Playback Mixer"}, - {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, - {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, - {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, - {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, + /* Route for digital part (AIF) */ + {"AIF DAC", NULL, "Digital L1 Playback Mixer"}, + {"AIF DAC", NULL, "Digital R1 Playback Mixer"}, + {"AIF DAC", NULL, "Digital L2 Playback Mixer"}, + {"AIF DAC", NULL, "Digital R2 Playback Mixer"}, {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, @@ -1493,15 +1540,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, - {"ADC Virtual Left1", NULL, "APLL Enable"}, - {"ADC Virtual Right1", NULL, "APLL Enable"}, - {"ADC Virtual Left2", NULL, "APLL Enable"}, - {"ADC Virtual Right2", NULL, "APLL Enable"}, - - {"ADC Virtual Left1", NULL, "AIF Enable"}, - {"ADC Virtual Right1", NULL, "AIF Enable"}, - {"ADC Virtual Left2", NULL, "AIF Enable"}, - {"ADC Virtual Right2", NULL, "AIF Enable"}, + /* Route for digital part AIF */ + {"ADC Virtual Left1", NULL, "AIF ADC"}, + {"ADC Virtual Right1", NULL, "AIF ADC"}, + {"ADC Virtual Left2", NULL, "AIF ADC"}, + {"ADC Virtual Right2", NULL, "AIF ADC"}, /* Analog bypass routes */ {"Right1 Analog Loopback", "Switch", "Analog Right"}, -- 1.6.3.3

2 6

[alsa-devel] [GIT PULL] ASoC updates for 2.6.35
by Liam Girdwood 27 Apr '10

27 Apr '10

The following changes since commit f4d593280652025f797d432e90f49d49b9334689: Peter Ujfalusi (1): ASoC: tlv320dac33: Fix for early interrupt in FIFO Mode1 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.35 Jarkko Nikula (4): ASoC: tlv320aic3x: Remove unused version string ASoC: tlv320aic3x: Remove needless power off from aic3x_set_bias_level ASoC: tlv320aic3x: Change bias management semantics ASoC: tlv320aic3x: Add basic regulator support Peter Ujfalusi (4): ASoC: tlv320dac33: Skip calculations in FIFO Bypass mode ASoC: tlv320dac33: Change magic numbers used in Mode7 ASoC: tlv320dac33: Calculate the interface speed during bursts ASoC: tlv320dac33: FIFO caused delay reporting sound/soc/codecs/tlv320aic3x.c | 81 +++++++------- sound/soc/codecs/tlv320dac33.c | 243 ++++++++++++++++++++++++++++++++++++++-- 2 files changed, 272 insertions(+), 52 deletions(-)

2 1

[alsa-devel] [PATCH 1/4] ASoC: tlv320aic3x: Remove unused version string
by Jarkko Nikula 27 Apr '10

27 Apr '10

Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 5be580e..af463d8 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -50,8 +50,6 @@ #include "tlv320aic3x.h" -#define AIC3X_VERSION "0.2" - /* codec private data */ struct aic3x_priv { struct snd_soc_codec codec; -- 1.7.0

3 5

[alsa-devel] [PATCH 0/3] ad193x set_pll, fix register setting and typo
by Barry Song 27 Apr '10

27 Apr '10

Hi Mark/Liam, The following 3 patches add new features or fix bugs in ad193x drivers. 1/3. add set_pll entry to support different clock input 2/3. fix wrong register setting in ad193x_set_dai_fmt 3/3. fix typo, delete redundant space Thanks Barry

3 9

[alsa-devel] ASoC: OMAP/TI upstream route
by Liam Girdwood 27 Apr '10

27 Apr '10

Now that I'm back working full time on ASoC I'm going to manage the upstream branch for the OMAP platform and TI codecs. Mark will continue to manage the upstream for all other codecs, platforms and core. As usual, please CC both Mark and I on new ASoC patches. The OMAP upstream branch tracks Takashi's topic/asoc branch and is here :- git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git #for-2.6.35 Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

3 5

[alsa-devel] [PATCH 01/11] ASoC: S3C: I2Sv2: Move defines closer to driver
by Jassi Brar 27 Apr '10

27 Apr '10

The header for I2Sv2 linux/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h contains only controller specific definitions and nothing SoC specific. So, it could be moved to sound/soc/s3c24xx/ Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- .../plat-samsung/include/plat/regs-s3c2412-iis.h | 82 -------------------- sound/soc/s3c24xx/regs-i2s-v2.h | 82 ++++++++++++++++++++ sound/soc/s3c24xx/s3c-i2s-v2.c | 3 +- sound/soc/s3c24xx/s3c2412-i2s.c | 3 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 2 +- 5 files changed, 85 insertions(+), 87 deletions(-) delete mode 100644 arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h create mode 100644 sound/soc/s3c24xx/regs-i2s-v2.h diff --git a/arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h deleted file mode 100644 index abf2fbc..0000000 --- a/arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h +++ /dev/null @@ -1,82 +0,0 @@ -/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h - * - * Copyright 2007 Simtec Electronics <linux(a)simtec.co.uk> - * http://armlinux.simtec.co.uk/ - * - * 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. - * - * S3C2412 IIS register definition -*/ - -#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H -#define __ASM_ARCH_REGS_S3C2412_IIS_H - -#define S3C2412_IISCON (0x00) -#define S3C2412_IISMOD (0x04) -#define S3C2412_IISFIC (0x08) -#define S3C2412_IISPSR (0x0C) -#define S3C2412_IISTXD (0x10) -#define S3C2412_IISRXD (0x14) - -#define S3C2412_IISCON_LRINDEX (1 << 11) -#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) -#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) -#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) -#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) -#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) -#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) -#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) -#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) -#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) -#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) -#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) - -#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) -#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) -#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) -#define S3C64XX_IISMOD_BLC_MASK (3 << 13) - -#define S3C64XX_IISMOD_IMS_PCLK (0 << 10) -#define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) - -#define S3C2412_IISMOD_MASTER_INTERNAL (0 << 10) -#define S3C2412_IISMOD_MASTER_EXTERNAL (1 << 10) -#define S3C2412_IISMOD_SLAVE (2 << 10) -#define S3C2412_IISMOD_MASTER_MASK (3 << 10) -#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) -#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) -#define S3C2412_IISMOD_MODE_TXRX (2 << 8) -#define S3C2412_IISMOD_MODE_MASK (3 << 8) -#define S3C2412_IISMOD_LR_LLOW (0 << 7) -#define S3C2412_IISMOD_LR_RLOW (1 << 7) -#define S3C2412_IISMOD_SDF_IIS (0 << 5) -#define S3C2412_IISMOD_SDF_MSB (1 << 5) -#define S3C2412_IISMOD_SDF_LSB (2 << 5) -#define S3C2412_IISMOD_SDF_MASK (3 << 5) -#define S3C2412_IISMOD_RCLK_256FS (0 << 3) -#define S3C2412_IISMOD_RCLK_512FS (1 << 3) -#define S3C2412_IISMOD_RCLK_384FS (2 << 3) -#define S3C2412_IISMOD_RCLK_768FS (3 << 3) -#define S3C2412_IISMOD_RCLK_MASK (3 << 3) -#define S3C2412_IISMOD_BCLK_32FS (0 << 1) -#define S3C2412_IISMOD_BCLK_48FS (1 << 1) -#define S3C2412_IISMOD_BCLK_16FS (2 << 1) -#define S3C2412_IISMOD_BCLK_24FS (3 << 1) -#define S3C2412_IISMOD_BCLK_MASK (3 << 1) -#define S3C2412_IISMOD_8BIT (1 << 0) - -#define S3C64XX_IISMOD_CDCLKCON (1 << 12) - -#define S3C2412_IISPSR_PSREN (1 << 15) - -#define S3C2412_IISFIC_TXFLUSH (1 << 15) -#define S3C2412_IISFIC_RXFLUSH (1 << 7) -#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) -#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) - - - -#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ - diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h new file mode 100644 index 0000000..abf2fbc --- /dev/null +++ b/sound/soc/s3c24xx/regs-i2s-v2.h @@ -0,0 +1,82 @@ +/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h + * + * Copyright 2007 Simtec Electronics <linux(a)simtec.co.uk> + * http://armlinux.simtec.co.uk/ + * + * 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. + * + * S3C2412 IIS register definition +*/ + +#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H +#define __ASM_ARCH_REGS_S3C2412_IIS_H + +#define S3C2412_IISCON (0x00) +#define S3C2412_IISMOD (0x04) +#define S3C2412_IISFIC (0x08) +#define S3C2412_IISPSR (0x0C) +#define S3C2412_IISTXD (0x10) +#define S3C2412_IISRXD (0x14) + +#define S3C2412_IISCON_LRINDEX (1 << 11) +#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) +#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) +#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) +#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) +#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) +#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) +#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) +#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) +#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) +#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) +#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) + +#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) +#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) +#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) +#define S3C64XX_IISMOD_BLC_MASK (3 << 13) + +#define S3C64XX_IISMOD_IMS_PCLK (0 << 10) +#define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) + +#define S3C2412_IISMOD_MASTER_INTERNAL (0 << 10) +#define S3C2412_IISMOD_MASTER_EXTERNAL (1 << 10) +#define S3C2412_IISMOD_SLAVE (2 << 10) +#define S3C2412_IISMOD_MASTER_MASK (3 << 10) +#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) +#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) +#define S3C2412_IISMOD_MODE_TXRX (2 << 8) +#define S3C2412_IISMOD_MODE_MASK (3 << 8) +#define S3C2412_IISMOD_LR_LLOW (0 << 7) +#define S3C2412_IISMOD_LR_RLOW (1 << 7) +#define S3C2412_IISMOD_SDF_IIS (0 << 5) +#define S3C2412_IISMOD_SDF_MSB (1 << 5) +#define S3C2412_IISMOD_SDF_LSB (2 << 5) +#define S3C2412_IISMOD_SDF_MASK (3 << 5) +#define S3C2412_IISMOD_RCLK_256FS (0 << 3) +#define S3C2412_IISMOD_RCLK_512FS (1 << 3) +#define S3C2412_IISMOD_RCLK_384FS (2 << 3) +#define S3C2412_IISMOD_RCLK_768FS (3 << 3) +#define S3C2412_IISMOD_RCLK_MASK (3 << 3) +#define S3C2412_IISMOD_BCLK_32FS (0 << 1) +#define S3C2412_IISMOD_BCLK_48FS (1 << 1) +#define S3C2412_IISMOD_BCLK_16FS (2 << 1) +#define S3C2412_IISMOD_BCLK_24FS (3 << 1) +#define S3C2412_IISMOD_BCLK_MASK (3 << 1) +#define S3C2412_IISMOD_8BIT (1 << 0) + +#define S3C64XX_IISMOD_CDCLKCON (1 << 12) + +#define S3C2412_IISPSR_PSREN (1 << 15) + +#define S3C2412_IISFIC_TXFLUSH (1 << 15) +#define S3C2412_IISFIC_RXFLUSH (1 << 7) +#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) +#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) + + + +#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ + diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 865f931..cb3652e 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -24,10 +24,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <plat/regs-s3c2412-iis.h> - #include <mach/dma.h> +#include "regs-i2s-v2.h" #include "s3c-i2s-v2.h" #include "s3c-dma.h" diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index f3148f9..7134465 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -32,12 +32,11 @@ #include <sound/soc.h> #include <mach/hardware.h> -#include <plat/regs-s3c2412-iis.h> - #include <mach/regs-gpio.h> #include <mach/dma.h> #include "s3c-dma.h" +#include "regs-i2s-v2.h" #include "s3c2412-i2s.h" #define S3C2412_I2S_DEBUG 0 diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index ab1fa15..b7e951f 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -18,7 +18,6 @@ #include <sound/soc.h> -#include <plat/regs-s3c2412-iis.h> #include <mach/gpio-bank-d.h> #include <mach/gpio-bank-e.h> #include <plat/gpio-cfg.h> @@ -27,6 +26,7 @@ #include <mach/dma.h> #include "s3c-dma.h" +#include "regs-i2s-v2.h" #include "s3c64xx-i2s.h" /* The value should be set to maximum of the total number -- 1.6.2.5

2 1